Grandstream Networks, Inc.
HT386 Dual FXS Port Analog Telephone Adaptor
HT386 User Manual Firmware Version 1.0.3.64
www.grandstream.com support@grandstream.com

TABLE OF CONTENTS

HT386 USER MANUAL WELCOME..... 4 Safety Compliances..... 4 Warranty..... 4 INSTALLATION..... 5 Equipment Packaging.... 5 Connecting Your ATA.... 5 Five easy steps to install the HT386... 6 PRODUCT OVERVIEW..... 7 Key Features..... 7 Hardware Specification..... 9 BASIC OPERATIONS..... 10 Get Familiar with Voice Prompt.... 10 Make Phone Calls..... 11 Calling Phone or Extension Numbers.... 11 Direct IP Calls..... 12 Call Hold..... 12 Call Waiting..... 12 Call transfer..... 12 3-way Conferencing..... 13 PSTN Pass Through.... 14 Fax Support..... 14 CALL FEATURES.... 15 CONFIGURATION GUIDE.... 17 Configuring HT386 through Voice Prompt.... 17 DHCP Mode..... 17 Static IP Mode..... 17 TFTP Server Address..... 17 Firmware Server IP Address.... 17 Configuration Server IP Address... 17 Upgrade Protocol.... 17 Firmware Upgrade Mode.... 17 Configuring HT-386 with Web Browser... 18 Access the Web Configuration Menu.... 18 End User Configuration.... 18 Advanced Configuration and FXS ports Parameters... 21 Saving the Configuration Changes.... 26 REBOOTING THE HT386 FROM REMOTE.... 26 Configuration through a Central Server... 27 SOFTWARE UPGRADE.... 28 Firmware Upgrade through TFTP/HTTP... 28 IVR METHOD..... 28 UPGRADE THROUGH TFTP.... 28 NO LOCAL TFTP SERVER.... 28 Configuration File Download.... 29 Firmware and Configuration File Prefix and Postfix... 29 Managing Firmware and Configuration File Download... 29 RESTORE FACTORY DEFAULT SETTING.... 30 Reset Via the Reset Button.... 30 Reset Via IVR..... 30 GLOSSARY OF TERMS.... 31
Grandstream Networks, Inc. HT-386 User Manual Firmware 1.0.3.64 Page 2 of 34 Last Updated: 2/2007

TABLE OF FIGURES

HT386 USER MANUAL FIGURE 1: FIGURE 2: FIGURE 3: FIGURE 4: FIGURE 5: CONNECTING THE HT386... 5 INTERCONNECTION DIAGRAM OF THE HT386... 6 SCREENSHOT OF CONFIGURATION LOG-IN PAGE... 18 SCREENSHOT OF CONFIGURATION UPDATE MODE... 26 SCREENSHOT OF REBOOTING SCREEN... 27

TABLE OF TABLES

HT386 USER MANUAL TABLE 1: DEFINITIONS OF THE HT386 CONNECTORS... 5 TABLE 2: HT386 TECHNICAL SPECIFICATIONS.... 8 TABLE 3: HT386 HARDWARE SPECIFICATION.... 9 TABLE 4: HT386 IVR MENU DEFINITIONS.... 10 TABLE 5: HT386 CALL FEATURE DEFINITIONS... 15 TABLE 6: HT386 LED DEFINITIONS.... 16 TABLE 7: HT386 DEVICE STATUS PAGE DEFINITIONS... 19 TABLE 8: HT386 BASIC SETTINGS PAGE DEFINITIONS.... 20 TABLE 9: HT386 ADVANCED SETTINGS PAGE DEFINITIONS... 21 TABLE 10: HT386 FXS PORT1/FXS PORT2 SETTINGS PAGES DEFINITIONS... 23 TABLE 11: HT386 CALL PROGRESS TONES SETTINGS PAGE DEFINITIONS.. 26

TABLE OF GUI INTERFACES

HT386 USER MANUAL (http://www.grandstream.com/GUI/GUI_HT386.rar)
1. 2. 3. 4. 5. 6. 7. 8. 9.
SCREENSHOT OF CONFIGURATION LOGIN PAGE STATUS CONFIGURATION PAGE DEFINITIONS SCREENSHOT OF BASIC SETTINGS CONFIGURATION PAGE SCREENSHOT OF ADVANCED SETTINGS CONFIGURATION PAGE SCREENSHOT OF FXS1 ACCOUNT CONFIGURATION SCREENSHOT OF FXS2 ACCOUNT CONFIGURATION SCREENSHOT OF CALL PROGRESS TONES CONFIGURATION PAGE SCREENSHOT OF SAVED CONFIGURATION CHANGES SCREENSHOT OF REBOOT PAGE
HT-386 User Manual Firmware 1.0.3.64
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WELCOME

Grandstream HandyTone Analog Telephone Adapters/IAD series offers a comprehensive line of affordable VoIP access devices based on Grandstreams 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. The HandyTone series is ultra-compact, works with any PSTN or cordless phone and fax machines and offers the simplicity of plug and dial, making it ideal for the basic IP telephony user. HandyTone 386 is a next generation VoIP integrated access device based on SIP standard, that supports dual- 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. HandyTone 386 features market-leading superb sound quality, rich functionalities, and a compact design.

SAFETY COMPLIANCES The HT386 adaptor complies with FCC/CE and various safety standards. The HT386 power adaptor is compliant with UL standard. Only use the universal power adapter provided with the HT386 package. The manufacturers warranty does not cover damages to the phone caused by unsupported power adaptors. WARRANTY If you purchased your HT386 from a reseller, please contact the company where you purchased your phone 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 before you return the product. Grandstream reserves the right to remedy warranty policy without prior notification. Warning: 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. Please do not use a different power adaptor with the HT386 as it may cause damage to the products and void the manufacturer warranty.
This document is contains links to Grandstream GUI Interfaces. Please download these examples http://www.grandstream.com/user_manuals/GUI/GUI_HT386.rar for your reference. This document is subject to change without notice. The latest electronic version of this user manual is available for download @: http://www.grandstream.com/user_manuals/HT386_User_Manual.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.
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INSTALLATION

EQUIPMENT PACKAGING
The HT386 ATA package contains: One HT386 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.

FIGURE 1: CONNECTING THE HT-386

LED (green/red)

+5V/1200mA

RJ-11 PSTN Line

RJ-11 FXS Port (Phone)
RJ-45 10M Ethernet LAN/WAN
TABLE 1: DEFINITIONS OF THE HT386 CONNECTORS +5V/1.2A LAN Port (RJ-45) PHONE1 (RJ-11) PHONE2 (RJ-11) LINE (RJ-11) BUTTON Power adapter connection Connect to the internal LAN network or router. FXS port to be connected to analog phones / fax machines. FXS port to be connected to analog phones / fax machines. FXO port should be connected to the PSTN line Button and two colors led indicator.
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FIVE EASY STEPS TO INSTALL THE HT386
Following are the steps to install a HT386: 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 HT386. and connect the other end of the telephone cable to a wall jack. 4. Insert the Ethernet cable into the LAN port of HT386. and connect the other end of the Ethernet cable to an uplink port (a router or a modem, etc.) 5. Insert the power adapter into the HT386 and connect it to a wall outlet. For more information, see Configuring the HT386. The HT386 is easy to configure and easy to interconnect with your existing communication devices.HT 386 has two FXS ports and one RJ-11 jack on the side that is a LINE port, used as a PSTN pass-through port. Each FXS port has a separate SIP account which allows both ports to make calls concurrently. FIGURE 2: INTERCONNECTION DIAGRAM OF THE HT386
Internet ADSL/Cable Modem Ethernet

Analog Phone

Cordless Phone
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PRODUCT OVERVIEW

The HT386 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 HT386 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. The HandyTone is compact, works with any PSTN or cordless phone and fax machines and offers the simplicity of plug and dial, making it ideal for the basic IP telephony user.

KEY FEATURES

Ethernet Ports

1 RJ45 (LAN)

FXS Port

PSTN Pass through

Voice Mail Indicator

Voice Codec

iLBC, G.723, G.711, G.729, G.726, T.38

Remote Configuration

TFTP/HTTP

Client

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TABLE 2: HT386 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-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 (802.1Q, VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS) Auto/manual provisioning system NAT-friendly remote software upgrade (via TFTP/HTTP) for deployed devices including behind firewall/NAT Syslog support Yes, Client Advanced Digital Signal Processing (DSP) Dynamic negotiation of codec and voice payload length Support for G.723,1 (5.3K/6.3K), G.729A, G.711 /A, G.726, and iLBC codecs In-band and out-of-band DTMF ((in audio, RFC2833, SIP INFO) Silence Suppression, VAD (voice activity detection), CNG (comfort noise generation), ANG (automatic gain control) Adaptive jitter buffer control Packet delay & loss concealment Support volume amplification Support configurable Call Progress Tones Caller ID display or block, Call waiting caller ID, Call waiting/flash, Call transfer, hold, forward, mute, 3-way conferencing Manual or dynamic host configuration protocol (DHCP) network setup; RTP and NAT support traversal via STUN T.38 compliant Group 3 Fax Relay up to 14.4kpbs and auto-switch to G.711 for Fax Pass-through (pending), Fax Datapump V.17, V.19, V.27ter, V.29 for T.38 fax relay DIGEST authentication and encryption using MD5 and MD5-sess Stylish and compact design; small universal power supply, ideal for travel
DHCP Server/Client Audio Features
Call Handling Features Network and Provisioning Fax over IP

Security Physical Design

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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 Compliance 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)
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BASIC OPERATIONS

GET FAMILIAR WITH VOICE PROMPT
The HT386 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.
TABLE 4: HT-386 IVR MENU DEFINITIONS Menu Main Menu Voice Prompt Enter a Menu Option Options Press * for the next menu option Press # to return to the main menu Enter 01-06, 47, 86, 99 menu options Press 9 to toggle the selection If using Static IP Mode, configure the IP address information using menus 02 to 05. If using Dynamic IP Mode, all IP address information comes from the DHCP server automatically after reboot. The current WAN IP address is announced If using Static IP Mode, enter 12 digit new IP address. Same as menu 02 Same as menu 02 Same as menu 02 Press 9 to move to the next selection in the list: WAN Port Web Access Firmware Server IP Address Configuration Server IP Address Upgrade Protocol Firmware Version Firmware Upgrade PCM U / PCM A G.723 G.729 G.726 iLBC
DHCP Mode, Static IP Mode
IP Address + IP address Subnet + IP address Gateway + IP address DNS Server + IP address Preferred Vocoder
Press 9 to toggle between enable / disable Announces current Firmware Server IP address. Enter 12 digit new IP address. Announces current Config Server Path IP address. Enter 12 digit new IP address. Upgrade protocol for firmware and configuration update. Press 9 to toggle between TFTP / HTTP Firmware version information. Firmware upgrade mode. Press 9 to toggle among the following three options: - always check - check when pre/suffix changes - never upgrade
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Direct IP Calling RESET

Enter a 12 digit IP address to make a direct IP call, after dial tone. (See Make a Direct IP Call.) Press 9 to reboot the device; or Enter encoded MAC address to restore factory default setting (See Restoring Factory Settings) Automatically returns to main menu

Invalid Entry

NOTE: Once the button is pressed, you will hear the voice prompt main menu. If the button is pressed again, while it is already in the voice prompt menu, it jumps to Direct IP Call option and a dial tone is prompted * shifts down to the next menu option # returns to the main menu 9 functions as the ENTER key in many cases to confirm an option All entered digit sequences have known lengths - 2 digits for menu option and 12 digits for IP address. For IP address, add 0 before the digits if the digits are less than 3 (like 192.168.0.26 should be key in like 192168000026, no dot needed while input). Once all of the digits are collected, the input will be processed. Key entry can not be deleted but the phone may prompt error once it is detected

MAKE PHONE CALLS

CALLING PHONE OR EXTENSION NUMBERS There are currently two methods to make an extension number call: a) Dial the numbers directly and wait for 4 (default) seconds. b) Dial the numbers directly, and press # (assuming that use # as dial key is selected in web configuration). Examples: To dial another extension on the same proxy, such as 1008, simply pick up the attached phone, dial 1008 and then press the # or wait for 4 seconds. To dial a PSTN number such as 6266667890, you might need to enter in some prefix number followed by the phone number. Please check with your VoIP service provider to get the information. If you phone is assigned with a PSTN-like number such as 6265556789, most likely you just follow the rule to dial 16266667890 as if you were calling from a regular analog phone of North America, then followed by pressing # or wait for 4 seconds.
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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 HT386 and other VoIP Device(i.e., another HT ATA or Budgetone SIP phone or other VoIP unit) have public IP addresses, or Both the HT386 and other VoIP Device are on the same LAN using private IP addresses, or Both the HT386 and other VoIP Device can be connected through a router using public or private IP addresses (with necessary port forwarding or DMZ).
To make a direct IP call, first pick up the analog phone or turn on the speakerphone on the analog phone, then access the voice menu prompt by dial *** or press the button on the HT-286, and dials 47 to access the direct IP call menu. User will hear a voice prompt Direct IP Calling and a dial tone. Enter a 12-digit target IP address to make a call. Destination ports can be specified by using *4 (encoding for :) followed by the port number. Examples: 1. If the target IP address is 192.168.0.10, the dialing convention is Voice Prompt with option 47, then followed by the # key if it is configured as a send key or wait for more than 5 seconds. 2. If the target IP address/port is 192.168.1.20:5062, then the dialing convention would be: Voice Prompt with option 47, then 192168001020*45062 followed by the # key if it is configured as a send key or wait for 4 seconds. NOTE: When completing a direct IP call, the Use Random Port should set to NO. You can NOT make direct IP calls between FXS1 to FXS2 since they are using same IP.
CALL HOLD Place a call on hold by pressing the flash button on the analog phone (if the phone has that button). Press the flash button again to release the previously held Caller and resume conversation. If no flash button is available, use hook flash (toggle on-off hook quickly). You may drop a call using hook flash. CALL WAITING Call waiting tone (3 short beeps) indicates an incoming call, if the call waiting feature is enabled. Toggle between incoming call and current call by pressing the flash button. First call is placed on hold. Press the flash button to toggle between two active calls. CALL TRANSFER Assume that call Caller A and B are in conversation. A wants to Blind Transfer B to C: 1. Caller A presses FLASH on the analog phone to hear the dial tone. 2. Caller A dials *87 then dials caller Cs number, and then # (or wait for 4 seconds) 3. Caller A will hear the confirm tone. Then, A can hang up.

FAX SUPPORT HT386 supports FAX in two modes: T.38 (Fax over IP) and fax pass through. T.38 is the preferred method because it is more reliable and works well in most network conditions. If the service provider supports T.38, please use this method by selecting Fax mode to be T.38 (default). If the service provider does not support T.38, pass-through mode may be used. To send or receive faxes in fax pass through mode, users must select all the Preferred Codecs to be PCMU/PCMA (G.711-u/a).
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CALL FEATURES

Following table shows the call features (* code) of HT-386. TABLE 5: HT386 CALL FEATURE DEFINITIONS Key *23 *30 *31 *67 *82 *50 *51 *70 *71 *72 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. Disable Call Waiting (for all-config change) Enable Call Waiting (for all-config change) Disable Call Waiting (Per Call) Enable Call Waiting (Per Call) Unconditional Call Forward. To use this feature, dial *72, wait for the dial tone. Then dial the forward number ended with #, wait for dial tone, hang up. Cancel Unconditional Call Forward To cancel Unconditional Call Forward, dial *73 and get the dial tone, then hang up. Blind Transfer Refer to section above for procedure to perform Blind Transfer. Busy Call Forward To use this feature, dial *90, wait for the dial tone. Then dial the forward number ended with #, wait for dial tone, hang up. Cancel Busy Call Forward To cancel Busy Call Forward, dial *91 and get the dial tone, then hang up Delayed Call Forward To use this feature, dial *92, wait for the dial tone. Then dial the forward number ended with #, wait for dial tone, hang up. Cancel Delayed Call Forward To cancel this Forward, dial *93 and get the dial tone, then hang up Toggles between active call and incoming call (call waiting tone). If not in conversation, flash/hook will switch to a new channel for a new call.

*73 *87 *90

*91 *92

*93 Flash/Hook

END USER CONFIGURATION

Once the HTTP request is entered and sent from a Web browser, the user will see a log-in screen. There are two default passwords for the login page: User Level: End User Level Administrator Level Password: 123 admin Web pages allowed: Only Status and Basic Settings Browse all pages
Only an administrator can access the ADVANCED SETTING configuration page. Once this HTTP request is entered and sent from a Web browser, the HT-386 will respond with the following login screen:
FIGURE 3: SCREENSHOT OF CONFIGURATION LOG-IN PAGE
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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. After a correct password is entered in the login screen, the embedded Web server inside the HT-386 will respond with the Configuration pages which are explained in details below.
TABLE 7: HT-386 DEVICE STATUS PAGE DEFINITIONS MAC Address IP Address Product Model Software Version The device ID, in HEX format. This is very important ID for ISP troubleshooting. This field shows IP address of the HT-386. This field contains the product model info, such as HT-386. Program: This is the main software release. This number is always used for firmware upgrade. Current release is 1.0.3.64 Bootloader: current version is 1.1.0.1. HTML: current version 1.0.3.64. VOC: current version is 1.0.0.13 This shows system up time since last reboot. Whether the unit is registered to service providers server. This shows whether the PPPoE is up if connected to DSL modem This shows what kind NAT the HT-386 is connected to. It is based on STUN protocol. If the detected NAT is symmetric NAT, STUN will not work and Outbound Proxy needed to make HT-386 functioning correctly.
System Uptime Registered PPPoE Link Up NAT

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DHCP hostname DHCP domain DHCP vendor class ID Time Zone Daylight Savings Time

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.
ADVANCED CONFIGURATION AND FXS PORTS PARAMETERS
To login to the Advanced Setting and FXS port configuration pages, administrator password is required. The default administrator password is admin. User can change the administrator password here. The password is case sensitive and the maximum length is 25 characters.
TABLE 9: HT-386 ADVANCED SETTINGS PAGE DEFINITIONS Admin Password Administrator password. Only administrator can configure the Advanced Settings page. Password field is purposely blanked for security reason after clicking update and saved. The maximum password length is 25 characters. Local area code for North American Dial Plan. 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. Layer 2 QoS settings. Default setting is blank. Other VLAN supported equipments required if configured these settings. Default is 4 seconds. User can short or extend that depends on digits dialed IP address or Domain name of the STUN server. Default is 20 seconds. The interval of sending dummy UDP packet to keep NAT pin hole open. NAT IP address used in SIP/SDP message. Default is blank. Default method is HTTP. Firmware upgrade may take up to 10 minutes depending on network environment. Do not interrupt the firmware upgrading process. IP address or domain name of firmware server. IP address or domain name of configuration server. Default is blank. If configured, HT-386 will request the firmware file with the prefix. This setting is useful for ITSPs. End user should keep it blank. Default is blank. End user should keep it blank. Default is blank. End user should keep it blank. Default is blank. End user should keep it blank. Default is Yes. For firmware encryption. It should be 32 digit in Hexadecimal Representation. End user should keep it blank. Bellcore (North America) CID (Canada) DTMF (Brazil) DTMF (Sweden) DTMF (Denmark)

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Preferred Vocoder

Voice Frames per TX
G723 Rate: iLBC frame size: iLBC payload type:
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. Use only if proxy supports 484 response Early Dial Sets the prefix added to each dialed number Dial Plan Prefix This parameter allows the user to configure the # key to be used as the Use # as Send(or Dial) key. Once set to Yes, pressing this key will immediately Dial/Send Key trigger the sending of dialed string collected so far. In this case, this key is essentially equivalent to the (Re)Dial key. If set to No, this # key will then be included as part of the dial string to be sent out. Default is NO. When set to Yes a SUBSCRIBE for Message Waiting Indication Subscribe for MWI: will be sent periodically. If this parameter is set to Yes, user ID will be sent as anonymous, essentially Send Anonymous block the Caller ID from displaying. If set to Yes, the configuration update via keypad is disabled. Lock keypad update NOTE: Since only FXS1 has LED for indication and IVR for keypad access, this field is not applied to FXS2 Used for Attended transfer Feature. The Refer-To header uses the Refer-To Uses Target transferred targets Contact header information. Contact. Default is Standard. Choose the selection to meet some special requirements Special Feature from Soft Switch vendors like Lucent, Nortel, BroadSoft, etc. Handset volume adjustment. RX is for receiving volume, TX is for transmission Volume Amplification volume. Default values are 0dB for both parameters. +6dB generates the highest volume and -6dB generates the lowest volume. Note: The explanations provided apply to both FXS port configuration parameters
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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. Otherwise it will ring ON ms and a pause of OFF ms and then repeat the pattern.
SAVING THE CONFIGURATION CHANGES

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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. Locate a needle-sized hole on the back panel of the HT386 unit next to the power connection. 2. Insert a pin in this hole, and press for about 7 seconds. The back LEDs for LAN and WAN will be solid on to indicate the reset. 3. Take out the pin. All unit settings are restored to factory settings.
RESET VIA IVR o Find the MAC address of the device. It is a 12 digits HEX number located on the bottom of the unit. Encode the MAC address. Please use the following mapping: 0-9: 0-9 A: 22 B: 222 C: 2222 D: 33 E: 333 F: 3333 For example, if the MAC address is 000b8200e395, it should be encoded as 0002228200333395. To perform factory reset: Pick up the headset and dial *** for voice prompt. Enter 99 and get the voice prompt Reset. Enter the encoded MAC address of the device. Wait for 15 seconds. The device will reboot automatically and restore to factory default setting.
NOTE: 1. Factory Reset will be disabled if the Lock keypad update is set to Yes. 2. Please be aware by default the HT-502 WAN side HTTP access is disabled. 3. After a factory reset, the devices web configuration page can be accessed only from its LAN port.
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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. ARP Address Resolution Protocol is a protocol used by the Internet Protocol (IP) [RFC826], specifically IPv4, to map IP network addresses to the hardware addresses used by a data link protocol. The protocol operates below the network layer as a part of the interface between the OSI network and OSI link layer. It is used when IPv4 is used over Ethernet ATA Analogue Telephone Adapter. Covert analogue telephone to be used in data network for VoIP, like Grandstream HT series products. CODEC Abbreviation for Coder-Decoder. It is an analog-to-digital (A/D) and digital-to-analog (D/A) converter for translating the signals from the outside world to digital, and back again. CNG Comfort Noise Generator, generate artificial background noise used in radio and wireless communications to fill the silent time in a transmission resulting from voice activity detection. DATAGRAM A data packet carrying its own address information so it can be independently routed from its source to the destination computer DECIMATE To discard portions of a signal in order to reduce the amount of information to be encoded or compressed. Lossy compression algorithms ordinarily decimate while sub-sampling. DECT Digital Enhanced Cordless Telecommunications: A standard developed by the European Telecommunication Standard Institute from 1988, governing pan-European digital mobile telephony. DECT covers wireless PBXs, telepoint, residential cordless telephones, wireless access to the public switched telephone network, Closed User Groups (CUGs), Local Area Networks, and wireless local loop. The DECT Common Interface radio standard is a multi-carrier time division multiple access, time division duplex (MC-TDMA-TDD) radio transmission technique using ten radio frequency channels from 1880 to 1930 MHz, each divided into 24 time slots of 10ms, and twelve full-duplex accesses per carrier, for a total of 120 possible combinations. A DECT base station (an RFP, Radio Fixed Part) can transmit all 12 possible accesses (time slots) simultaneously by using different frequencies or using only one frequency. All signaling information is transmitted from the RFP within a multi-frame (16 frames). Voice signals are digitally encoded into a 32 Kbit/s signal using Adaptive Differential Pulse Code Modulation. DNS Short for Domain Name System (or Service or Server), an Internet service that translates domain names into IP addresses DID Direct Inward Dialing. The ability for an outside caller to dial to a PBX extension without going through an attendant or auto-attendant. DSP Digital Signal Processor. A specialized CPU used for digital signal processing. Grandstream products all have DSP chips built inside. DTMF Dual Tone Multi Frequency. The standard tone-pairs used on telephone terminals for dialing using in-band signaling. The standards define 16 tone-pairs (0-9, #, * and A-F) although most terminals support only 12 of them (0-9, * and #).

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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. An FXS interface will accept calls from FXS or PSTN interfaces. All countries and regions have their own standards. FXS is complimentary to FXS (and the PSTN). FXS Foreign eXchange Station. An FXS device has hardware to generate the ring signal to the FXS extension (usually an analog phone). An FXS device will allow any FXS device to operate as if it were connected to the phone company. This makes your PBX the POTS+PSTN for the phone. The FXS Interface connects to FXS devices (by an FXS interface, of course). DHCP The Dynamic Host Configuration Protocol (DHCP) is an Internet protocol for automating the configuration of computers that use TCP/IP. DHCP can be used to automatically assign IP addresses, to deliver TCP/IP stack configuration parameters such as the subnet mask and default router, and to provide other configuration information such as the addresses for printer, time and news servers. ECHO CANCELLATION Echo Cancellation is used in telephony to describe the process of removing echo from a voice communication in order to improve voice quality on a telephone call. In addition to improving quality, this process improves bandwidth savings achieved through silence suppression by preventing echo from traveling across a network. There are two types of echo of relevance in telephony: acoustic echo and hybrid echo. Speech compression techniques and digital processing delay often contribute to echo generation in telephone networks. H.323 A suite of standards for multimedia conferences on traditional packet-switched networks. HTTP Hyper Text Transfer Protocol; the World Wide Web protocol that performs the request and retrieve functions of a server IP Internet Protocol. A packet-based protocol for delivering data across networks. IP-PBX IP-based Private Branch Exchange IP Telephony (Internet Protocol telephony, also known as Voice over IP Telephony) A general term for the technologies that use the Internet Protocol's packet-switched connections to exchange voice, fax, and other forms of information that have traditionally been carried over the dedicated circuit-switched connections of the public switched telephone network (PSTN). The basic steps involved in originating an IP Telephony call are conversion of the analog voice signal to digital format and compression/translation of the signal into Internet protocol (IP) packets for transmission over the Internet or other packet-switched networks; the process is reversed at the receiving end. The terms IP Telephony and Internet Telephony are often used to mean the same; however, they are not 100 per cent interchangeable, since Internet is only a subcase of packet-switched networks. For users who have free or fixed-price Internet access, IP Telephony software essentially provides free telephone calls anywhere in the world. However, the challenge of IP Telephony is maintaining the quality of service expected by subscribers. Session border controllers resolve this issue by providing quality assurance comparable to legacy telephone systems.

FOR IMMEDIATE RELEASE

Wind Currents Communications Chooses the Grandstream GXV3000 Video Phone and HT386 Analog Telephone Adaptor for its Residential VoIP Offering
Wind Currents offers clear, reliable and affordable choices with Grandstream products and V2VIP calling plans.
San Francisco, CA VoiceCon, Moscone Convention Center, (August 21, 2007) Grandstream Networks, Inc. a leading designer and manufacturer of IP Voice and Video endpoints for broadband networks, and Wind Currents Communications, Inc. (www.v2vip.com) an Internet Phone company that provides videophone and IP telephony calling plans to the global residential market announced today that the GXV3000 IP Video Phone and the HT386 Analog Telephone Adaptor will be IP terminal endpoints for the V2VIP telephony and video end-to-end solutions. After extensive evaluation testing, were confident that both the performance and price of the HT386 and GXV3000 complement our V2VIP calling plans, said John Monahan, President of Wind Currents Communications (WCCI). We believe that the superior video quality and overall functionality, performance and price of the GXV3000 and the clear, reliable voice communication of the HT386 allow V2VIP customers to choose how to lower their toll/long distance bills and revolutionize their communication experience. Our V2VIP Basic PlanGXV3000 Bundle was developed and priced for TeleHealth markets, where service-delivery is expanded dramatically with voice-and-video patient communication. Grandstream and Wind Currents Communications are cooperating for optimum outcomes; so, were poised for success in two rapidly growing markets. V2VIP service is a SIP-based platform that can replace traditional telephone service while adding video to the call. Three calling plans are available with the GXV3000 to offer a variety of options--unlimited inbound/out-bound, local, long-distance (US, Canada) callingas well as unlimited, worldwide in-thenetwork voice-and-video calling. For camera-shy users, the V2VIP Voice plan offers the HT386 at no charge. All V2VIP plans offer numerous call features, such as: voice-mail, caller ID, call forward, 3-way, etc., and each assures savings of both time and money. The V2VIP Basic-GXV3000 Bundle allows home / rural care agencies to expand and enrich service delivery and a choice of payment options developed to off-set budgetary constraints. The Grandstream GXV3000 is an advanced IP video phone based on SIP and H.264/H.263 standards, and is competitively priced. The GXV3000 combines great design and technology with excellent audio and picture quality, ease-of-use and broad interoperability with 3rd party SIP products. The phone supports real-time (up to 30fps) high-quality video at very modest bandwidth levels (as low as 64kbps). The large 5.6 inch TFTP adjustable LCD screen and the VGA camera enable high-quality videoconferences with any other video device, anywhere, anytime. It is ideal for any multi-media communication environment. The HandyTone 386 Analog Telephone Adaptor is SIP-based, lightweight and compact, and works with any phone or fax, supports a broad range of codecs including G.711, G.729 A/B, iLBC, G.723.1, G.726 and T.38 fax, and provides a network port and dual FXS telephone ports. The HT386 offers caller ID, voicemail with LED indicator, 3-way conferencing, all of the popular telephony features and supports two independent SIP accounts or SIP server platforms and a PSTN pass through line for toggling operations between SIP and PSTN networks.

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We are excited about our partnership with Wind Currents Communications, said David Li, CEO of Grandstream Networks. This partnership confirms that the Grandstream IP video and ATA product portfolio is a best-in-class solution which enables innovative service providers such as Wind Currents to offer advanced IP multimedia communication services to both consumer and specialty professional markets. We look forward to a long and prosperous partnership with Wind Currents. About Grandstream Networks Grandstream Networks, Inc. is an award-winning designer and manufacturer of next generation IP voice & video products for broadband networks. Grandstreams products deliver superb sound and picture quality, rich telephony features, full compliance with industry standards, and broad interoperability with most service providers and 3rd party SIP based VoIP products. Grandstream is consistently recognized in the VoIP industry for their innovation, affordability and superior value in their products. Grandstream Networks is a private company headquartered in Brookline, MA with offices in Los Angeles, Dallas and China. For more information, please visit www.grandstream.com. About Wind Currents Communications Wind Currents Communications, Inc. (WCCI), a wholly-owned subsidiary of Wind Currents, Inc. (WCI), launched its Internet Phone Service calling plans (V2VIP) and products for residential customers and professional clients this year. Wind Currents, Inc. (WCI), d.b.a.: Wind Currents Technology, celebrated its 10th year of business in 2007, as it continues to serve governmental and institutional clients with highquality, affordable videophone solutions (POTS and IP) for business, TeleHealth, judicial, and border security applications. Wind Currents, Inc. is a member of the Chamber of Commerce of Ulster County, the Hudson Valley Center for Innovation, and the Minnesota Telehealth Association. Both companies are based in Woodstock, New York. For more information about V2VIP phone services, please visit www.v2vip.com. For information about Wind Currents Technology, visit www.videophoneconnection.com. Grandstream Networks Contact Marianne Rocco mrocco@grandstream.com (617) 566-9300 - ### Wind Currents Communications Contact John Monahan jpm@wccivoip.com (866) 435 0213