Grandstream Networks, Inc.
HT286 Analog Telephone Adaptor
HT286 User Manual Firmware Version 1.1.0.31
www.grandstream.com support@grandstream.com
TABLE OF CONTENTS HT-286 User Manual WELCOME..... 4 SAFETY COMPLIANCES.... 4 WARRANTY..... 4 INSTALLATION..... 5 EQUIPMENT PACKAGING.... 5 CONNECTING YOUR ATA..... 5 FIGURE 1: CONNECTING THE HT286... 5 THREE EASY STEPS TO INSTALL THE HT286... 6 FIGURE 2: HT286 CONNECTION DIAGRAM.... 6 PRODUCT OVERVIEW..... 7 KEY FEATURES..... 7 BASIC OPERATIONS..... 9 GET FAMILIAR WITH VOICE PROMPT.... 9 PLACING A PHONE CALL..... 10
Phone or Extension Numbers... 10 Direct IP Calls.... 11
CALL HOLD.... 11 CALL WAITING.... 11 CALL TRANSFER..... 11 3-WAY CONFERENCING.... 12 CALL FEATURES.... 13 T.38 FAX...... 13 LED LIGHT PATTERN INDICATION.... 13 CONFIGURATION GUIDE.... 15 CONFIGURING HT286 THROUGH VOICE PROMPT.... 15 CONFIGURING HT286 WITH WEB BROWSER... 15
Access the Web Configuration Menu... 15 End User Configuration... 15 Advanced User Configuration.... 18
SAVING THE CONFIGURATION CHANGES.... 25 REBOOTING THE HT286 FROM REMOTE.... 25 CONFIGURATION THROUGH A CENTRAL SERVER... 25 SOFTWARE UPGRADE..... 26 FIRMWARE UPGRADE THROUGH TFTP/HTTP... 26 CONFIGURATION FILE DOWNLOAD.... 27 FIRMWARE AND CONFIGURATION FILE PREFIX AND POSTFIX.. 27 MANAGING FIRMWARE AND CONFIGURATION FILE DOWNLOAD... 27 RESTORE FACTORY DEFAULT SETTING.... 28
TABLE OF FIGURES HT-286 User Manual
Figure 1: Connecting the HT286.... 5 Figure 2: HT286 Connection Diagram... 6
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TABLE OF TABLES HT-286 User Manual TABLE 1: DEFINITIONS OF THE HT286 CONNECTORS... 5 TABLE 2: HT286 TECHNICAL SPECIFICATIONS.... 7 TABLE 3: HT286 HARDWARE SPECIFICATIONS.... 8 TABLE 4: HT286 IVR MENU DEFINITIONS.... 9 TABLE 5: IVR ERROR REPORT..... 10 TABLE 6: HT286 CALL FEATURE DEFINITIONS... 13 TABLE 7: HT286 LED DEFINITIONS.... 14 TABLE 8: HT286 BASIC CONFIGURATION SETTINGS DEFINITIONS... 16 TABLE 9: HT286 DEVICE STATUS PAGE DEFINITIONS... 17 TABLE 10: HT286 ADVANCED CONFIGURATION PAGE DEFINITIONS.. 18
CONFIGURATION GUI INTERFACE EXAMPLES
HT286 User Manual (http://www.grandstream.com/user_manuals/GUI/GUI_HT286.rar) 1. 2. 3. 4. 5. 6. 7. SCREENSHOT OF LOGIN PAGE SCREENSHOT OF STATUS PAGE SCREENSHOT OF BASIC SETTINGS CONFIGURATION PAGE SCREENSHOT OF ADVANCED SETTING1 CONFIGURATION PAGE SCREENSHOT OF ADVANCED SETTING2 CONFIGURATION PAGE SCREENSHOT OF SAVED CONFIGURATION CHANGES SCREENSHOT OF REBOOT PAGE
HT286 User Manual Firmware 1.1.0.31
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Welcome

Congratulations on becoming an owner of HandyTone-286. You made an excellent choice and we hope you will enjoy all its capabilities. Grandstream's award-wining HandyTone-286 is innovative Analog Telephone Adaptor that offers a rich set of functionality and superb sound quality at ultra-affordable price. They are fully compatible with SIP industry standard and can interoperate with many other SIP compliant devices and software on the market.

SAFETY COMPLIANCES

The HT286 complies with FCC/CE and various safety standards. The HT286 power adaptor is compliant with UL standard. Only use the universal power adapter provided with the HT286 package. The manufacturers warranty does not cover damages to the phone caused by unsupported power adaptors.

WARRANTY

If you purchased your HT286 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: Please do not use a different power adaptor with the HT286 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_HT286.rar as 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/HandyTone.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 HT286 ATA package contains: One HT286 Main Case One Universal Power Adaptor One Ethernet Cable

CONNECTING YOUR ATA

The HT286 is easy to configure. HandyTone-286 is a VoIP Analog Telephone Adaptor designed to work with an ordinary analog telephone. The following photo illustrates the appearance of a HandyTone-286.
FIGURE 1: CONNECTING THE HT286
TABLE 1: DEFINITIONS OF THE HT286 CONNECTORS Power Cable LAN Port (RJ-45) PHONE BUTTON Power adapter connection Connect the LAN port with an Ethernet cable to your PC. FXS port to be connected to analog phones / fax machines. Button and two colors led indicator.
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THREE EASY STEPS TO INSTALL THE HT286
1. Connect a standard touch-tone analog telephone (or fax machine) to FXS port. 2. Insert the Ethernet cable into the LAN port of HT286 and connect the other end of the Ethernet cable to an uplink port (a router or a modem, etc.) 3. Insert the power adapter into the HT286 and connect it to a wall outlet.

FIGURE 2: HT286 CONNECTION DIAGRAM
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Product Overview

HT-286 supports one FXS port for Internet data, voice, and fax.

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

TABLE 2: HT286 TECHNICAL SPECIFICATIONS Lines/SIP Accounts Protocol Support Feature Keys LAN Interface Device Management 1 lines / 1 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 (on Rev 2.0) 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), Line Echo Cancellation( G.168) 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(on Rev 2.0) Manual or dynamic host configuration protocol (DHCP) network setup; RTP
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DHCP Server/Client Audio Features
Call Handling Features Network and

Provisioning Fax over IP

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

Security Physical Design

TABLE 3: HT286 HARDWARE SPECIFICATIONS LAN Interface FXS phone port Button LED Universal Switching Power Adaptor Dimension Weight Temperature Humidity Compliance 1 x RJMbps 1 x FXS 1 Green and red / solid state & blinking state Input: 100-240VAC 50-60 Hz Output: +5VDC, 1200mA UL certified 65mm (W) x 93mm (D) x 27mm (H) 0.57 lbs (0.26kg) 32 - 104oF / 0 - 40oC 10% - 90% (non-condensing) FCC/CE/C-Tick
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NOTE: 1. Once the button is pressed, it enters 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 Calling option and a dial tone is prompted 2. * 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 3. 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. 4. Key entry can not be deleted but the phone may prompt error once it is detected

PLACING A PHONE CALL

PHONE OR EXTENSION NUMBERS 1. Dial the number directly and wait for 4 seconds (Default No Key Entry Timeout); or 2. Dial the number directly and press # (Use # as dial key must be configured in web configuration). Examples: 1. Dial an extension directly on the same proxy, (e.g. 1008), and then press the # or wait for 4 seconds. 2. Dial an outside number (e.g. (626) 666-7890), first enter the prefix number (usually 1+ or international code) followed by the phone number. Press # or wait for 4 seconds. Check with your VoIP service provider for further details on prefix numbers.
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DIRECT IP CALLS Direct IP calling allows two parties with VoIP devices (e.g. a HT286and another VoIP Device), to talk to each other in an ad hoc fashion without a SIP proxy. Direct IP calls are possible if: a) Both VoIP devices use a public IP address; or b) Both VoIP devices are on the same LAN using private IP addresses, or c) Both VoIP devices can be connected through a router using public or private IP addresses (with necessary port forwarding or DMZ). To make a Direct IP Call, use either the handset or the speakerphone. Access the IVR using ***, the button on the HT286or 47. After dial tone, enter a 12-digit target IP address to place a call. Destination ports can be specified by using *4 (encoding for :) followed by the port number. Examples: a) If the target IP address is 192.168.0.160, enter the 12 digit IP address (e.g. 1921680160) after the voice prompt followed by the # key or wait for 4 seconds. The default destination port 5060 is used if no port is specified. b) If the target IP address/port is 192.168.1.20:5062, enter the 12 digit IP address + *45062 (e.g. 1921680160*45062) after the voice prompt followed by the # key or wait for 4 seconds.

CALL HOLD

Press the flash button (or hook flash on older models) to place a caller on hold. Press the flash button again to release call hold.

CALL WAITING

If call waiting feature is enabled, call waiting tone (3 short beeps) indicates an incoming call. 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

The HT286 supports both blind and attended transfer: 1. Blind Transfer: Press flash button (or hook flash on older models), dial *87, then dial the number to transfer call and press the # key (or wait 4 seconds) to complete transfer of active call. Expected outcomes: a) A quick confirmation tone (call waiting tone) followed by a dial tone. This indicates the transfer is successful. Hang up or place another call. b) A quick busy tone followed by a restored call (on supported platforms only). The busy tone indicates the transfer failed. c) Continuous busy tone. The phone call has timed out. Note: continuous busy tone does not indicate the transfer has been successful, nor does it indicate the transfer has failed. It often means there was a failure to receive second NOTIFY check firmware for most recent release. NOTE: Enable Call Feature must be set to Yes in web configuration page.
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2. Attended Transfer:

Assuming that call party A and B are in conversation. A wants to Attend Transfer B to C:
(1) A presses FLASH (on the analog phone, or Hook Flash for old model phones) to get a dial tone (2) A then dial Cs number then # (or wait for 4 seconds). (3) If C answers the call, A and C are in conversation. Then A can hang up to complete transfer. (4) If C does not answer the call, A can press flash back to talk to B. NOTE: If Attended Transfer fails, and party A hangs up, the HandTone-496 will ring party A to remind A that party B is still on the line. Party A can pick up the phone to resume conversation with party B.

3-WAY CONFERENCING

HT286 supports Star Code Style or Bellcore Style 3-way Conference. Star Code Style 3-way Conference Assuming that call party A and B are in conversation. A (HT-286) wants to bring C in a conference: 1. A presses FLASH (on the analog phone, or Hook Flash for old model phones) to get a dial tone. 2. A dials *23 then Cs number then # (or wait for 4 seconds). 3. If C answers the call, then A presses FLASH to bring B, C in the conference. 4. If C does not answer the call, A can press FLASH back to talk to B. 5. If A presses FLASH during conference, C will be dropped out. Bellcore Style 3-way Conference Bellcore style 3-way conference is also supported. To do this, user needs to enable Use Bell-style 3-way Conference in ADVANCED SETTINGS web configuration. Assuming that call party A and B are in conversation. A (HT-286) wants to bring C in a conference: 1. A presses FLASH (on the analog phone, or Hook Flash for old model phones) to get a dial tone. 2. A dials Cs number then # (or wait for 4 seconds). 3. If C answers the call, then A presses FLASH to bring B, C in the conference. 4. If C does not answer the call, A can press FLASH back to talk to B. 5. If A presses FLASH during conference, C will be dropped out.

GREEN LED indicates normal status Button flashes every 2 seconds. Button flashes at 1/10 second. Button flashes every second. Green light steady. Message Waiting Indication RINGING RINGING INTERVAL In Conversation
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Configuration Guide

CONFIGURING HT286 THROUGH VOICE PROMPT
DHCP Mode Select voice menu option 01 to enable HT286 to use DHCP. STATIC IP Mode Select voice menu option 01 to enable HT286 to use STATIC IP mode, then use option 02, 03, 04, 05 to set up IP address, Subnet Mask, Gateway and DNS server respectively. 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 voice menu option 15 to choose firmware and configuration upgrade protocol. User can choose between TFTP and HTTP. Firmware Upgrade Mode Select voice menu option 17 to choose firmware upgrade mode among the following three options: 1) always check, 2) check when pre/suffix changes, and 3) never upgrade
CONFIGURING HT286 WITH WEB BROWSER
HT286 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 HT286 through a Web browser such as Microsofts IE and AOLs Netscape. ACCESS THE WEB CONFIGURATION MENU HandyTone-286 has embedded HTML pages that allows a user to configure the HandyTone-286 through a Web browser. 1. Find the IP address of the HT286 using voice prompt menu option 02. 2. Access the HT286 Web Configuration page by the following URI via WAN port: http://HandyTone-IP-Address (the HandyTone-IP-Address is the IP address for the HT286). NOTE: If using a web browser to enter the configuration page, strip the leading 0s because the browser will parse in octet. (i.e. if the IP address is: 192.168.001.014, please type in: 192.168.1.14).
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
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Only an administrator can access the ADVANCED SETTING configuration page.
NOTE: If you cannot log into the configuration page by using default password, please check with the VoIP service provider. The service provider may have provisioned and configured the device for you. The Basic Configuration Page is the first web GUI the user will see.
TABLE 8: HT286 BASIC CONFIGURATION SETTINGS DEFINITIONS End User Password Web Port IP Address Password to access the Web Configuration Menu. This field is case sensitive with a maximum length of 25 characters. By default, HTTP uses port 80. This field is for customizable web port. There are two modes to operate the HT286: DHCP mode: all the field values for the Static IP mode are not used (even though they are still saved in the Flash memory.) The HT286 acquires its IP address from the first DHCP server it discovers from the LAN it is connected. Using the PPPoE feature: set the PPPoE account settings. The HT286 will establish a PPPoE session if any of the PPPoE fields is set. Static IP mode: configure the IP address, Subnet Mask, Default Router IP address, DNS Server 1 (primary), DNS Server 2 (secondary) fields. These fields are set to zero by default. This option specifies the name of the client. This field is optional but may be required by some Internet Service Providers. Default is blank. This option specifies the domain name that client should use when resolving hostnames via the Domain Name System. Default is blank. Used by clients and servers to exchange vendor-specific information. Default is blank. PPPoE username. Necessary if ISP requires you to use a PPPoE (Point to Point Protocol over Ethernet) connection. PPPoE account password. This field is optional. If your ISP uses a service name for the PPPoE connection, enter the service name here. Default is blank. Controls how the date/time displays according to the specified time zone.

DHCP hostname DHCP domain DHCP vendor class ID PPPoE account ID PPPoE password PPPoE Service Name Time Zone
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Daylight Savings Time

Controls whether displayed time is daylight savings time or not. If set to Yes and the Optional Rule is empty, then the displayed time will be 1 hour ahead of normal time. The Automatic Daylight Saving Time Rule has the following syntax: o start-time;end-time;saving. Both start-time and end-time have the same syntax: month,day,weekday,hour,minute o month: 1,2,3,.,12 (for Jan, Feb,., Dec) o day: [+|-]1,2,3,.,31 o weekday: 1, 2, 3,., 7 (for Mon, Tue,., Sun), or 0 indicating daylight savings rule is based on the day of the month. o hour: hour (0-23), o minute: minute (0-59) If weekday is 0, it means the date to start or end daylight savings is the given date. The day value must not be negative. If weekday is not zero and day is positive, then daylight saving starts on the first day of the week (1st Sunday, 3rd Tuesday etc). If weekday is not zero and day is negative, then daylight savings starts on the last day of the week (last Sunday, 3rd last Tuesday etc). Daylight savings is in minutes. If preceded by a (-), then subtract the number of minutes. The default value for Automatic Daylight Saving Time Rule is US time: 04,01,7,02,00;10,-1,7,02,00;60. Example: US/Canada, where daylight savings is applicable: 04,01,7,02,00;10,-1,7,02,00;60 Daylight savings starts from the first Sunday of April at 2AM and ends the last Sunday of October at 2AM.).
In addition to the Basic Settings configuration page, end users also have access to the Device Status page.
TABLE 9: HT286 DEVICE STATUS PAGE DEFINITIONS MAC Address WAN IP Address Product Model Software Version System Up Time Registered PPPoE Link Up NAT The device ID in HEX format. This is needed for ISP troubleshooting. Shows WAN IP address of HT286. Contains the product model info. Program: This is the main software release. Boot and Loader are seldom changed. Shows system up time since the last reboot. Indicates whether the HT286 is registered to the service providers server. Indicates whether the PPPoE connection is up if the HT286 is connected to DSL modem. Indicates the type of NAT the HT286 is connected to via its WAN port. Based on STUN protocol.
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NAT Mapped IP NAT Mapped Port Statistical Status
WAN side mapped IP if HandyTone-286 is connected to a NAT router. WAN side mapped port if HandyTone-286 is connected to a NAT router. Self-explainable. Please refer to the page displayed.
ADVANCED USER CONFIGURATION Log in to the advanced user configuration page the same way as for the basic configuration page. The password is case sensitive and the factory default password for Advanced User is admin. Advanced User configuration includes the end user configuration and the advanced configurations including: a) SIP configuration, b) Codec selection, c) NAT Traversal Setting and d) other miscellaneous configuration. TABLE 10: HT286 ADVANCED CONFIGURATION PAGE DEFINITIONS Admin Password Administrator password. Only administrator can configure the Advanced Settings page. Password field is purposely left blank for security reason after clicking update and saved. The maximum password length is 25 characters. This field contains the URI string or the IP address. e.g. sip.my-voipprovider.com; 192.168.1.200:5066 This field contains the URI string or the IP of the outbound proxy. If there is no outbound proxy, this field SHOULD be left blank. If it is not blank, all outgoing requests will be sent to this outbound proxy. This field contains the user part of the SIP address for this phone. e.g., if the SIP address is: sip:my_user_id@my_provider.com, then the SIP User ID is: my_user_id. Please do NOT include the preceding sip: scheme or the host portion of the SIP address in this field. It is given by VoIP service provider. SIP service subscribers Authenticate ID used for authentication. It can be identical to or different from SIP User ID and given by VoIP service provider. SIP service subscribers account password. It is given by VoIP service provider. SIP service subscribers name which will be used for Caller ID display. Local area code for North American Dial Plan. HandyTone-286 supports up to 7 different vocoder types including G711ulaw (PCMU), G711-alaw (PCMA), G723, G729A, G726-32, and iLBC. Depending on the product model, some of these vocoders may not be provided in standard release. A user can configure vocoders in a preference order that will be offered in SIP INVITE message. This defines the encoding rate for G723 vocoder. By default, 6.3kbps rate is chosen.

Local SIP port Local RTP port

Use Random Port

SIP Registration Failure Retry Wait Time
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NAT Traversal

This parameter defines whether the phone NAT traversal mechanism will be activated or not. If activated (by choosing Yes) and a STUN server is also specified, then the phone will behave according to the STUN client specification. Under this mode, the embedded STUN client inside the phone will attempt to detect if and what type of firewall/NAT it is behind by sending appropriate request to the specified STUN server. If this field is set to Yes with no specified STUN server, then the phone will only periodically (every 20 seconds by default) send a blank UDP packet (with no payload data) to the SIP server to keep the mapped port open on the NAT. The HandyTone-286 sends a UDP package to the SIP server periodically in order to keep the port open on the router. This parameter defines the interval time that HT286 send the UDP package. The default setting is 20 second. NAT IP address is used in SIP/SDP message. Default is blank.
Use STUN keep-alive to detect WAN side network problems. If keep-alive request does not yield any response for configured number of times, the device will restart the TCP/IP stack. If the STUN server does not respond when the device boots up, the feature is disabled.

keep-alive interval

Use NAT IP Use STUN keep-alive to detect networks connectivity Proxy-Require SUBSCRIBE for MWI Off hook Auto-Dial
SIP Extension to notify SIP server that the unit is behind the NAT/Firewall. Default is No. When set to Yes a SUBSCRIBE for Message Waiting Indication will be sent periodically This parameter allows the user to configure a User ID or extension number to be automatically dialed upon off hook. Please note that only the user part of a SIP address needs to be entered here. The phone will automatically append the @ and the host portion of the corresponding SIP address. Default is Yes. If set to Yes, call features using star codes are supported locally. Conference mode, default option is No. If set to yes, the feature code for coference *23 would be disabled. Default is No. Default is No. This parameter specifies the mechanism to transmit DTMF digit. There are 3 modes supported: in audio which means DTMF is combined in audio signal (not very reliable with low-bit-rate codec), via RTP (RFC2833), or via SIP INFO. This parameter sets the payload type for DTMF using RFC2833 Default is No. If set to yes, flash will be sent as DTMF event. The amount of time the hookflash is pressed that will cause the device to on hook. Default is 800ms. Selects the impedance of the analog telephone connected to the Phone port.
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Enable Call Feature Use Bell-style 3-way conference Disable Call Waiting Disable Call-Waiting Caller-ID Send DTMF

DTMF Payload Type Send Flash Event Onhook Threshold FXS Impedance

Caller ID Scheme

Select the Caller ID Scheme to suit the standard of different area. Bellcore (North America) CID (Canada) DTMF (Brazil) DTMF (Denmark) DTMF (Sweden) ETSI-FSK (France, Germany, Norway, Taiwan, UK-CCA) ETSI-DTMF (Finland, Sweden)

Onhook Voltage

The onhook voltage can be selected according to the line voltage depending on the analog phone used. The low power/high power will increase/decrease the output current. Selecting "low power" will make loop current limit = 20mA, selecting "high power" make loop current limit = 32mA. The default selection is 36V (High Power). Select Polarity Reversal to adapt some call charge/billing system. Default is No. This parameter defines the URI or IP address of the NTP server which the IP phone will use to display the current date/time. If this parameter is set to Yes, the device is employing the mechanism to block its ID. If it is set to Use from header. Callers SIP user ID will be sent as anonymous, essentially block the Caller ID from displaying. If it is set to User privacy header, the SIP INVITE message contains a privacy header, and the server blocks the caller ID from the called party. This setting allows user to adjust the ring time of the phone. Default is 60 seconds. Default is Standard. Choose the selection to meet some special requirements from Soft Switch vendors like Nortel, Broadsoft, CBCOM etc. SIP, RT(C)P and T.38 modes, 1.0 and 1.1 Key to be used by CBCOM The IP address or URL of System log server. This feature is especially useful for ITSP (Internet Telephone Service Provider)
Polarity Reversal NTP server Send Anonymous Anonymous Method
Time to ring Special Features CBCOM Encode CBCOM Encoder 1.1 Key: Syslog Server
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Syslog Level

Select the ATA to report the log level. Default is NONE. The level is one of DEBUG, INFO, WARNING or ERROR. Syslog messages are sent based on the following events: product model/version on boot up (INFO level) NAT related info (INFO level) sent or received SIP message (DEBUG level) SIP message summary (INFO level) inbound and outbound calls (INFO level) registration status change (INFO level) negotiated codec (INFO level) Ethernet link up (INFO level) SLIC chip exception (WARNING and ERROR levels) memory exception (ERROR level) The Syslog uses USER facility. In addition to standard Syslog payload, it contains the following components: GS_LOG: [device MAC address][error code] error message Here is an example: May 19 02:40:38 192.168.1.14 GS_LOG: [00:0b:82:00:a1:be][000] Ethernet link is up

Session Expiration

Grandstream implemented SIP Session Timer. The session timer extension enables SIP sessions to be periodically refreshed via a re-INVITE request. Once the session interval expires, if there is no refresh via a re-INVITE message, the session will be terminated. Session Expiration is the time (in seconds) at which the session is considered timed out, if no successful session refresh transaction occurs beforehand. The default value is 180 seconds.

Min-SE Caller Request Timer
The minimum session expiration (in seconds). The default value is 90 seconds. If selecting Yes the phone will use session timer when it makes outbound calls if remote party supports session timer.
Callee Request Timer If selecting Yes the phone will use session timer when it receives inbound calls with session timer request. Force Timer If selecting Yes the phone will use session timer even if the remote party does not support this feature. Selecting No will allow the phone to enable session timer only when the remote party support this feature. To turn off Session Timer, select No for Caller Request Timer, Callee Request Timer, and Force Timer. As a Caller, select UAC to use the phone as the refresher, or UAS to use the Callee or proxy server as the refresher. As a Callee, select UAC to use caller or proxy server as the refresher, or UAS to use the phone as the refresher. Session Timer can be refreshed using INVITE method or UPDATE method. Select Yes to use INVITE method to refresh the session timer.
UAC Specify Refresher UAS Specify Refresher Force INVITE
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Firmware Upgrade and Provisioning
Default HTTP. Firmware upgrading may take up to 10 minutes depends on network environment. Do not interrupt the firmware upgrading process.
Firmware Server Path IP address or domain name of firmware server. Config Server Path Firmware File Prefix Firmware File Postfix Config File Prefix Config File Postfix Automatic Upgrade IP address or domain name of configuration server Default blank. If it is configured, HT286 rev. 3.0 will request the firmware file with the prefix. 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. Choose Yes to enable automatic upgrade and provisioning and input the number, in minutes, you want the HT to check for an update. When set to No, HT286 will only do upgrade once at boot up. Always check for New Firmware at Boot up will check for new firmware every time the device reboots. Check New Firmware only when F/W pre/suffix changes will check for updates only when the pre/suffix has been changed. 32 digit in Hexadecimal Representation. Useful for ITSP to encrypt firmware. End user should keep it blank. Default No. End user should use default setting. If this parameter is set to Yes, except for IVR MENU items 1 to 5, the configuration update via keypad is disabled. If set to Yes, these four fields: SIP User ID, Authenticate ID, Authenticate Password and Name will be included in Basic Settings configuration page. Override the MTU size.

Firmware Key Authenticate Conf File Lock keypad update Allow conf SIP Account in Basic Settings Override MTU size
Volume Amplification Handset volume adjustment. RX is for receiving volume, TX is for transmission volume. Default values are 0dB for both parameters. +6dB generates the highest volume and -6dB generates the lowest volume. Powerline Ring Tone Call Progress Tones This setting allows user to configure the ringing frequencies and cadences. Using these settings, users can configure various call progress tone frequencies and cadences according to their country standard. By default they are set to North American standard. 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. Up to three cadences are supported. Default is No. If set to Yes, echo canceller is not used.
Disable Line Echo Canceller (LEC):
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SAVING THE CONFIGURATION CHANGES
After making a change, click the Update button in the Configuration page. The HT286 will display a screen to confirming changes. Reboot or power cycle the HT286 to enable the changes.
REBOOTING THE HT286 FROM REMOTE
The administrator of the HT286 can remotely reboot the HT286 by clicking the Reboot button at the bottom of the configuration page. When finished, re-login to the HT286 after waiting about 30 seconds.
CONFIGURATION THROUGH A CENTRAL SERVER
Grandstream HandyTone ATA can be automatically configured from a central provisioning system. When HandyTone ATA boot up, it will send TFTP or HTTP request to download configuration file cfg000b82xxxxxx, where 000b82xxxxxx is the MAC address of the HandyTone ATA. The configuration file can be loaded into devices via TFTP or HTTP from the central provisioning server, so the service provider or an enterprise with large deployment of HandyTone ATAs can easily manage the configuration and service provision to individual devices remotely. Grandstream has a provisioning system called GAPS (Grandstream Automated Provisioning System) that is used to support automated configuration of Grandstream devices. GAPS uses enhanced (NAT friendly) TFTP or HTTP (thus no NAT issues) and other communication protocols to communicate with each individual Grandstream device for firmware upgrade, remote reboot, etc. Grandstream provides GAPS service to VoIP service providers. Use GAPS for either simple redirection or with certain special provisioning settings. At boot-up, Grandstream devices by default point to Grandstream provisioning server GAPS, based on the unique MAC address of each device, GAPS provision the devices with redirection settings so that they will be redirected to customers TFTP or HTTP/HTTPS server for further provisioning. Grandstream also provide GAPSLite software package which contains our NAT friendly TFTP server and a configuration tool to facilitate the task of generating device configuration files. The GAPSLite configuration tool is now free to end users. The tool and configuration template are available for download from http://www.grandstream.com/configurationtool.html

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Software Upgrade

Software upgrade can be done via either TFTP or HTTP. The corresponding configuration settings are in the ADVANCED SETTINGS configuration page.
FIRMWARE UPGRADE THROUGH TFTP/HTTP
To upgrade via TFTP or HTTP, the Firmware Upgrade and Provisioning upgrade via field needs to be set to TFTP or HTTP, respectively. Firmware Server Path needs to be set to a valid URL of a TFTP or HTTP server, server name can be in either FQDN or IP address format. Here are examples of some valid URL. e.g. firmware.mycompany.com:6688/Grandstream/1.1.0.31 e.g. 168.75.215.189 NOTES: Firmware upgrade server in IP address format can be configured via IVR. Please refer to the CONFIGURATION GUIDE section for instructions. If the server is in FQDN format, it must be set via the web configuration interface. Grandstream recommends end-user use the Grandstream TFTP server. Its address can be found at http://www.grandstream.com/firmware.html. Currently the TFTP firmware server IP address is 168.75.215.189. For large companies, we recommend to maintain their own TFTP/ HTTP/HTTPS server for upgrade and provisioning procedures. Once a Firmware Server Path is set, user needs to update the settings and reboot the device. If the configured firmware server is found and a new code image is available, the HT ATA will attempt to retrieve the new image files by downloading them into the HT ATAs SRAM. During this stage, the HT ATAs LEDs will blink until the checking/downloading process is completed. Upon verification of checksum, the new code image will then be saved into the Flash. If TFTP/HTTP fails for any reason (e.g., TFTP/HTTP server is not responding, there are no code image files available for upgrade, or checksum test fails, etc), the HT ATA will stop the TFTP/HTTP process and simply boot using the existing code image in the flash. Firmware upgrade may take as long as 1 to 20 minutes over Internet, or just 20+ seconds if it is performed on a LAN. It is recommended to conduct firmware upgrade in a controlled LAN environment if possible. For users who do not have a local firmware upgrade server, Grandstream provides a NAT-friendly TFTP server on the public Internet for firmware upgrade. Please check the Services section of Grandstreams Web site to obtain our public TFTP servers IP address. Grandstreams latest firmware is available http://www.grandstream.com/firmware.html. Oversea users are strongly recommended to download the binary files and upgrade firmware locally in a controlled LAN environment. Alternatively, user can download a free TFTP or HTTP server and conduct local firmware upgrade. A free windows version TFTP server is available for download from http://support.solarwinds.net/updates/New-customerFree.cfm.

Instructions to download a free TFTP Server: 1. Unzip the file and put all of them under the root directory of the TFTP server. 2. Put the PC running the TFTP server and the HT286 device in the same LAN segment. 3. Please 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 phones web configuration page 5. Configure the Firmware Server Path with the IP address of the PC 6. Update the change and reboot the unit
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End users can also choose to download the free HTTP server from http://httpd.apache.org/ or use Microsoft IIS web server.
CONFIGURATION FILE DOWNLOAD
Grandstream SIP Device can be configured via Web Interface as well as via Configuration File through TFTP or HTTP. Config Server Path is the TFTP or HTTP server path for configuration file. It needs to be set to a valid URL, either in FQDN or IP address format. The Config Server Path can be same or different from the Firmware Server Path. A configuration parameter is associated with each particular field in the web configuration page. A parameter consists of a Capital letter P and 2 to 3 (Could be extended to 4 in the future) digit numeric numbers. i.e., P2 is associated with Admin Password in the ADVANCED SETTINGS page. For a detailed parameter list, please refer to the corresponding firmware release configuration template. When Grandstream Device boots up or reboots, it will issue request for configuration file named cfgxxxxxxxxxxxx, where xxxxxxxxxxxx is the MAC address of the device, i.e., cfg000b820102ab. The configuration file name should be in lower cases.
FIRMWARE AND CONFIGURATION FILE PREFIX AND POSTFIX
Firmware Prefix and Postfix allows device to download the firmware name with the matching Prefix and Postfix. This makes it the possible to store ALL of the firmware with different version in one single directory. Similarly, Config File Prefix and Postfix allows device to download the configuration file with the matching Prefix and Postfix. Thus multiple configuration files for the same device can be stored in one directory. In addition, when the field Check New Firmware only when F/W pre/suffix changes is set to Yes, the device will only issue firmware upgrade request if there are changes in the firmware Prefix or Postfix.
MANAGING FIRMWARE AND CONFIGURATION FILE DOWNLOAD
When Automatic Upgrade is set to Yes, Service Provider can use P193 (Auto Check Interval, in minutes, default and minimum is 60 minutes) to have the devices periodically check with either Firmware Server or Config Server, whenever they are defined. This allows the device periodically check if there are any new changes need to be taken on a scheduled time. By defining different intervals in P193 for different devices, Server Provider can spread the Firmware or Configuration File download in minutes to reduce the Firmware or Provisioning Server load at any given time.

User Manual

HandyTone-486

Analog Telephone Adaptor

For Firmware Version 1.1.0.31
Grandstream Networks, Inc.

www.grandstream.com

HandyTone-486 User Manual

Table of Contents

3 WELCOME.... 4 INSTALLATION... 5 WHAT IS INCLUDED IN THE PACKAGE... 7 3.1 3.4.1 4.SAFETY COMPLIANCES... 7 WARRANTY..... 7 KEY FEATURES... 8 HARDWARE SPECIFICATION... 9

PRODUCT OVERVIEW.... 8

BASIC OPERATIONS.... 10 5.1 GET FAMILIAR WITH KEY PAD AND VOICE PROMPT.. 10 5.2 MAKE PHONE CALLS.... 12 5.2.1 Calling phone or extension numbers... 12 5.2.2 Direct IP calls.... 12 5.2.3 Call Hold.... 13 5.2.4 Call Waiting... 13 5.2.5 Call Transfer.... 13
5.2.5.1 5.2.5.2 Blind Transfer..... 13 Attended Transfer..... 13

5.2.6.1 5.2.6.2

3-way Conferencing... 14
Star Code Style 3-way Conference.... 14 Bellcore Style 3-way Conference.... 14
5.2.7 PSTN Pass Through/Life line... 14 5.3 CALL FEATURES.... 15 5.3.1 Call Features Table (star code)... 15 5.4 FAX.... 16 5.5 LED LIGHT PATTERN INDICATION... CONFIGURATION GUIDE... 17 6.1 CONFIGURING HANDYTONE-486 THROUGH VOICE PROMPT.. 17 6.1.1 DHCP Mode.... 17 6.1.2 STATIC IP Mode... 17 6.1.3 Firmware Server IP Address.... 17 6.1.4 Configuration Server IP Address... 17 6.1.5 Upgrade Protocol.... 17 6.1.6 Firmware Upgrade Mode.... 17 6.1.7 WAN Port Web Access.... 17 6.2 CONFIGURING HANDYTONE-486 WITH WEB BROWSER.. 18 6.2.1 Access the Web Configuration Menu... 18 2
6.2.2 End User Configuration... 18 6.2.3 Advanced User Configuration... 22 6.2.4 Saving the Configuration Changes... 29 6.2.5 Rebooting the HandyTone-486 from remote... 29 6.3 CONFIGURATION THROUGH A CENTRAL SERVER.. SOFTWARE UPGRADE... 31 7.1 7.2 7.3 7.FIRMWARE UPGRADE THROUGH TFTP/HTTP... 31 CONFIGURATION FILE DOWNLOAD... 32 FIRMWARE AND CONFIGURATION FILE PREFIX AND POSTFIX. 32 MANAGING FIRMWARE AND CONFIGURATION FILE DOWNLOAD.. 32
RESTORE FACTORY DEFAULT SETTING... 34

Welcome

Congratulations on becoming an owner of HandyTone-486. You made an excellent choice and we hope you will enjoy all its capabilities. Grandstream's award-wining HandyTone-486 is an all-in-one VoIP integrated access device that features superb audio quality, rich functionalities, high level of integration, compactness and ultraaffordability. The HandyTone-486 is fully compatible with SIP industry standard and can interoperate with many other SIP compliant devices and software on the market. Grandstream HandyTone-486 has been awarded the Best of Show product in 2004 Internet Telephony Conference and Expo. This document is subject to changes without notice. The latest electronic version of this user manual can be downloaded from the following location: http://www.grandstream.com/user_manuals/HandyTone-486UserManual.pdf

Installation

HandyTone-486 Analog Telephone Adaptor is an all-in-one VoIP integrated device designed to be a total solution for networks providing VoIP services. The HandyTone-486 VoIP functionalities are available via a regular analog telephone. The following photo illustrates the appearance of a HandyTone-486.

Top View

Side Views
RJ11 Telephone (Phone) RJ45 10M Ethernet LAN - WAN

+5V/1200mA

BUTTON RED LED GREEN LED

RJ11 Phone Line (Line)

Interconnection Diagram of the HandyTone-486:
Internet ADSL/Cable Modem Ethernet Analog Phone

Cordless Phone

Following are the steps to install a HandyTone-486: 1. Connect a standard touch-tone analog telephone (or fax machine) to PHONE port. 2. Connect a PSTN telephone line to LINE port (optional). 3. Insert the Ethernet cable into the WAN port of HandyTone-486 and connect the other end of the Ethernet cable to an uplink port (a router or a modem, etc.) 4. Connect a PC to the LAN port of HandyTone-486. 5. Insert the power adapter into the HandyTone-486 and connect it to a wall outlet. Please follow the instructions in section 6.2.1 to configure the HandyTone-486.

What is Included in the Package
The HandyTone-486 package contains: 1) One HandyTone-486 2) One universal power adaptor 3) One Ethernet cable

Safety Compliances

The HandyTone-486 is compliant with various safety standards including FCC/CE and C-tick. Its power adaptor is compliant with UL standard. The HandyTone-486 should only operate with the universal power adaptor provided in the package.

Warranty

Grandstream has a reseller agreement with our reseller customer. End users should contact the company from whom you purchased the product for replacement, repair or refund. If you purchased the product directly from Grandstream, contact your Grandstream Sales and Service Representative for a RMA (Return Materials Authorization) number. Grandstream reserves the right to remedy warranty policy without prior notification.
Warning: Please do not attempt to use a different power adaptor. Using other power adaptor may damage the HandyTone-486 and will void the manufacturer warranty.
Caution: Changes or modifications to this product not expressly approved by Grandstream, or operation of this product in any way other than as detailed by this User Manual, could void your manufacturer warranty.
Information in this document is subject to change without notice. No part of this document may be reproduced or transmitted in any form or by any means, electronic or mechanical, for any purpose without the express written permission of Grandstream Networks, Inc.

Product Overview

Key Features

Supports SIP 2.0(RFC 3261), TCP/UDP/IP, RTP/RTCP, HTTP, ICMP, ARP/RARP, DNS, DHCP (both client and server), NTP, PPPoE, STUN, TFTP, etc. Built-in router, NAT, Gateway and DMZ port forwarding. Can also be configured to function as a two Ethernet ports bridge (NAT function is disabled) Device bridge mode support Powerful digital signal processing (DSP) to ensure superb audio quality; advanced adaptive jitter control and packet loss concealment technology Support various codecs including G.711 (PCM a-law and u-law), G.723.1 (5.3K/6.3K), G.726 (32K), as well as G.729A and iLBC Support Caller ID/name display or block, Call waiting caller ID, Hold, Call Waiting/Flash, Call Transfer, 3-way conference (on Rev. 2.0), Call Forward, in-band and out-of-band DTMF, etc. Support fax pass through (via PCMU or PCMA) and T.38 FoIP (Fax over IP) Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control) Support standard encryption and authentication (DIGEST using MD5 and MD5-sess) Support for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS) Support automated NAT traversal without manual manipulation of firewall/NAT Support device configuration via built-in IVR, Web browser or encrypted configuration files through TFTP or HTTP server Support firmware upgrade via TFTP or HTTP Support PSTN pass through (on Rev.2.0) Support SIP Session Timer Support Syslog (on Rev.2.0) Support volume amplification Support configurable Call Progress Tones Ultra compact (wallet size) and lightweight design, great companion for travelers Compact, lightweight Universal Power adapter

Hardware Specification

The table below lists the hardware specification of HandyTone-486.
Model LAN interface WAN interface FXS telephone port PSTN port Button LED Universal Switching Power Adaptor Dimension
HandyTone-486 1xRJ45 10Base-T 1xRJ45 10Base-T 1xFXS 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) 130mm (D) 27mm (H) 0.6lbs (0.3kg) 40 - 130oF 5 45oC 10% - 90% (non-condensing)
Weight Temperature Humidity Compliance
NOTE: HandyTone-486 has two hardware revisions. This information can be found on the label at the bottom of the device. The difference between HandyTone-486 Rev.1.0 and HandyTone-486 Rev.2.0 is that a HandyTone-486 Rev.2.0 line port can function as PSTN pass through while a HandyTone-486 (Rev.1.0, old model, no longer shipped) line port is just a life line port and will bridge to PSTN only when the device is out of power.

Basic Operations

Get Familiar with Key Pad and Voice Prompt
HandyTone-486 stores a voice prompt menu (Interactive Voice Response or IVR) for quick browsing and simple configuration. To enter this voice prompt menu, simply press the button on the HandyTone-486 or pick up the phone and dial ***. The following table shows how to use the voice prompt menu to configure the device.

Menu Main Menu

Voice Prompt Enter a Menu Option
DHCP Mode, or Static IP Mode

IP Address + IP address

Subnet + IP address Gateway + IP address DNS Server + IP address Preferred Vocoder

WAN Port Web Access

Firmware Server IP Address Configuration Server IP Address Upgrade Protocol
Users Options Enter * for the next menu option Enter # to return to the main menu Enter 01 06, 47, 86 or 99 Menu option Enter 9 to toggle the selection If user selects Static IP Mode, user need configure all the IP address information through menu 02 to 05. If user selects Dynamic IP Mode, the device will retrieve all IP address information from DHCP server automatically when user reboots the device. The current WAN IP address is announced Enter 12-digit new IP address if in Static IP Mode. Same as Menu option 02 Same as Menu option 02 Same as Menu option 02 Enter 9 to go to the next selection in the list: - PCM U - PCM A - G-723 - G-729 - iLBC - G-726 Enter 9 to toggle between: - enable - disable The current Firmware Server IP address is announced. Enter 12 digit new IP address. The current Config Server Path IP address is announced. Enter 12 digit new IP address. Upgrade protocol for firmware and configuration update. Enter 9 to toggle between: - TFTP - HTTP 10

Firmware Version Firmware Upgrade

Direct IP Calling

Invalid Entry
Firmware version information. Firmware upgrade mode. Enter 9 to rotate among the following three options: - always check - check when pre/suffix changes - never upgrade When entered, user will be prompted a dial tone, dial a 12-digit IP address to make a direct IP call. (For details, see 4.2.2 Make a Direct IP Call.) Enter 9 to reboot the device; or Enter MAC address to restore factory default setting (For details, see section 8.) Automatically returns to Main Menu
IVR supports error reporting when the following problems occur. User will hear silence when picking up the handset. After pressing ***, user will hear one or more error codes listed below. User may hear one or more error codes depending on errors detected such as E104E103E. Upon hearing error code, user can press # to get into the IVR main menu. E101E E102E E103E E104E Ethernet link down No IP address obtained (DHCP or PPPoE mode) Device is not registered to SIP server No STUN responses
NOTES: Once the LED button is pressed, it enters voice prompt main menu. If the button is pressed again while it is already in the voice prompt menu state, it jumps to Direct IP Calling option and dial tone plays in this state * 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. Once all digits are accumulated, the device will automatically process them For IP address input, omit the dot and enter the digits directly, add 0 for those octets with less than three digits. e.g.: IP: 192.168.1.10, key in: 192168001010 Key entry cannot 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 seconds (Default No Key Entry Timeout). Or 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 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, followed by pressing the # or wait for 4 seconds.

The password is case sensitive with a 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 ADVANCED SETTINGS configuration page. NOTE: If you can not log into the configuration page by using the default password, please check with your VoIP service provider. Most likely, the service provider has provisioned the device and configured for you and changed the default password.
After the correct password is entered into the login screen, the embedded Web server inside the device will respond with a BASIC SETTINGS configuration page.
End User Password Web Port IP Address
This field contains the password to access the Web Configuration Menu. The password is case sensitive with a maximum of 25 characters. This is the devices internal HTTP server port. Default is 80. There are 2 modes under which the HandyTone ATA can operate: - 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) and the IP phone will acquire its IP address from the first DHCP server it discovers on the LAN it attaches to. To use PPPoE feature please set the PPPoE account settings if the HandyTone ATA is connected directly to a DSL modem. The HandyTone ATA will attempt to establish a PPPoE session if any of the PPPoE fields are set. - If Static IP mode is selected, then the IP address, Subnet Mask, Default Router IP address, DNS Server 1 (primary), DNS Server 2 (secondary) fields will need to be configured. These fields are set to zero by default.
DHCP hostname DHCP domain DHCP vendor class ID PPPoE account ID PPPoE password
This option specifies the name of the client. This field is optional but may be required by some Internet Service Providers. Default is blank. This option specifies the domain name that client should use when resolving hostnames via the Domain Name System. Default is blank. This option is used by clients and servers to exchange vendor-specific information. Default is blank. PPPoE username. Fill this field if your ISP requires you to use a PPPoE (Point to Point Protocol over Ethernet) connection. PPPoE account password.
Time Zone Daylight Savings Time
This parameters decides how the displayed date/time will be adjusted according to the specified time zone. This parameter controls whether the displayed time will be daylight savings time or not. If set to Yes and the Optional Rule is empty, then the displayed time will be 1 hour ahead of normal time. The Automatic Daylight Saving Time Rule shall have the following syntax: start-time;end-time;saving Both start-time and end-time have the same syntax: month,day,weekday,hour,minute month: 1,2,3,.,12 (for Jan, Feb,., Dec) day: [+|-]1,2,3,.,31 weekday: 1, 2, 3,., 7 (for Mon, Tue,., Sun), or 0 which means the daylight saving rule is not based on week days but based on the day of the month. hour: hour (0-23), minute: minute (0-59) If weekday is 0, it means the date to start or end daylight saving is at exactly the given date. In that case, the day value must not be negative. If weekday is not zero and day is positive, then the daylight saving starts on the first dayth iteration of the weekday (1st Sunday, 3rd Tuesday etc). If weekday us not zero and day is negative, then the daylight saving starts on the last dayth iteration of the weekday (last Sunday, 3rd last Tuesday etc). The saving is in the unit of minutes. The saving time may also be preceded by a negative (-) sign if subtraction is desired instead of addition. The default value for Automatic Daylight Saving Time Rule shall be set to 04,01,7,02,00;10,-1,7,02,00;60 which is the rule for US. Examples 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).

PSTN Access Code Device Mode WAN Side HTTP Access
Default is *00. User can switch the phone to PSTN line connected to the Line port of ATA and make outgoing calls. Default is NAT router mode. HandyTone-486 Rev.2.0 can be configured in Bridge mode so the device functions as a bridge. Default is No. The access to configuration page via WAN port is disabled. This setting has no effect if the device is in Bridge mode. 20
Reply to ICMP on WAN port Cloned WAN MAC Address: LAN Subnet Mask
Unit will not respond to PING from WAN side if set to No. Allow user to set a specific MAC address. Set in Hex format. Sets the LAN subnet mask. Default value is 255.255.255.0. If bridge mode is selected, the LAN settings have no effect.
LAN DHCP Base IP: Base IP for the LAN port, which functions as a gateway for its LAN. Default value is 192.168.2.1 DHCP IP Lease Time DMZ IP: Port Forwarding: The amount of time that a given IP address will be valid for a LAN client. Value is set in units of hours. Default value is 120hr (5 Days). Forward all WAN IP traffic to a specific IP address if no matching port is used by HandyTone-486 itself or in the defined port forwarding. Allow user to forward a matching (TCP/UDP) port to a specific LAN IP address with a specific (TCP/UDP) port.
In addition to the Basic Settings configuration page, end users also have access to the device Status page. MAC Address WAN IP Address Product Model Software Version The device ID, in HEX format. troubleshooting. This is very important ID for ISP
This field shows WAN port IP address. This field contains the product model info, such as HT486 Rev:2.0 Program: This is the main software release. This number is always used for firmware upgrade. Current release is 1.0.8.32. Bootloader: current version is 1.0.8.11. HTML: current version 1.0.8.32. VOC: current version is 1.0.0.12 This shows how long the device has been up since the last reboot. This shows whether the unit is registered to service providers server or not. This field shows whether the PPPoE connection is up if the HandyTone ATA is connected to DSL modem. This shows what kind NAT the HandyTone ATA is connected to via its WAN port. It is based on STUN protocol. 21

Fax Mode Layer 3 QoS Layer 2 QoS (VoIP) Layer 2 QoS ( PC)
Allow incoming SIP If set to Yes, the device will ignore any SIP message that does not come from messages from SIP the IP address (Source IP in the IP header) that it is registered to. Default setting is No. proxy only Use DNS SRV: User ID is Phone Number SIP Registration Unregister on Reboot Default is No. If set to Yes the client will use DNS SRV to lookup for the server. If set to yes, a user=phone parameter will be attached to the From header in SIP request. This parameter decides whether the HandyTone ATA needs to send REGISTER messages to the proxy server. The default setting is Yes. Default is No. If set to Yes, the device will first send registration request to remove previous bindings.
Register Expiration This parameter allows the user to specify the time frequency (in minutes) the HandyTone ATA will refresh its registration with the specified registrar. The default interval is 60 minutes (or 1 hour). The maximum interval is 65535 minutes (about 45 days). Early Dial Allow outgoing call without Registration Dial Plan Prefix No Key Entry timeout Use # as Send Key Default is No. Use only if proxy supports 484 response. Default is No. When set to Yes, the device still has a dial tone and allows user make outgoing calls even if the device is not registered to the SIP proxy. Sets the prefix added to each dialed number. Default is 4 seconds. This parameter allows user to configure the # key to be used as the Send (or Dial) key. Once set to Yes, pressing this key will immediately trigger the sending of dialed string collected so far. If set to No, the # key will then be included as part of the dialed string to be sent out. This parameter defines the local SIP port the HandyTone ATA listens and transmits. The default value is 5060. This parameter defines the local RTP-RTCP port pair the HandyTone ATA listens and transmits. It is the base RTP port for channel 0. When configured, channel 0 will use this port_value for RTP and the port_value+1 for its RTCP; channel 1 will use port_value+2 for RTP and port_value+3 for its RTCP. The default value is 5004. Default No. If set to Yes, the device will pick randomly-generated SIP and RTP ports. This is usually necessary when multiple HandyTone ATAs are behind the same NAT.
Local SIP port Local RTP port

Use Random Port

SIP Registration Failure Retry Wait Time NAT Traversal
Retry registration if the process failed. Default is 20 seconds.
This setting decides whether the NAT traversal mechanism is activated. It should be set to Yes if the device is behind a NAT router. If no outbound proxy is configured, a STUN server needs to be set to activate STUN detection mechanism. Usually ITSP will provide these settings. If this field is set to Yes, then the device will periodically (every Keep-alive interval) send a dummy UDP packet to the SIP server to pinhole the NAT. Default is 20 seconds. The minimum value allowed is 10 seconds. This is the interval of sending dummy UDP packet to keep the NAT pin hole open. If configured, the NAT IP address will be used in SIP/SDP message. Default is blank. Use STUN keep-alive to detect WAN side network problems. If keep-alive request does not yield any response for configured number of times, the device will restart the TCP/IP stack. If the STUN server does not respond when the device boots up, the feature is disabled. SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.

DTMF Payload Type Send Flash Event Onhook Threshold FXS Impedance Caller ID Scheme

Onhook Voltage

Polarity Reversal NTP server Send Anonymous
Anonymous Method If it is set to Use from header. Callers SIP user ID will be sent as anonymous, essentially block the Caller ID from displaying. If it is set to User privacy header, the SIP INVITE message contains a privacy header, and the server blocks the caller ID from the called party. Time to Ring Special Features The duration of ringing when a call is not answered. Default is 60 seconds. Default is Standard. Choose the selection to meet some special requirements from Soft Switch vendors like Nortel, Broadsoft, etc.
CBCOM Encode CBCOM Encoder 1.1 Key Syslog Server Syslog Level
SIP, RT(C)P and T.38 modes, 1.0 and 1.1 Key to be used by CBCOM The IP address or URL of System log server. This feature is especially useful for ITSP (Internet Telephone Service Provider). Select the ATA to report the log level. Default is NONE. The level is one of DEBUG, INFO, WARNING or ERROR. Syslog messages are sent based on the following events: product model/version on boot up (INFO level) NAT related info (INFO level) sent or received SIP message (DEBUG level) SIP message summary (INFO level) inbound and outbound calls (INFO level) registration status change (INFO level) negotiated codec (INFO level) Ethernet link up (INFO level) SLIC chip exception (WARNING and ERROR levels) memory exception (ERROR level)
The Syslog uses USER facility. In addition to standard Syslog payload, it contains the following components: GS_LOG: [device MAC address][error code] error message Here is an example: May 19 02:40:38 [00:0b:82:00:a1:be][000] Ethernet link is up Session Expiration 192.168.1.14 GS_LOG:
Grandstream implemented SIP Session Timer. The session timer extension enables SIP sessions to be periodically refreshed via a re-INVITE request. Once the session interval expires, if there is no refresh via a re-INVITE message, the session will be terminated. Session Expiration is the time (in seconds) at which the session is considered timed out, if no successful session refresh transaction occurs beforehand. The default value is 180 seconds.

Min-SE

The minimum session expiration (in seconds). The default value is 90 seconds.
Caller Request Timer Callee Request Timer Force Timer
If selecting Yes the phone will use session timer when it makes outbound calls if remote party supports session timer. If selecting Yes the phone will use session timer when it receives inbound calls with session timer request. If selecting Yes the phone will use session timer even if the remote party does not support this feature. Selecting No will allow the phone to enable session timer only when the remote party support this feature. To turn off Session Timer, select No for Caller Request Timer, Callee Request Timer, and Force Timer. As a Caller, select UAC to use the phone as the refresher, or UAS to use the Callee or proxy server as the refresher. As a Callee, select UAC to use caller or proxy server as the refresher, or UAS to use the phone as the refresher. Session Timer can be refreshed using INVITE method or UPDATE method. Select Yes to use INVITE method to refresh the session timer. 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, HT486 rev. 2.0 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.

UAC Specify Refresher UAS Specify Refresher Force INVITE Firmware Upgrade and Provisioning Firmware Server Path Config Server Path Firmware File Prefix Firmware File Postfix Config File Prefix Config File Postfix
Automatic Upgrade Choose Yes to enable automatic upgrade and provisioning and input the number, in minutes, you want the HT to check for an update. When set to No, HT502 will only do upgrade once at boot up. Always check for New Firmware at Boot up will check for new firmware every time the device reboots. Check New Firmware only when F/W pre/suffix changes will check for updates only when the pre/suffix has been changed. Firmware Key For firmware encryption. It should be 32 digit in Hexadecimal Representation. End user should keep it blank.
Authenticate Conf File Lock keypad update Allow conf SIP Account in Basic Settings Volume Amplification Powerline Ring Tone
Default is No. If set to Yes, configuration file would be authenticated before acceptance. End user should use default setting. If this parameter is set to Yes, except for IVR MENU items 1 to 5, the configuration update via keypad is disabled. Default No. If set to Yes, user ID, authentication IP, authentication password and display name can be configured in BASIC SETTINGS page. Handset volume adjustment. RX is for receiving volume, TX is for transmission volume. Default values are 0dB for both parameters. +6dB generates the highest volume and -6dB generates the lowest volume. This setting allows user to configure the ringing frequencies and cadences.
Call Progress Tones Using these settings, users can configure various call progress tone frequencies and cadences according to their country standard. By default they are set to North American standard. 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. Up to three cadences are supported. Disable Line Echo Canceller (LEC): Default is No. If set to Yes, echo canceller is not used.
Saving the Configuration Changes
Once a change is made, users should click on the Update button in the Configuration page. The HandyTone-486 will then display a screen to confirm that the changes have been saved.

Rebooting the HandyTone-486 from remote
The user/administrator of the HandyTone-486 can remotely reboot the HandyTone-486 by pressing the Reboot button at the bottom of the configuration page. Once done, a screen will be displayed to indicate that rebooting is underway.
Configuration through a Central Server
Grandstream HandyTone ATAs can be automatically configured from a central provisioning system. 29
When HandyTone ATA boot up, it will send TFTP or HTTP request to download configuration file, cfg000b82xxxxxx, where 000b82xxxxxx is the MAC address of the HandyTone ATA. The configuration files can be downloaded via TFTP or HTTP from the central server. A service provider or an enterprise with large deployment of HandyTone ATA can easily manage the configuration and service provisioning of individual devices remotely from a central server. Grandstream has a provisioning system called GAPS (Grandstream Automated Provisioning System) that is used to support automated configuration of Grandstream devices. GAPS uses enhanced (NAT friendly) TFTP or HTTP (thus no NAT issues) and other communication protocols to communicate with each individual Grandstream device for firmware upgrade, remote reboot, etc. Grandstream provides GAPS service to VoIP service providers. Use GAPS for either simple redirection or with certain special provisioning settings. At boot-up, Grandstream devices by default point to Grandstream provisioning server GAPS, based on the unique MAC address of each device, GAPS provision the devices with redirection settings so that they will be redirected to customers TFTP or HTTP/HTTPS server for further provisioning. Grandstream also provide GAPSLite software package which contains our NAT friendly TFTP server and a configuration tool to facilitate the task of generating device configuration files. The GAPSLite configuration tool is now free to end users. The tool and configuration template are available for download from http://www.grandstream.com/configurationtool.html

Software Upgrade

Software upgrade can be done via either TFTP or HTTP. The corresponding configuration settings are in the ADVANCED SETTINGS configuration page.
Firmware Upgrade through TFTP/HTTP
To upgrade via TFTP or HTTP, the Firmware Upgrade and Provisioning upgrade via field (IVR option 17) needs to be set to TFTP or HTTP, respectively. Firmware Server Path needs to be set to a valid URL of a TFTP or HTTP server, server name can be in either FQDN or IP address format. Here are examples of some valid URL. e.g. firmware.mycompany.com:6688/Grandstream/1.1.0.31 e.g. 168.75.215.189 NOTES: Firmware server in IP address format can be configured via IVR. Please refer to section 5.1 for instructions. If firmware server is in FQDN format, it must be set via web configuration interface. Grandstream recommends end-user use the Grandstream TFTP server. Its address can be found at http://www.grandstream.com/firmware.html. Currently the TFTP firmware server IP address is 168.75.215.189. For large companies, we recommend to maintain their own TFTP/ HTTP/HTTPS server for upgrade and provisioning procedures. Once a Firmware Server Path and the upgrade protocol are set, user needs to update the settings and reboot the device. If the configured firmware server is found and a new code image is available, the HandyTone ATA will attempt to retrieve the new image files by downloading them into the HandyTone ATAs SRAM. During this stage, the HandyTone ATAs LEDs will blink until the checking/downloading process is completed. Upon verification of checksum, the new code image will then be saved into the Flash. If TFTP/HTTP fails for any reason (e.g., TFTP/HTTP server is not responding, there are no code image files available for upgrade, or checksum test fails, etc), the HandyTone ATA will stop the TFTP/HTTP process and simply boot using the existing code image in the flash. Firmware upgrade may take as long as 1 to 20 minutes over Internet, or just 20+ seconds if it is performed on a LAN. It is recommended to conduct firmware upgrade in a controlled LAN environment if possible. For users who do not have a local firmware upgrade server, Grandstream provides a NAT-friendly TFTP server on the public Internet for firmware upgrade. Please check the Services section of Grandstreams Web site to obtain our public TFTP servers IP address. Alternatively, user can download a free TFTP or HTTP server and conduct local firmware upgrade. A free windows version TFTP server is available for download from http://support.solarwinds.net/updates/New-customerFree.cfm.

Instructions to download a free TFTP Server: 1. Unzip the file and put all of them under the root directory of the TFTP server. 31
2. Put the PC running the TFTP server and the HT486 device in the same LAN segment. 3. Please 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 phones web configuration page 5. Configure the Firmware Server Path with the IP address of the PC 6. Update the change and reboot the unit Please be advised that our client will pull out firmware from the WAN side, if the TFTP server is connected to the devices LAN port, the firmware upgrade will not work by design.
Configuration File Download
Grandstream SIP Device can be configured via Web Interface as well as via Configuration File through TFTP or HTTP. Config Server Path is the TFTP or HTTP server path for configuration file. It needs to be set to a valid URL, either in FQDN or IP address format. The Config Server Path can be same or different from the Firmware Server Path. A configuration parameter is associated with each particular field in the web configuration page. A parameter consists of a Capital letter P and 2 to 3 (Could be extended to 4 in the future) digit numeric numbers. i.e., P2 is associated with Admin Password in the ADVANCED SETTINGS page. For a detailed parameter list, please refer to the corresponding firmware release configuration template. When Grandstream Device boots up or reboots, it will issue request for configuration file named cfgxxxxxxxxxxxx, where xxxxxxxxxxxx is the MAC address of the device, i.e., cfg000b820102ab. The configuration file name should be in lower cases.
Firmware and Configuration File Prefix and Postfix
Starting from firmware version 1.0.7.11 for HandyTone-486 Rev 2.0, adding prefix and postfix for both firmware and configuration file is supported. Firmware Prefix and Postfix allows device to download the firmware name with the matching Prefix and Postfix. This makes it the possible to store ALL of the firmware with different version in one single directory. Similarly, Config File Prefix and Postfix allows device to download the configuration file with the matching Prefix and Postfix. Thus multiple configuration files for the same device can be stored in one directory. In addition, when the field Check New Firmware only when F/W pre/suffix changes is set to Yes, the device will only issue firmware upgrade request if there are changes in the firmware Prefix or Postfix.

Managing Firmware and Configuration File Download
When Automatic Upgrade is set to Yes, Service Provider can use P193 (Auto Check Interval, in minutes, default and minimum is 60 minutes) to have the devices periodically check with either Firmware Server or Config Server, whenever they are defined. This allows the device periodically check if there are any new changes need to be taken on a scheduled time. By defining different intervals in P193 for different devices, Server Provider can spread the Firmware or Configuration File download in minutes to reduce the Firmware or Provisioning Server load at any given time.

Warning !!!

Restore Factory Default Setting
Restore the Factory Default Setting will DELETE all configuration information of the device. 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 service provider. Please disconnect network cable and power cycle the unit before trying to reset the unit to factory default. The steps are as follows: Step 1: Find the MAC Address of the device. It is a 12 digits HEX number located on the bottom of the unit. Step 2: 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. Step 3: To perform factory reset: a. b. c. d. Press *** or the LED button 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: Please be aware by default the HandyTone-486 WAN side HTTP access is disabled. After a factory reset, the devices web configuration page can be accessed only from its LAN port, please refer to instructions in section 6.2.1 for details. 34