GrandStream HT488 User Manual

Grandstream Networks, Inc.

HT-488
Dual FXS/FXO Port
Analog Telephone Adaptor

HT–488 User Manual www.grandstream.com
Firmware Version 1.0.3.64 [e-mail]

TABLE OF CONTENTS

HT-488

USER MANUAL

WELCOME.......................................................................................................................................................4

SAFETY COMPLIANCES................................................................................................................................4

WARRANTY.................................................................................................................................................4

INSTALLATION.................................................................................................................................................5

EQUIPMENT PACKAGING..............................................................................................................................5

CONNECTING YOUR ATA.............................................................................................................................5

FIVE EASY STEPS TO INSTALL THE HT488.....................................................................................................6

PRODUCT OVERVIEW .......................................................................................................................................7

KEY FEATURES...........................................................................................................................................7

HARDWARE SPECIFICATION .........................................................................................................................8

BASIC OPERATIONS ........................................................................................................................................9

GET FAMILIAR WITH VOICE PROMPT.............................................................................................................9

MAKE PHONE CALLS .................................................................................................................................10

CALLING PHONE OR EXTENSION NUMBERS.................................................................................................10

DIRECT IP CALLS......................................................................................................................................10

CALL HOLD...............................................................................................................................................11

CALL WAITING ..........................................................................................................................................11

CALL TRANSFER........................................................................................................................................11

3-WAY CONFERENCING .............................................................................................................................12

PSTN PASS THROUGH .............................................................................................................................13

VOIP-TO-PSTN CALLS ............................................................................................................................13

PSTN-TO-VOIP CALLS.............................................................................................................................14

ROUTE CALLS TO PSTN............................................................................................................................14

FORWARD CALLS TO PSTN.......................................................................................................................14

FORWARD CALLS TO VOIP........................................................................................................................15

FAX SUPPORT...........................................................................................................................................15

CALL FEATURES ...........................................................................................................................................16

LED LIGHT PATTERN INDICATION ..........................................................................................................16

CONFIGURATION GUIDE.................................................................................................................................17

CONFIGURING HT488 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- 488 WITH WEB BROWSER.............................................................................................17

ACCESS THE WEB CONFIGURATION MENU .............................................................................................17

END USER CONFIGURATION ......................................................................................................................18

ADVANCED CONFIGURATION AND FXS/FXO PORTS PARAMETERS...............................................................21

SAVING THE CONFIGURATION CHANGES.....................................................................................................25

REBOOTING THE HT488 FROM REMOTE ................................................................................................25

CONFIGURATION THROUGH A CENTRAL SERVER .........................................................................................26

SOFTWARE UPGRADE ...................................................................................................................................27

FIRMWARE UPGRADE THROUGH TFTP/HTTP ............................................................................................27

CONFIGURATION FILE DOWNLOAD .............................................................................................................28

FIRMWARE AND CONFIGURATION FILE PREFIX AND POSTFIX........................................................................28

MANAGING FIRMWARE AND CONFIGURATION FILE DOWNLOAD ....................................................................28

RESTORE FACTORY DEFAULT SETTING..........................................................................................................29

GLOSSARY OF VOIP TERMS ..........................................................................................................................30

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TABLE OF FIGURES

HT–488

USER MANUAL

FIGURE 1: CONNECTING THE HT488 ..................................................................................................5

FIGURE 2: INTERCONNECTION DIAGRAM OF THE HT488 ......................................................................6

FIGURE 3: SCREENSHOT OF CONFIGURATION LOG-IN PAGE............................................................... 18

FIGURE 4: SCREENSHOT OF REBOOTING SCREEN..............................................................................25

ABLE OF TABLES

T

HT–488

USER MANUAL

TABLE 1: DEFINITIONS OF THE HT488 CONNECTORS ..........................................................................5

TABLE 2: HT488 TECHNICAL SPECIFICATIONS ....................................................................................7

TABLE 3: HT488 HARDWARE SPECIFICATION......................................................................................8

TABLE 4: HT488 IVR MENU DEFINITIONS........................................................................................... 9

TABLE 5: HT488 CALL FEATURE DEFINITIONS .................................................................................. 16

TABLE 6: HT488 LED DEFINITIONS..................................................................................................16

TABLE 7: HT488 DEVICE STATUS PAGE DEFINITIONS........................................................................19

TABLE 8: HT488 BASIC SETTINGS PAGE DEFINITIONS.......................................................................19

TABLE 9: HT488 ADVANCED SETTINGS PAGE DEFINITIONS ...............................................................21

TABLE 10: HT488 FXS PORT SETTINGS PAGES DEFINITIONS.......................................................... 22

TABLE 11: HT488 FXO PORT SETTINGS PAGES DEFINITIONS..........................................................24

TABLE 12: HT488 CALL PROGRESS TONES SETTINGS PAGE DEFINITIONS......................................... 25

TABLE OF GUI INTERFACES

HT–488

USER MANUAL

1. S

CREENSHOT OF CONFIGURATION LOGIN PAGE

TATUS CONFIGURATION PAGE DEFINITIONS

2. S

3. SCREENSHOT OF BASIC SETTINGS CONFIGURATION PAGE

4. S

CREENSHOT OF ADVANCED SETTINGS CONFIGURATION PAGE

5. SCREENSHOT OF FXS ACCOUNT CONFIGURATION

CREENSHOT OF FXO ACCOUNT CONFIGURATION

6. S

7. S

CREENSHOT OF CALL PROGRESS TONES CONFIGURATION PAGE

8. SCREENSHOT OF SAVED CONFIGURATION CHANGES

CREENSHOT OF REBOOT PAGE

9. S

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WELCOME

Congratulations on becoming an owner of HT488. You made an excellent choice and we hope you enjoy
all of its capabilities.

Grandstream's HT488 is an all-in-one VoIP integrated access device that features superb audio quality,
rich functionalities, high level of integration, compactness and ultra-affordability. The HT488 is fully
compatible with SIP industry standard and can interoperate with many other SIP compliant devices and
software on the market.

Grandstream HT488 is a new addition to the popular HT product family. It is an enhanced model
compared to the award-winning HT488 in that it allows call origination and termination from/to the PSTN
network (via FXO port) remotely and automated emergency call routing through PSTN network.

Grandstream HT488 has been awarded the Best of Show product in 2005 Internet Telephony Conference
and Expo.

SAFETY COMPLIANCES

The HT488 adaptor complies with FCC/CE and various safety standards. The HT488 power adaptor is
compliant with UL standard. Only use the universal power adapter provided with the HT488 package.
The manufacturer’s warranty does not cover damages to the phone caused by unsupported power
adaptors.

WARRANTY

If you purchased your HT488 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.

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.

• This document is contains links to Grandstream GUI Interfaces. Please remember to download these
examples

• This document is subject to change without notice. The latest electronic version of this user manual is
available for download from the following location:

http://www.grandstream.com/pdf/HT488UserManual.pdf

• Reproduction or transmittal of the entire or any part, in any form or by any m eans, electronic or print,
for any purpose without the express written permission of Grandstream Networks, Inc. is not
permitted.

http://www.grandstream.com/user_manuals/GUI/GUI_HT488.rar for your reference.

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INSTALLATION

EQUIPMENT PACKAGING

The HT488 ATA package contains:

• One HT488 Main Case

• One Universal Power Ad

One Ethernet Cable

•

CONNECTING YOUR ATA

The HT488 Analog Telephone Adaptor is an all-in-one VoIP integrated device designed to be a total
solution for networks providing VoIP services. The HT488 VoIP features and functions are available
using a regular analog telephone.

FIGURE 1: CONNECTING THE HT488

aptor

RJ-11

FXS Port

(Phone)

RJ-45

RJ-45

10M net Ether

10M net

Ether

LAN/WAN

LAN/WAN

+5V/1200mA

LED Button
(green/red)

RJ-11

FXO Port

PSTN Line

The HT488 has one FXS port and one FXO port. The PHONE port next to the LAN port is a FXS port.
The LINE port on the side of the HandyTone-488 is a FXO port. Both the FXS port and the FXO port can
have a separate SIP account. This is a key feature of HT488 as it supports simultaneous

calls on both
the FXS port and FXO port. Telephone calls can be originated from or terminated on the PSTN network
remotely via the FXO port.

TABLE EFINITIONS OF THE ONNECTORS 1: D HT488 C

+5V/1.2A
LAN Port (RJ-45)
WAN Port (RJ-45)
PHONE (RJ-11)
LINE (RJ-11)
BUTTON

Power adapter connection
Connect the LAN port with an Ethernet cable to your PC.
Connect to the internal LAN network or router.
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 HT488

The HT488 is designed for easy configuration and easy installation. Configure the HT488 following the
directions in the Configuration section of this manual.

1. Connect a standard touch-tone analog teleph one to the PHONE port.

2. Insert a standard RJ11 telephone cable into the LINE port and connect the other end of the
telephone cable to a wall jack.

3. Insert the Ethernet cable into the WAN port of HT488 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 488 if HT488 is used as a router.

5. Insert the power adapter into the HandyTone 488 and connect it to a wall outlet.

FIGURE 2: INTERCONNECTION DIAGRAM OF THE HT488

Internet

ADSL/Cable

Modem Ethernet

WAN

FXS FXO

Analog Phone

PSTN

Cloud

LAN

Cordless Phone

Fax

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PRODUCT OVERVIEW

HT488 is a next generation dual-port SIP IAD for Internet data, voice, and fax. It has rich features and
had the added functionality of a bridge that enables you to make remote calls over the internet.

K

EY FEATURES

Ethernet

Ports

DHCP

FXS
Port

PSTN Pass

– through

Voice Mail

Indicator

Voice Codec

Remote

Configuration

2 RJ45

(LAN/WAN)

Server/

Client

1 Yes Yes

iLBC, G.723,

G.711, G.729,

G.726, T.38

TFTP/HTTP

T

ABLE 2: HT488 TECHNICAL SPECIFICATIONS

Lines/SIP Accounts
Protocol Support

Feature Keys
LAN/WAN Interface
Device Management

Support device configuration via built-in IVR, Web browser or central configuration file

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

Syslog support

DHCP Server/Client
Audio Features

Call Handling Features

Network and
Provisioning

Fax over IP

Security
Physical Design

2 lines / 2 SIP accounts
SIP 2.0 (RFC 3261), TCP/UDP/IP, RTP/RTCP, HTTP, ARP/RARP, ICMP, DNS, DHCP,

NTP, TFTP, PPPoE protocols
1 button

RJ-45 10 Mbps
Web interface or via secure (AES encrypted) central configuration file for mass

deployment

through TFTP or HTTP

behind firewall/NAT

Yes
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 µ/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

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HARDWARE SPECIFICATION

The table below lists the hardware specification of HT488.

TABLE 3: HT488 HARDWARE SPECIFICATION

LAN interface 1xRJ45 10Base-T
WAN interface 1xRJ45 10Base-T
FXS telephone port 1 x FXS
FXO telephone port (PSTN Port) 1x PSTN pass-through and life line port
Button 1
LED Green and Red color
Universal Switching

Power Adaptor

Dimension 70mm (W)

Weight 0.6lbs (0.3kg)
Temperature 40 - 130oF

Humidity 10% - 90%

Compliance

Input: 100-240VAC 50-60 Hz
Output: +5VDC, 1200mA
UL certified

130mm (D)
27mm (H)

5 – 45oC

(non-condensing)

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BASIC OPERATIONS

GET FAMILIAR WITH VOICE PROMPT

HT488 has a stored voice prompt menu for quick browsing and simple configuration. Currently, the voice
prompt menu and the LED button is designed for the FXS port only
press the LED button or “***” from the analog phone.

TABLE 4: HT488 IVR MENU DEFINITIONS

Menu Voice Prompt Options

Main Menu

01

02

03
04
05

07

12
13

14

15

16
17

47

99

“Enter a Menu Option”

“DHCP Mode”,
“Static IP Mode”

“IP Address “ + IP address The current WAN IP address is announced

“Subnet “ + IP address Same as menu 02
“Gateway “ + IP address Same as menu 02
“DNS Server “ + IP address Same as menu 02

Preferred Vocoder

WAN Port Web Access
Firmware Server IP Address

Configuration Server IP
Address

Upgrade Protocol

Firmware Version
Firmware Upgrade

“Direct IP Calling”

“RESET”

“Invalid Entry”

Press “*” for the next menu option
Press “#” to return to the main menu
Enter 01-05, 07,12-17,47 or 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.

If using “Static IP Mode”, enter 12 digit new IP address.

Press “9” to move to the next selection in the list:

• PCM U / PCM A

• G.723

• G.729

• G.726

• iLBC

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
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

. To enter the voice prompt menu,

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NOTE:

• 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 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 the web

configuration).

E

XAMPLES:

• 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 may need a prefix number followed by the
phone number. Please check with your VoIP service provider for this information. If your phone is
assigned a PSTN-like number such as 6265556789, you will most likely follow the rule 1 + (the
number) – 16266667890. Press # or wait for 4 seconds.

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 HT-488 and other VoIP Device (i.e. another Handytone ATA or Budgetone SIP phone or
other VoIP unit) have public IP addresses, or

• Both HT-488 and other VoIP Device are on the same LAN using private IP addresses, or

• Both HT-488 and other VoIP Device can be connected through a router using public or private IP

addresses (with necessary port forwarding or DMZ).

O PLACE A DIRECT IP CALL:

T

1. Pick up the analog phone (or use the speakerphone),

2. Access the voice menu prompt by dial “***” or press the button on the HT-488

3. Dial “47” to access the direct IP call menu

4. At voice prompt “Direct IP Calling” and 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.

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EXAMPLES:

1. If the target IP address is 192.168.0.10, the dialing convention is
Voice Prompt with option 47, then 192 168 000 010
followed by pressing the “#” key if it is configured as a send key or wait for more than 5 secon ds.

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 pressing the “#” key if it is configured as a send key or wait for 4 seconds.

NOTE: When placing a direct IP call, the “Use Random Port” should be set to “NO”.

CALL HOLD

This function is applicable on the FXS port for VoIP calls only.
button on the connected phone (if the phone has that button) places the remote end on hold. Pressing the
“flash” button again releases the previously held party and the conversation can resume. If no “flash”
button is available, then on-off hook quickly (hook flash) will do the same thing. You may lose the call if
‘hook flash’ is not quick enough.

CALL WAITING

This function is applicable on FXS port for VoIP calls only.
user is in a conversation, he will hear a special stutter tone if there is another incoming call. User can
press the flash button to put the current call party on hold and switch to the other call. Pressing flash
button toggles between two active calls.

CALL TRANSFER

The HT488 supports both blind transfer and attended transfer.

Blind Transfer

This function is applicable using the FXS port for VoIP calls only.
conversation. Party A wants to Blind Transfer Party B to C:

1. A presses FLASH on the analog phone to hear the dial tone.

2. Then A dials *87, then dials C’s number, and then presses #

3. A can hang up.

NOTE: “Enable Call Feature” has to be set to “Yes” in web configuration page.
Three situations can follow the transfer:

1. A quick confirmation tone (temporarily using the call waiting indication tone) followed by a
dial tone. This indicates the transfer was successful (transferee has received a 200 OK from
transfer target). A can either hang up or make another call.

2. A quick busy tone followed by a restored call (on supported platforms only). This means the
transferee has received a 4xx response for the INVITE and we will try to recover the call. The
busy tone indicates the transfer has failed.

3. Busy tone keeps playing. This means we have failed to receive the second NOTIFY from the
transferee and the call has timed out. Note: this does not indicate the transfer has been
successful, nor does it indicate the transfer has failed. When transferee is a client that does not
support the second NOTIFY (such as our own earlier firmware), this situation occurs. In bad
network scenarios, this could also happen, although the transfer may have been completed
successfully.

While in conversation, pressing the “flash”

If call waiting feature is enabled, while the

Assume that parties A and B are in

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Attended Transfer

This function is applicable on the FXS port for VoIP calls only.
conversation. Party A wants to Attend Transfer Party B to C:

1. A presses FLASH on the analog phone to get a dial tone;

2. A then dial C’s number followed by #.

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: When Attended Transfer fails and A hangs up, the HT488 will ring user A back again to remind
A that party B is still on the call. Party A can pick up the phone to resume a conversation with party B.

3-WAY CONFERENCING

The HT488 supports both Star Code Style and Bellcore Style 3-way conferencing.

Assume that parties A and B are in

Star Code Style 3-way Conference

This function is applicable on the FXS port for VoIP calls only.
conversation. Party A wants to bring C into a 3-way 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 C’s 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

5. If A presses FLASH during conference, C will be dropped out.

Note: “Enable Call Feature” has to be set to YES in FXS PORT in the web configuration page.

Bellcore Style 3-way Conference

To use the Bellcore Style conference, the “Use Bell-style 3-way Conference” field in FXS PORT web
configuration must be enabled.

Assume that parties A and B are in conversation. Party A (using the HT-488) wants to bring C into a 3­way conference:

1. A presses FLASH (on the analog phone, or Hook Flash for old model phones) to get a dial
tone.

2. A dials C’s 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

5. If A presses FLASH during the conference, C will be dropped out.

the call, A can press FLASH back to talk to B.

the call, A can press FLASH back to talk to B.

Assume that parties A and B are in

Note: Party A is the call initiator for both calls with party B and party C.

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PSTN PASS THROUGH

HT488 supports PSTN pass through using the FXS port. The user can place and receive PSTN calls
when the FXS port is in use.

• To receive PSTN calls, pick up the phone when it rings;

• To complete a PSTN call, press the PSTN access code (*00 is default, or any number configured

in web configuration page) to switch to the PSTN line, hear a dial tone, then dial the number.

It the HT488 looses power, it will function as a jack, enabling a direct connection to the PSTN Line.

VOIP-TO-PSTN CALLS

This function is available using the FXO port. The FXO port functions as a bridge between the Internet
and PSTN. The user can remotely use a PSTN line to initiate a call.

T

O MAKE A VOIP-TO-PSTN CALL:

1. Dial the FXO SIP account phone number to establish the VoIP session. The caller will hear the
ring back tone once
special continuous tone is played if the pin code is configured, otherwise, the caller will hear a dial
tone.

2. Enter the pin code (configured on the configuration page). The caller will hear a dial tone and be
connected to the PSTN line if the pin code is valid. If the pin code is invalid, the continuous tone
is played to prompt caller to enter the pin code again. The user may try up to 3 times to enter a
correct pin code. After three (3) tries, the HT488 hangs up.

3. After the caller hears a dial tone from PSTN line, the caller can place the next call.

Note:

• Users can choose whether or not to apply password protection for VoIP-to-PSTN calls. A PIN
(Pin for PSTN calls) consists of up to 8 numeric digits and can be configured using the BASIC
SETTINGS of the web configuration page. By default, there is no password protection. (i.e. there
is no authentication required for callers on the use of PSTN line through HT488).

• When a PIN is configured for VOIP-to-PSTN call flow, the VoIP device that calls into the HT488
FXO account needs to configure RFC2833 or SIP Info for DTMF digit transmission.

• The special continuous tone is the prompt to enter a valid PIN code. If a caller doesn’t enter a
valid PIN, the HT488 times out after 10 seconds. Users may press the “#” key to indicate the end
of an input or wait 4 seconds.

• On the web configuration page, if the “Forward to PSTN” is configured, the second stage dialing
format is eliminated, so after dialing into the FXO SIP account number, the PSTN number will be
called automatically

. Then the caller hears either a special continuous tone or a dial tone. The

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PSTN-TO-VOIP CALLS

This function is available using the FXO port. The FXO port functions as a bridge between the Internet
and PSTN and enables calls to be passed from the PSTN network to VoIP. The user can make VoIP calls
remotely by dialing into the FXO line port on HT488.

T

O MAKE A PSTN-TO-VOIP CALL:

1. Make an incoming call to the PSTN line on FXO port. The phone will ring for 4 times by default
(this setting is configurable on the configuration page).

2. If no one answers the call after 4 rings (default configuration), then the caller hears either a
special continuous tone (prompting a PIN number) or a dial tone.

3. Enter a valid PIN. The caller will hear dial tone and be bridged to VoIP. If an incorrect PIN is
inputted, the continuous tone prompts caller to enter a valid PIN. The caller may try 3 times to
enter a valid PIN, then the HT488 will hang up.

4. The caller can dial a VoIP number followed by # (or wait for 4 seconds), the VoIP call will be
initiated from the SIP account configured on the FXO port.

Note:

• Users can choose whether or not to apply password protection for VoIP-to-PSTN calls. A PIN
(Pin for PSTN calls) consists of up to 8 numeric digits and can be configured using the BASIC
SETTINGS of the web configuration page. By default, there is no password protection. (i.e. there
is no authentication required for callers on the use of PSTN line through HT488).

• When a PIN is configured for VOIP-to-PSTN call flow, the VoIP device that calls into the HT488
FXO account needs to configure RFC2833 or SIP Info for DTMF digit transmission.

• The special continuous tone is the prompt to enter a valid PIN code. If a caller doesn’t enter a
valid PIN, the HT488 times out after 10 seconds. Users may press the “#” key to indicate the end
of an input or wait 4 seconds.

• On the web configuration page, if the “Forward to VoIP” is configured, the second stage dialing
format is eliminated, so after dialing into the FXO SIP account number, the PSTN number will be
called automatically

ROUTE CALLS TO PSTN

The FXO port enables access to the PSTN network. By default, the HT488 is in VoIP mode at off-hook.
If “Route call to PSTN” is configured, certain calls will be initiated from the FXO PSTN line port. This call
feature is especially useful for emergency calls or local telephone calls.

To use this feature, users need to specify a prefix or a telephone number in the “Route call to PSTN” in
the BASIC SETTINGS web configuration page. If the dialed digits match the specified prefix, outbound
calls will be initiated from PSTN line.

For example
initiated from the PSTN line.

FORWARD CALLS TO PSTN

Any VOIP call may be forwarded to a specified PSTN number if the call is not answered after a pre
configured numbers of rings. By default “Number of Rings” parameter has value 4.

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, if “Route call to PSTN” is configured as 626, all outgoing calls starting with 626 will be

For example, if the end-user has configured a cell phone number in the field “Forward to PSTN” under
BASIC SETTINGS configuration page, all calls will be forwarded to the cell phone number after 4 rings.

FORWARD CALLS TO VOIP

By default, each incoming PSTN call is received over the FXS port. The end-user may forward such a
call to any preconfigured VoIP extension, in case the call is not answered in a certain number of rings.
The Default value of the parameter “Number of Rings” is 4. If during 4 rings, the incoming from the PSTN
call is not answered, the call will be forwarded to another VoIP number previously configured in the field:
”Forward to VoIP”. This parameter can also be found under BASIC SETTINGS configuration page.

FAX SUPPORT

HT488 supports FAX in two modes: 1) T.38 (Fax over IP) and 2) 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

TABLE 5: HT488 CALL FEATURE DEFINITIONS

Key Call Features

*23 3-way Conferencing

Please refer to 3 way Calling section.
*30 Block Caller ID (for all subsequent calls)
*31 Send Caller ID (for all subsequent calls)
*67 Block Caller ID (per call). Dial “*67” + ” number ”. No dial tone will be played in the middle.
*82 Send Caller ID (per call). Dial “*82” + ” number ”. No dial tone will be played in the middle.
*50 Disable Call Waiting (for all-config change)
*51 Enable Call Waiting (for all-config change)

*70 Disable Call Waiting (Per Call)
*71 Enable Call Waiting (Per Call)
*72 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.

*73 Cancel Unconditional Call Forward

To cancel “Unconditional Call Forward”, dial “*73” and get the dial tone, then hang up.

*87 Blind Transfer

Please refer to Blind Transfer section.
*90 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.

*91 Cancel Busy Call Forward

To cancel “Busy Call Forward”, dial “*91” and get the dial tone, then hang up

*92 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.

*93 Cancel Delayed Call Forward

To cancel this Forward, dial “*93” and get the dial tone, then hang up

Flash/Hook

When in conversation, flash/hook switches to incoming call (call waiting is enabled).

When in conversation and no incoming call heard, flash/hook switches provides a new dial tone.

LED Light Pattern Indication

T

ABLE 6: HT488 LED DEFINITIONS

RED LED always indicates not normal status

DHCP Failed or WAN No Cable Button flashes every 2 seconds (if DHCP is configured)
HT- 488 fails to register Button flashes every 2 seconds (if SIP server is configured)
Firmware Upgrading Button flashes every 2 seconds
Device Malfunctions Red light steady on

GREEN LED mostly indicates normal working status

Message Waiting Indication Button flashes every 2 seconds
RINGING Button flashes at 1/10 second
RINGING INTERVAL Button flashes every second
In Conversation Green light steady on

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CONFIGURATION GUIDE

CONFIGURING HT488 THROUGH VOICE PROMPT

DHCP MODE

Follow Table 3 with voice menu option 01 to enable HT488 to use DHCP.

STATIC IP MODE

Follow Table 3 with voice menu option 01 to enable HT488 to use STATIC IP mode, then use option 02,
03, 04 to set up HT488’s IP, Subnet Mask, Gateway respectively.

TFTP SERVER ADDRESS

Follow Table 3 with voice menu option 06 to configure the IP address of the TFTP server.

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

1) always check, 2) check when pre/suffix changes, and 3) never upgrade

to choose firmware upgrade mode. There are three options:

CONFIGURING HT- 488 WITH WEB BROWSER

HT488 ATA 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 HT488 through a Web browser such as
Microsoft’s IE, AOL’s Netscape or Mozilla Firefox installed on Windows or Unix OS. (Macintosh OS is not
included).

Access the Web Configuration Menu

The HT488 HTML configuration page can be accessed via LAN or WAN ports.

• From the LAN port:

1. Directly connect a computer to the LAN port

2. Open a command window on the computer

3. Type in “ipconfig /release”, the IP address etc becomes 0

4. Type in “ipconfig /renew”, the computer gets an IP address in 192.168.2.x segment by
default

5. Open a web browser, type in the default IP address of the LAN port. http://192.168.2.1

1. You will see the log in page of the device.

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• From the WAN port:

1. Follow table 4 to find the WAN side IP address.

2. Open a web browser, type in the WAN side IP address – for example:

http://HT-488-WAN-IP-Address

Note:

• WAN side HTTP access is disabled by default for security reason. You can enable HTTP access
on the configuration page by setting “WAN side HTTP access” to be YES.

• Initial access to the configuration pages is always from the LAN port. The instructions are listed
above.

• The IVR announces 12 digits IP address, you need to strip out the leading “0” in IP address. For
ex. IP address: 192.168.001.014, you need to type in

http://192.168.1.14 in the web browser.

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: Password: Web pages allowed:

End User Level 123 Only Status and Basic Settings
Administrator Level admin Browse all pages

Only an administrator can access the “ADVANCED SETTING” configuration page. Please reference the
GUI pages using the following link:

Once this HTTP request is entered and sent from a Web browser, the HT488 will respond with the
following login screen:

FIGURE 3: SCREENSHOT OF CONFIGURATION LOG-IN PAGE

http://www.grandstream.com/user_manuals/GUI/GUI_HT488.rar.

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 an administrator can access the
“ADVANCED SETTING” configuration page.

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NOTE: If you can not log into the configuration page by using the default password, please check with
the VoIP service provider. It is 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 will respond with the
Configuration pages which are explained in details below.

TABLE 7: HT488 DEVICE STATUS PAGE DEFINITIONS

MAC Address
IP Address
Product Model
Software Version

System Uptime
Registered
PPPoE Link Up
NAT

The device ID, in HEX format. This is very important ID for ISP troubleshooting.
This field shows IP address of the HT-488.
This field contains the product model info, such as HT-488.

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 provider’s server.
This shows whether the PPPoE is up if connected to DSL modem
This shows what kind NAT the HT386 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 HT386 functioning correctly.

TABLE 8: HT488 BASIC SETTINGS PAGE DEFINITIONS

End User Password

Web Port
IP Address

DHCP hostname

DHCP domain

DHCP vendor class ID

Time Zone

Daylight Savings Time

This contains the password for end user to access the Web Configuration Menu. User
can put new password here. This field is case sensitive with maximum of 25 characters

This is the device’s internal HTTP server port. Default is 80.

• If DHCP mode is enabled, then all the field values for the Static IP mode are not
used (even though they are still saved in the Flash memor y.) The HT386 will acquire
its IP address from DHCP in the network.

• PPPoE settings is usually for DSL/ADSL modem users. The HT will attempt to
establish a PPPoE session if PPPoE account is set.

• If Static IP mode is selected, the IP address, Subnet Mask, Default Router IP
address, DNS Server 1 (mandatory), DNS Server 2 (optional) fields need to be
configured.

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.

This parameter controls how the displayed date/time will be adjusted according to the
specified time zone.

This parameter controls time displayed in daylight savings time. If set to “Yes”, then
the displayed time will be 1 hour ahead of normal time.

The “Optional Rule” is configured to automatically adjust the Daylight Savings Time
(DST) based on the rule set in this field. Rule Syntax:

• start-time;end-time;saving

• Both start-time and end-time have the same syntax:

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Cloned WAN MAC
Address:

LAN Subnet Mask

LAN DHCP Base IP:

DMZ IP:

Port Forwarding:

Number of rings

PSTN access code
PIN for PSTN calls
PIN for VoIP calls
Route Call to PSTN

Forward to PSTN

Forward to VoIP

• 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 “day” th iteration of the
weekday (e.g.: 1st Sunday, 3rd Tuesday etc).

If “weekday” is not zero and “day” is negative, then the daylight saving starts on the last
“day”th 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 is set for US, the “Automatic Daylight Saving Time Rule” shall be set
to “3,2,7,2,0;11,1,7,2,0;60”

Examples
US/Canada where daylight saving time is applicable:
03,02,7,02,00;11,1,7,02,00;60
This means the daylight saving time starts from the second Sunday of March at 2AM
and ends the first Sunday of November at 2AM. The saving is 60 minutes.

Allow the user to set a specific MAC address. Set in Hex format

Sets the LAN subnet mask. Default value is 255.255.255.0

Base IP for the LAN port, which functions as default gateway for its LAN. Default value
is 192.168.2.1

Forward all WAN IP traffic to a specific IP address if no matching port is used by
HandyTone-488 itself or in the defined port forwarding.

Allow users to forward a matching (TCP/UDP) port to a specific LAN IP address with a
specific (TCP/UDP) port.

Default is 4. It specifies number of phone rings before a PSTN incoming call is bridged
to VoIP

The code to access the PSTN line. Default is “*00”.
PIN code to bridge from VoIP to PSTN
PIN code to bridge from PSTN to VoIP
If the dialed digits match one of the specified prefix here, outbound calls will be initiated

from PSTN line. This field is especially useful for emergency calls.
Calls are unconditionally forwarded to the specified PSTN phone number for all

incoming VoIP calls on FXO port.
Calls are unconditionally forwarded to the specified VoIP phone numb er for all

incoming PSTN calls.

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ADVANCED CONFIGURATION AND FXS/FXO 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: HT488 ADVANCED SETTINGS PAGE DEFINITIONS

Admin Password

Home NPA
Layer 3 QoS

Layer 2 QoS

No Key Entry timeout
STUN Server
Keep-alive interval

Use NAT IP
Firmware Upgrade and

Provisioning
Firmware Server Path
Config Server Path
Firmware File Prefix

Firmware File Postfix
Config File Prefix
Config File Postfix
Automatic Upgrade
Firmware Key

Caller ID Scheme

Onhook Voltage
Polarity Reversal

NTP server

Syslog Server

Syslog Level

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- 488 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)

• ETSI-DTMF (Finland, Sweden)

• ETSI-FSK (France, Germany, Norway, Taiwan, UK-CCA)

Select the onhook voltage to suit different area or PBX.
Default is No. If set to Yes, polarity will be reversed upon call establishment

and termination.
URI or IP address of the NTP (Network Time Protocol) server, which the HT386 will

use to synchronize the date/time.
The IP address or URL of syslog server, especially useful for ITSP (Internet Telephone

Service Provider)
Select the ATA to report the log level. Default is NONE. The level is either 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)

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• 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

T

ABLE 10: HT488 FXS PORT SETTINGS PAGES DEFINITIONS

SIP Server

Outbound Proxy

SIP User ID

Authenticate ID

Authentication Password

Name
Use DNS SRV:
User ID is Phone Number

SIP Registration

Unregister on Reboot

Register Expiration

Local SIP port

Local RTP port

Use Random Port

This field contains the URI string or the IP address (and port, if different from 5060) of
the SIP proxy server. e.g., the following are some valid examples: sip.my-voip­provider.com, or sip:my-company-sip-server.com, or 192.168.1.200:5066

IP address or Domain name of Outbound Proxy, or Media Gateway, or Session Border
Controller. Used by ATA for firewall or NAT penetration in different network
environment. If symmetric NAT is detected, STUN will not work and ONLY Outbound
Proxy wi l l wo r k.

User account information, provided by VoIP service provider (ITSP), usually has the
form of digit similar to phone number or actually a phone number. 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.

Do NOT include the preceding “sip:” scheme or the host portion of the SIP address in
this field.

ID used for authentication, usually same as SIP user ID, but could be different and
decided by ITSP.

Password for ATA to register to (SIP) servers of ITSP. Purposely blank out once saved
for security. Maximum length is 25.

SIP service subscriber’s name which will be used for Caller ID display
Default is No. If set to Yes the client will use DNS SRV to lookup for the SIP server.
If “Yes” is set, a “user=phone” parameter will be attached to the “From” header in SIP

request
This parameter controls 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 all

previous bindings. Use only if proxy supports this remove bindings request .
This parameter allows the user to specify the time frequency (in minutes) the

HandyTone ATA refreshes its registration with the specified registrar. The default
interval is 60 minutes (or 1 hour). The maximum interval is 65535 minutes (about 45
days).

This parameter defines the local SIP port the HandyTone ATA will listen and transmit.
The default value for FXS port is 5060.

This parameter defines the local RTP-RTCP port pair the HandyTone ATA will listen
and transmit. 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 for FXS port is

5004.

Default No. If set to Yes, the device will pick randomly-generated SIP and RTP ports.

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This is usually necessary when multiple HandyTone ATAs are behind the same NAT.

DTMF Payload Type
Send DTMF

Send Flash Event
Enable Call Features

Use Bell-style
3-way Conference

Offhook
Auto-Dial

Proxy-Require
Disable Call Waiting
NAT Traversal (STUN)

No Key Entry Timeout

Preferred Vocoder The HandyTone ATA supports 6 different Vocoder t ypes including

Voice Frames per TX

G723 Rate:

This parameter sets the payload type for DTMF using RFC2833
This parameter specify 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. Multiple DTMF
transmission schema can be selected.

Default is NO. If set to yes, flash will be sent as DTMF event.
Default is Yes. Advance call features and feature codes functions are supported

locally.
Default setting is No. When it is set to yes, the user will use Bell-style to initiate

conference.
This parameter allows users to configure a User ID or extension number to be

automatically dialed upon offhook. Please note that only the user part of a SIP address
needs to be entered here. The HandyTone ATA will automatically append the “@” and
the host portion of the corresponding SIP address.
Note: Please write down the IP address of the ATA if you use this feature as it
will prevent you to access the IVR and the only way to access the device
configuration is via the web configuration page.

SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.
Default is No.
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 4 seconds.

1. G.711 A/µ law,

2. G.723.1,

3. G.726-32,

4. G.729A,

5. iLBC

Users can configure Vocoders in a preference list that will be included with the same
preference order in SDP message.

This field contains the number of voice frames to be transmitted in a single packet.
When setting this value, the user should be aware of the requested packet time (used
in SDP message) as a result of configuring this parameter. This parameter is
associated with the first vocoder in the vocoder preference list or the actual used
payload type negotiated between the 2 conversation parties at run time.

FOR EXAMPLE:

1. If the first vocoder is configured as G723 and the “Voice Frames per TX” is set to
2, then the “ptime” value in the SDP message of an INVITE request will be 60ms
because each G723 voice frame contains 30ms of audio.

2. Similarly, if this field is set to 2 a nd if the first vocoder chosen is G72 9 or G711 or
G726, then the “ptime” value in the SDP message of an INVITE request will be
20ms.

If the configured voice frames per TX exceeds the maximum allowed value, the
HandyTone ATA will use and save the maximum allowed value for the corresponding
first vocoder choice.

The maximum value for PCM is 10(x10ms) frames; for G726, it is 20 (x10ms) frames;
for G723, it is 32 (x30ms) frames; for G729/G728, 64 (x10ms) and 64 (x2.5ms) frames
respectively.

This defines the encoding rate for G723 vocoder. Default setting is 6.3kbps.

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iLBC frame size:
iLBC payload type:

Silence Suppression

Fax Mode
Early Dial
Dial Plan Prefix
Use # as

Send Key

Subscribe for MWI:

Send Anonymous

Lock keypad update
Refer-To Uses Target

Contact.
Special Features

Onhook Threshold

FXS Impedance
Caller ID Scheme

Onhook Voltage
Polarity Reversal
Volume Amplification

This sets the iLBC size in 20ms or 30ms
This defines payload type for iLBC. Default value is 97. The valid range is between 96

and 127.
This controls the silence suppression/VAD feature of G723. If set to “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 fax Pass-Through.
Default is No. Use only if proxy supports 484 response
Sets the prefix added to each dialed number
This parameter allows users to configure the “#” key to be used as the “Send” (or

“Dial”) key. If set to “Yes”, pressing this key will immediately trigger the sending of
dialed string collected so far. 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 will be
sent periodically.

If this parameter is set to “Yes”, user ID will be sent as anonymous, essentially blocking
the Caller ID from displaying.

If this parameter is set to “Yes”, the configuration update via keypad is disabled.
Used for Attended transfer Feature. Default is NO. If set to YES, the “Refer-To” header

uses the transferred target’s “Contact” header information.
Default is Standard. Choose the selection to meet some special requirements from Soft

Switch vendors like Nortel, Broadsoft, etc.
Default setting is 800ms. If the flash event is longer than the settings, it is processed as

on-hook event.
Selects the impedance of the analog telephone connected to the Phone port.
Select the Caller ID Scheme to suit the standard of different area.

• Bellcore (North America)

• CID - Canada

• DTMF (Brazil)

• DTMF (Sweden)

• DTMF (Denmark)

• ETSI-DTMF (Finland, Sweden)

• ETSI-FSK (France, Germany, Norway, Taiwan, UK-CCA)
Select the onhook voltage to suit the analog phone.
Select Polarity Reversal to adapt some call charge/billing system. Default is No.
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.

TABLE 11: HT488 FXO PORT SETTINGS PAGES DEFINITIONS

Local SIP port
Local RTP port
PSTN AC Termination
PSTN Disconnect Tone

PSTN Disconnect Tone
Cadence

PSTN Silence Timeout

The default value for FXO port is 5062.
The default value for FXO port is 5008.
Selects the impedance of the analog PSTN line connected to the Line port.
This configuration should be configured by the VoIP service provider. Some country

use single frequency tone to signal PSTN disconnection, some country use double
frequency tone.

This setting can be configured to suit the telephone company’s standard in different
country.

Terminate call after long silence detected. Default setting is 60 sec, max 65536

Note: General settings have same meaning as explained in above for FXS port page. Here are described
only a parameters related to FXO port.

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TABLE 12: HT488 CALL PROGRESS TONES SETTINGS PAGE DEFINITIONS

Call Progress Tones Using these settings, user can configure 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 tone, OFF should be zero. Otherwise it will ring ON ms and a pause of OFF ms and
then repeat the pattern.

S

AVING THE CONFIGURATION CHANGES

Once a change is made, users should click on the “Update” button in the Configuration page. The HT488
will display a confirmation screen to confirm that the changes have been saved. Click ‘Reboot’ to save
all changes. Please reference the GUI pages using the following link:

http://www.grandstream.com/user_manuals/GUI/GUI_HT488.rar.

Rebooting the HT488 from Remote

The administrator can remotely reboot the HT488 by clicking on the “Reboot” button at the bottom of the
configuration page. Once done, the following screen will be displayed to indicate that rebooting is
underway. You can login again after about 30 seconds.

FIGURE 4: SCREENSHOT OF REBOOTING SCREEN

Grandstream Device Configuration

The device is rebooting now...

You may relogin by clicking on the link below in 30 seconds.

Click to relogin

All Rights Reserved Grandstream Networks, Inc. 2004

NOTE: Interrupting the ‘booting up’ process could permanently damage the device.

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CONFIGURATION THROUGH A CENTRAL SERVER

The Grandstream HT488 can be automatically configured from a central provisioning sy stem.
When the HT488 boots up, it will send TFTP or HTTP request to download configuration file,

“cfg000b82xxxxxx”, where “000b82xxxxxx” is the LAN side MAC address of the HT488
The configuration files can be downloaded via TFTP or HTTP from the central server. A service provider

or an enterprise with large deployment of HT488 can easily manage the configuration and service
provisioning of individual devices remotely from a central server.

Grandstream provides a licensed provisioning system called GAPS that can be used to support
automated configuration of HT488. GAPS (Grandstream Automated Provisioning System) uses
enhanced (NAT friendly) TFTP or HTTP (thus no NAT issues) and other communication protocols to
communicate with each individual HT488 for firmware upgrade, remote reboot, etc.

Grandstream provide GAPS (Grandstream Automated Provisioning System) service to VoIP service
providers. It could be either simple redirection or with certain special provisioning settings. Initially upon
booting 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 customer’s TFTP or HTTP 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/1.0.3.64
e.g. 168.75.215.190

NOTES:

1. TFTP server in IP address format can be configured via IVR. Please refer to section
CONFIGURATION GUIDE for instructions. If TFTP server is in FQDN format, it must be set via
web configuration interface.

2. End users recommended using our TFTP server. Its address can be found at

http://grandstream.com/y-firmware.htm. Currently, the TFTP server, your HT-488 can be

upgraded from has an IP address 168.75.215.190. For companies, we recommend to maintain
their own TFTP/ HTTP server for upgrade and provisioning procedures.

3. 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 ATA’s SRAM. During
this stage, the HT ATA’s 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.

4. 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 Grandstream’s Web site to obtain our public TFTP server’s
IP address.

5. 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. Our latest official release can be

downloaded from http://www.grandstream.com/y-firmware.htm.

Directions 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 GXW400X 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 phone’s 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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The end-user 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”, the 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, however they are defined. This allows the device to 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, the Server Provider can spread the Firmware or Configuration File download in minutes
to reduce the Firmware or Provisioning Server load at any given time.

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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.

FACTORY RESET
IVR Command

Reset default factory settings using the IVR Prompt (Table 5):

1. Dial “***” for voice prompt.

2. Enter “99” and wait for “reset” voice prompt.

3. Enter the encoded MAC address (Look below on how to encode MAC address).

4. Wait 15 seconds and device will automatically reboot and restore factory settings.

Encoding the MAC Address

1.

Locate the MAC address of the device. It is the 12 digit HEX number on the bottom of the
unit.

2. Key in the MAC address. Use the following mapping:

0-9: 0-9
a.

A: 22 (press the “2” key twice, “A” will show on the LCD)
b. B: 222
c.

C: 2222
d. D: 33 (press the “3” key twice, “D” will show on the LCD)
e. E: 333
f.

F: 3333

For example: if the MAC address is 000b8200e395, it should be keyed in as “0002228200333395”.

NOTE:

1. Factory Reset will be disabled if the “Lock keypad update” is set to “Yes”.

2. Please be aware by default the GXW400X WAN side HTTP access is disabled. After a factory reset,
the device’s web configuration page can be accessed only from its LAN port.

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GLOSSARY OF VOIP 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
control the
real world conditions.

ARP Address Resolution Protocol is a protocol used by the
IPv4, to map
operates below the network layer as a part of the interface between the OSI network and OSI link layer. It
is used when

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's 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
communications to fill the

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-sam pling.

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

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 #).

gain of a system in order to maintain some measure of performance over a changing range of

IP network addresses to the hardware addresses used by a data link protocol. The protocol

IPv4 is used over Ethernet

silent time in a transmission resulting from voice activity detection.

electronic system found in many types of devices. Its purpose is to

Internet Protocol (IP) [RFC826], specifically

noise used in radio and wireless

Internet service that translates domain

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FQDN Fully Qualified Domain Name. A FQDN consists of a host and domain name, including top-level
domain. For example,
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

echo from a voice communication in order to improve voice quality on a telephone call. In addition to

improving quality, this process improves
preventing echo from traveling across a
acoustic echo and hybrid echo.
contribute to echo generation in

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.

IVR IVR is a software application that accepts a combination of voice telephone input and touch-tone
keypad selection and provides appropriate responses in the form of voice, fax, callback, e-mail and
perhaps other media.

www.grandstream.com is a fully qualified domain name. www is the host,

telephony to describe the process of removing

bandwidth savings achieved through silence suppression by

network. There are two t ypes of echo of relevance in telephony:

Speech compression techniques and digital processing delay often

telephone networks.

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MTU A Maximum Transmission Unit (MTU) is the largest size packet or frame, specified in octets (eight-
bit bytes), that can be sent in a packet- or frame-based network such as the Internet. The maximum for
Ethernet is 1500 byte.

NAT Network Address Translation
NTP Network Time Protocol, a protocol to exchange and synchronize time over networks The port used

is UDP 123 Grandstream products using NTP to get time from Internet
OBP/SBC Outbound Proxy or another name Session Border Controller. A device used in

OBP/SBCs are put into the signaling and media path between calling and called Caller. The OBP/SBC
acts as if it was the called VoIP phone and places a second call to the called Caller. The effect of this
behavior is that not only the signaling traffic, but also the media traffic (voice, video etc) crosses the
OBP/SBC. Without an OBP/SBC, the media traffic travels directly between the VoIP phones. Private
OBP/SBCs are used along with
Public VoIP service providers use OBP/SBCs to allow the use of VoIP protocols from private networks

internet connections using NAT.

with
PPPoE Point-to-Point Protocol over Ethernet is a network protocol for encapsulating PPP frames in

Ethernet frames. It is used mainly with cable modem and DSL services.
PSTN Public Switched Telephone Network. The phone service we use for every ordinary phone call, or

called POT (Plain Old Telephone), or circuit switched network.
RTCP Real-time Transport Control Protocol, defined in

Transport Protocol (RTP), It partners RTP in the delivery and packaging of multimedia data, but does not

transport any data itself. It is used periodically to transmit control packets to participants in a streaming
multimedia session. The primary function of RTCP is to provide feedback on the quality of service being
provided by RTP.

RTP Real-time Transport Protocol defines a standardized packet format for delivering audio and video
over the Internet. It was developed by the Audio-Video Transport Working Group of the
published in 1996 as

SDP Session Description Protocol is a format for describing
has been published by the

SIP Session Initiation Protocol, An IP telephony signaling protocol developed by the IETF (RFC3261).
SIP is a text-based protocol suitable for integrated voice-data applications. SIP is designed for voice
transmission and uses fewer resources and is considerably less complex than H.323. All Grandstream
products are SIP based

STUN Simple Traversal of UDP over NATs is a
NATs) to find out its public address, the type of NAT it is behind and the internet side port associated by
the NAT with a particular local port. This information is used to set up UDP communication between two
hosts that are both behind NAT routers. The protocol is defined in
with non-symmetric NAT routers.

TCP Transmission Control Protocol is one of the core protocols of the
applications on networked hosts can create connections to one another, over which they can exchange
data or

TFTP Trivial File Transfer Protocol, is a very simple
basic form of

packets. The protocol guarantees reliable and in-order delivery of sender to receiver data.

FTP; It uses UDP (port 69) as its transport protocol.

RFC 1889

firewalls to enable VoIP calls to and from a protected enterprise network.

RFC 3550, a sister protocol of the Real-time

streaming media initialization parameters. It

IETF as RFC 2327.

network protocol allowing clients behind NAT (or multiple

RFC 3489. STUN will usually work well

Internet protocol suite. Using TCP,

file transfer protocol, with the functionality of a very

VoIP networks.

IETF and first

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UDP User Datagram Protocol (UDP) is one of the core protocols of the Internet protocol suite. Using
UDP, programs on networked computers can send short messages known as
UDP does not provide the reliability and ordering guarantees that
order or go missing without notice. However, as a result, UDP is faster and more efficient for many
lightweight or time-sensitive purposes.

VAD Voice Activity Detection or Voice Activity Detector is an algorithm used in
wherein, the presence or absence of human speech is detected from the audio samples.

VLAN A virtual
on a single physical

VoIP Voice over the Internet. VoIP encompasses many protocols. All the protocols do some form of
signaling of call capabilities and transport of voice data from one point to another. e.g.: SIP, H.323, etc.

LAN, known as a VLAN, is a logically-independent network. Several VLANs can co-exist

switch. It is usually refer to the IEEE 802.1Q tagging protocol.

TCP does; datagrams may arrive out of

datagrams to one another.

speech processing

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