Grandstream Networks HandyTone-488 User manual

User Manual
HandyTone-488
Analog Telephone Adaptor
For SW Release Version 1.0.3.44
Grandstream Networks, Inc.
www.grandstream.com

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Table of Contents
1WELCOME.....................................................................................................................4
2INSTALLATION............................................................................................................ 5
3WHAT IS INCLUDED IN THE PACKAGE...............................................................7
3.1 SAFETY COMPLIANCES................................................................................................7
3.2 WARRANTY.................................................................................................................7
4PRODUCT OVERVIEW...............................................................................................8
4.1 KEY FEATURES............................................................................................................ 8
4.2 HARDWARE SPECIFICATION ........................................................................................9
5BASIC OPERATIONS.................................................................................................10
5.1 GET FAMILIAR WITH VOICE PROMPT.........................................................................10
5.2 MAKE PHONE CALLS................................................................................................. 11
5.2.1 Calling phone or extension numbers via VoIP FXS port ..................................11
5.2.2 Direct IP calls....................................................................................................11
5.2.3 Call hold ............................................................................................................12
5.2.4 Call waiting .......................................................................................................12
5.2.5 Blind Transfer....................................................................................................12
5.2.6 Attended Transfer ..............................................................................................13
5.2.7 3-way Conferencing...........................................................................................13
5.2.8 PSTN Pass Through/life line .............................................................................13
5.2.9 VoIP-to-PSTN Calls...........................................................................................14
5.2.10 PSTN-to-VoIP Calls...........................................................................................14
5.2.11 Route Calls to PSTN..........................................................................................15
5.3 CALL FEATURES........................................................................................................15
5.3.1 Call Features Table...........................................................................................15
5.4 FAX ...........................................................................................................................16
5.5 LED LIGHT PATTERN INDICATION ............................................................................16
6CONFIGURATION GUIDE........................................................................................18
6.1 CONFIGURING HANDYTONE-488 WAN IP THROUGH VOICE PROMPT.......................18
6.1.1 DHCP Mode ......................................................................................................18
6.1.2 STATIC IP Mode................................................................................................18
6.1.3 TFTP Server Address.........................................................................................18
6.2 CONFIGURING HANDYTONE-488 WITH WEB BROWSER ............................................ 18
6.2.1 Access the Web Configuration Menu.................................................................18
6.2.2 End User Configuration ....................................................................................19
6.2.3 Advanced User Configuration...........................................................................25
General settings have same meaning as explained in above section for FXS port page. Special
settings on FXO port are explained below...................................................................... 36
6.2.4 Saving the Configuration Changes.................................................................... 37
6.2.5 Rebooting the HandyTone-488 from Remote ....................................................37

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6.3 CONFIGURATION THROUGH A CENTRAL SERVER....................................................... 38
7SOFTWARE UPGRADE.............................................................................................39
7.1 FIRMWARE UPGRADE THROUGH TFTP/HTTP...........................................................39
7.2 CONFIGURATION FILE DOWNLOAD ........................................................................... 40
7.3 FIRMWARE AND CONFIGURATION FILE PREFIX AND POSTFIX....................................40
7.4 MANAGING FIRMWARE AND CONFIGURATION FILE DOWNLOAD ..............................40
8RESTORE FACTORY DEFAULT SETTING..........................................................41
9GLOSSARY OF TERMS.............................................................................................42

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1Welcome
Congratulations on becoming an owner of HandyTone-488. You made an excellent choice and we
hope you enjoy all of its capabilities.
Grandstream's HandyTone-488 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
HandyTone-488 is fully compatible with SIP industry standard and can interoperate with many other
SIP compliant devices and software on the market.
Grandstream HandyTone-488 is a new addition to the popular HandyTone product family. It is an
enhanced model compared to the award-winning HandyTone-486 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 HandyTone-488 has been awarded the Best of Show product in 2005 Internet Telephony
Conference and Expo.

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2Installation
HandyTone-488 Analog Telephone Adaptor is an all-in-one VoIP integrated device designed to be a
total solution for networks providing VoIP services.
The HandyTone-488 VoIP functionalities are available via a regular analog telephone.
The following photo illustrates the appearance of a HandyTone-488.
Top View Side Views
Interconnection Diagram of the HandyTone-488:
Fax
Cordless Phone
Analog Phone
Internet ADSL/Cable
Modem Ethernet
PSTN
PC PCPC
LAN
WAN
FXO FXS
BUTTON
RED LED
GREEN LED
RJ45
10M Ethernet
LAN - WAN
RJ11
FXS Port
(Phone)
+5V/1200mA
RJ11
FXO Port
(Phone Line)

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HandyTone-488 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 HandyTone-488 as it supports
simultaneous call on both FXS port and FXO port. Telephone calls can be originated from or
terminated on the PSTN network via FXO port remotely.
Following are the steps to install a HandyTone-488:
1. Connect a standard touch-tone analog telephone 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 HandyTone-488 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.
Please follow the instructions in section 6.2 to configure the HandyTone-488.

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3What is Included in the Package
The HandyTone-488 package contains:
1) One HandyTone-488
2) One universal power adaptor
3) One Ethernet cable
3.1 Safety Compliances
The HandyTone-488 is compliant with various safety standards including FCC/CE and C-tick. Its
power adaptor is compliant with UL standard. The HandyTone-488 should only operate with the
universal power adaptor provided in the package.
3.2 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-488 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..

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4Product Overview
4.1 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
•Supports call origination and termination from/to the PSTN network (via FXO port)
•Powerful digital signal processing (DSP) to ensure superb audio quality; advanced adaptive
jitter control and packet loss concealment technology
•Support various vocoders 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 advanced call features on FXS port, includes Caller ID/Name display or block, Hold,
Call Waiting/Flash, Call Transfer, Call Forward, 3-way conference, in-band and out-of-band
DTMF, etc.
•Support fax pass through (for PCMU and 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)
•Support automated NAT traversal without manual manipulation of firewall/NAT
•Support device configuration via built-in IVR, Web browser or central configuration file
through TFTP or HTTP server
•Support firmware upgrade via TFTP or HTTP
•Support PSTN pass through
•Ultra compact (wallet size) and lightweight design, great companion for travelers
•Compact, lightweight Universal Power adapter.

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4.2 Hardware Specification
The table below lists the hardware specification of HandyTone-488.
Model HandyTone-488
LAN interface 1xRJ45 10Base-T
WAN interface 1xRJ45 10Base-T
FXS telephone port 1xFXS
FXO port 1xFXO
Button 1
LED Green and red color
Universal Switching
Power Adaptor Input: 100-240VAC 50-60 Hz Output: +5VDC, 1200mA,
UL certified
Dimension 70mm (W)
130mm (D)
27mm (H)
Weight 0.6lbs (0.3kg)
Temperature 40 - 130oF
5 – 45oC
Humidity 10% - 90%
(non-condensing)
Compliance

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5Basic Operations
5.1 Get Familiar with Voice Prompt
HandyTone-488 has stored 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 or dial
“***” from the analog phone. The following table shows how to use the voice prompt menu to
configure the device.
Menu Voice Prompt Options
Main Menu “Enter a Menu Option” Enter “*” for the next menu option
Enter “#” to return to the main menu
Enter 01-06, 47, 86, 99 menu option
01 “DHCP Mode”,
“Static IP Mode” 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.
02 “IP Address “ + IP address The current WAN IP address is announced
Enter 12 digit new IP address if in Static IP
Mode
03 “Subnet “ + IP address Same as menu 02
04 “Gateway “ + IP address Same as menu 02
05 “DNS Server “ + IP address Same as menu 02
06 “TFTP Server “ + IP address Same as menu 02
47 “Direct IP Calling” When entered, you will be prompted a dial
tone, then enter 12 digit IP address
This menu can also be entered by pressing the
button again
(For details, see “4.2.2 Make a Direct IP
Call”.)
99 “RESET” Enter “9” to reboot the phone
Enter encoded MAC address to restore factory
default setting
“Invalid Entry” Automatically returns to main menu
Notes:
•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

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•“#” 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 of the digits are collected, they will be processed.
•For IP address input, ignore the dot and key in the digit directly, add 0 before octet with less
than three digits. e.g.: IP: 192.168.1.10, key in: 192 168 001 010
•Key entry can not be deleted but the phone may prompt error once it is detected
5.2 Make Phone Calls
5.2.1 Calling phone or extension numbers via VoIP FXS port
There are currently two methods to make an extension number call:
a) Dial the numbers directly and wait for 4 (default) seconds.
b) Dial the numbers directly, and press # (assuming that “use # as dial key” is selected in web
configuration).
Examples:
To dial another extension on the same proxy, such as 1008, simply pick up 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.
5.2.2 Direct IP calls
Direct IP calling allows two parties, that is, a HandyTone ATA 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 HandyTone ATA and other VoIP Device(i.e., another HandyTone ATA or Budgetone SIP
phone or other VoIP unit) have public IP addresses, or
•both HandyTone ATA and other VoIP Device are on the same LAN using private IP addresses,
or
•Both HandyTone ATA and other VoIP Device can be connected through a router using public
or private IP addresses (with necessary port forwarding or DMZ).
To make a direct IP to IP call, first pick up the analog phone or turn on the speakerphone on the analog
phone, then follow Section 4.1 with voice prompt 47, followed by the 12-digit target IP address. User
will hear a voice prompt “Direct IP Calling” and a dial tone. Enter a 12-digit target IP address to make
a call. Destination ports can be specified by using “*4” (encoding for “:”) followed by the port
number.
Examples:

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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 seconds.
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.
5.2.3 Call hold
This function is applicable on FXS port for VoIP calls only. While in conversation, pressing the
“FLASH” button on the phone will put the remote end on hold. Pressing the “FLASH” button again
will release the previously Hold state and the bi-directional media resume.
5.2.4 Call waiting
This function is applicable on FXS port for VoIP calls only. If call waiting feature is enabled, while the
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 essentially becomes toggling between two active calls.
5.2.5 Blind Transfer
This function is applicable on FXS port for VoIP calls only. Assume that call party A and B are in
conversation. A wants to Blind Transfer B to C:
1. A press “FLASH” on the analog phone to hear the dial tone.
2. Then A dials *87 then dials C’s number, and then #(or wait for 4 seconds)
3. A can hang up.
NOTES: “Enable Call Feature” has to be set to YES in web configuration page. “Send flash event”
needs to be set to NO.
A can hold on to the phone and wait one of the three following behaviors:
•A quick confirmation tone (temporarily using the call waiting indication tone) follows by a
dial tone. This indicates the transfer has been successful.
•A quick busy tone followed by a restored call (On supported platforms only). This means that
B is transfered to C with no success. The busy tone is just to indicate A that the transfer has
failed.
•Busy tone keeps playing. This means A have failed to receive the second NOTIFY from B
and decide to time out. Note: this does not indicate the transfer has been successful, nor does
it indicate the transfer has failed. When B is a client that does not support the second NOTIFY,
this will be the case. In bad network scenarios, this could also happen, although the transfer
may have been completed successfully.

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5.2.6 Attended Transfer
This function is applicable on FXS port for VoIP calls only. 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 to get a dial tone
2. A then dial C’s number followed by # (or wait for 4 seconds).
3. If C answers the call, A and C are in conversation. Then A hang up to complete transfer.
4. If C does not answer the call, A can press “FLASH” back to talk to B.
NOTES:
•When antended Transfer failed and A hang up, the HandyTone 488 will ring user A back again
to remind A that B is still on the call. A can pick up the phone to restore conversation with B.
5.2.7 3-way Conferencing
This function is applicable on FXS port for VoIP calls only. Assuming that call party A and B are in
conversation. A wants to bring C in a conference:
Bell style conference:
1. A press “FLASH” button on the analog phone to get a dial tone.
2. A dials C’s number followed by # (or wait for 4 seconds).
3. If C answers the call, then A press “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.
NOTES: A is the call initiator for both calls with B and C.
User feature code *23:
1. A press “FLASH” button on the analog phone to get a dial tone.
2. A dials *23 followed by C’s number and # (or wait for 4 seconds).
3. If C answers the call, then A press “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.
NOTES: “Enable Call Feature” has to be set to YES in web configuration page.
5.2.8 PSTN Pass Through/life line
HandyTone-488 supports PSTN pass through. The user can send and receive PSTN call with attached
analog phone.
To receive PSTN calls, simply make phone off hook when the analog phone rings.

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To make a PSTN call, simply press the PSTN access code (*00 is default, or any number configured in
web configuration page) to switch to the PSTN line and get dial tone, then dial the PSTN number.
When HandyTone-488 is out of power, it will function as a jack. The user will be automatically
connected to the PSTN Line.
5.2.9 VoIP-to-PSTN Calls
This function is applicable on FXO port that functions as a bridge between VoIP and PSTN. The user
can remotely use PSTN line to initiate a call.
To 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. Then the caller hears either a special continuous tone or a dial tone. The
special continuous tone is played if the pin code is configured, or the dial tone otherwise.
2. Enter in the pin code that is configurable on the configuration page. The caller will hear the dial
tone and get connected to the PSTN line if the pin code is valid, otherwise the continuous tone
is played again to prompt caller to enter in the pin code again. The use may try up to 3 times to
enter in pin code, if none is valid, HT488 will hang up.
3. After the caller hears dial tone from PSTN line, the caller can start dialing number to make
calls.
Note:
•Users can choose whether apply password protection for VoIP-to-PSTN calls or not. A PIN
(Pin for PSTN calls) consists of up to 8 numeric digits can be configured through 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.
•Upon hearing the special continuous tone for PIN code input, if the caller don’t enter any digit,
HT488 will time out and hang up the call in 10 seconds. During any stage of DTMF digits
input, a 4 seconds timeout is applied to serve as an end of PIN or destination number input.
Users may also use the “#” key to indicate the end of an input.
•On the web configuration page, if the “Forward to PSTN” is configured, the second stage
dialing is eliminated, i.e., after dialing into the FXO SIP account number, the PSTN number
will be called automatically.
5.2.10 PSTN-to-VoIP Calls
This function is applicable on FXO port that functions as a bridge between VoIP and PSTN. The user
can make VoIP calls remotely by dialing into FXO Line port on HT488.
To make a PSTN-to-VoIP call:

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1. Make an incoming call to the PSTN line on FXO port. The attached analog phone will ring for
4 times by default, this setting is configurable on the configuration page.
2. If no one picks up the phone on FXS port after 4 rings, then the caller hears either a special
continuous tone or a dial tone. The continuous tone is played if the pin code is configured, or
the dial tone otherwise.
3. Enter in the pin code that is configurable on the configuration page. The caller will hear the dial
tone and get bridged to VoIP if the pin code is valid, otherwise the continuous tone is played
again to prompt caller to enter in the pin code again. The use may try up to 3 times to enter in
pin code, if none is valid, 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 apply password protection for PSTN-to-VoIP calls or not. A PIN
(Pin to VoIP calls) consists of up to 8 numeric digits can be configured through 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 VoIP SIP account on FXO port.
•Upon hearing the special continuous tone for PIN code input, if the caller don’t enter any digit,
HT488 will time out and hang up the call in 10 seconds. During any stage of DTMF digits
input, a 4 seconds timeout is applied to serve as an end of PIN or destination number input.
Users may also use the “#” key to indicate the end of an input.
•On the web configuration page, if the “Forward to VoIP” is configured, the second stage
dialing is eliminated, i.e., after bridging to VoIP, the configured VoIP number will be called
automatically.
5.2.11 Route Calls to PSTN
This function is applicable on FXO port that can access the PSTN network. By default, HT488 is in
VoIP mode upon off-hook. If “Route call to PSTN” is configured, certain calls will be initiated from
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”
on BASIC SETTINGS web configuration page. If the dialed digits match one of the specified prefix,
outbound calls will be initiated from PSTN line.
For ex, if “Route call to PSTN” is configured to be 626, all outgoing calls start with 626 will be
initiated from PSTN line.
5.3 Call Features
All the call feature codes are applicable to FXS port for VoIP calls only.
5.3.1 Call Features Table
Following table shows the call features of HandyTone-488.

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Key Call Features
*30 Block Caller ID (for all subsequent calls)
*31 Send Caller ID (for all subsequent calls)
*67 Block Caller ID (per call)
*82 Send Caller ID (per call)
*50 Disable Call Waiting (for all subsequent calls)
*51 Enable Call Waiting (for all subsequent calls)
*70 Disable Call Waiting. (Per Call)
*71 Enable Call Waiting (Per Call)
*72 Unconditional Call Forward.
To use this feature, dial “*72” and get the dial tone. Then dial
the forward number and “#” for a dial tone, then hang up.
*73 Cancel Unconditional Call Forward
To cancel “Unconditional Call Forward”, dial “*73” and get
the dial tone, then hang up.
*90 Busy Call Forward
To use this feature, dial “*90” and get the dial tone. Then dial
the forward number and “#” for a dial tone, then 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” and get the dial tone. Then dial
the forward number and “#” for a dial tone, then 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, this action will switch to the new
incoming call if there is a call waiting indication.
When in conversation without an incoming call, this action
will switch to a new channel for a new call.
5.4 Fax
HandyTone-488 supports FAX in two modes: T.38 (Fax over IP) and fax pass through. T.38 is the
preferred method because it is more reliable and works well in most network conditions. If the service
provider supports T.38, please use this method by selecting Fax mode to be T.38. 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 will need to select all the Preferred Codecs to be PCMU/PCMA.
5.5 LED Light Pattern Indication
Following tables show the LED light pattern indication.

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RED LED always indicates not abnormal status
DHCP Failed or WAN No Cable Button flashes every 2 seconds (if DHCP is
configured)
HandyTone-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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6Configuration Guide
6.1 Configuring HandyTone-488 WAN IP through Voice Prompt
6.1.1 DHCP Mode
Follow section 5.1 with voice menu option 01 to enable HandyTone-488 to use DHCP.
6.1.2 STATIC IP Mode
Follow section 5.1 with voice menu option 01 to enable HandyTone-488 to use STATIC IP mode, then
use option 02, 03, 04, 05 to set up HandyTone-488’s IP, Subnet Mask, Gateway, and DNS server
respectively.
6.1.3 TFTP Server Address
Follow section 5.1 with voice menu option 06 to configure the IP address of the TFTP server.
6.2 Configuring HandyTone-488 with Web Browser
HandyTone series 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 HandyTone-488 through a Web
browser such as Microsoft’s IE and AOL’s Netscape.
6.2.1 Access the Web Configuration Menu
The HandyTone-488 HTML configuration page can be accessed via LAN or WAN port:
•From the LAN port:
Directly connect a computer to the LAN port.
Open a command window on the computer
Type in “ipconfig /release”, the IP address etc becomes 0.
Type in “ipconfig /renew”, the computer gets an IP address in 192.168.2.x segment by
default
Open a web browser, type in the default gateway IP address. You will see the log in
page of the device.
http://192.168.2.1
•From the WAN port:
Follow section 5.1 to find out the WAN side IP address.
Open a web browser, type in the WAN side IP address.

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http://HandyTone-WAN-IP-Address,
Note:
•WAN side HTTP access is by default disabled for security reason. You can enable it on the
configuration page by setting “WAN side HTTP access” to be YES. The very first time to
access the configuration page is always from 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.
6.2.2 End User Configuration
Log in page.
Grandstream Device Configuration
Password
Login
All Rights Reserved Grandstream Networks, Inc. 2005
The password is case sensitive with maximum length of 25 characters. The factory default password
for End User and administrator is “123” and “admin” respectively. Only administrator can get access
to “ADVANCED SETTING” configuration page.
Note:
•If you can not log into the configuration page by using default password, please check with the
voip service provider. Most likely the service provider has provisioned the device and changed
the login password.
Status Page
Grandstream Device Configuration
STATUS BASIC
SETTINGS ADVANCED SETTINGS FXS PORT FXO PORT
MAC Address: 00.0B.82.03.DA.36
WAN IP Address: 10.10.11.225
Product Model: HT488
Software Version: Program-- 1.0.3.44 Bootloader-- 1.0.8.11 HTML-- 1.0.3.44 VOC-- 1.0.0.10

HandyTone-488 User Manual Grandstream Networks, Inc.
20
System Up Time: 0 day(s) 1 hour(s) 47 minute(s)
Re
g
istered: Yes
PPPoE Link Up: disabled
NAT: detected NAT type is full cone
All Rights Reserved Grandstream Networks, Inc. 2005
MAC Address The unique device ID, in HEX format. This is very important ID for ISP
troubleshooting.
WAN IP Address This field shows WAN port IP address.
Product Model This field contains the product model info, such as HT488
Software Version Program: This is the main software release. This number is always used for
firmware upgrade. Current release is 1.0.3.44.
Bootloader: current version is 1.0.8.11.
HTML: current version 1.0.3.44
VOC: current version is 1.0.0.10
System Uptime This shows system up time since last reboot.
Registered Whether the unit is registered to the SIP server.
PPPoE Link Up This shows whether the PPPoE link is up if connected to DSL modem
NAT This shows what kind of NAT the HandyTone is behind. NAT detection
mechanism is based on STUN protocol.
Basic settings page
Grandstream Device Configuration
STATUS BASIC
SETTINGS ADVANCED SETTINGS FXS PORT FXO PORT
End User
Password: (purposely not displayed for security protection)
Web Port: 80 (default for HTTP is 80)
IP Address: dynamically assigned via DHCP (default) or PPPoE
(will attempt PPPoE if DHCP fails and following is non-blank)
DHCP hostname:
DHCP domain: