User Manual
HandyTone-488
Analog Telephone Adaptor
For SW Release Version 1.0.0.4
Grandstream Networks, Inc.
HandyTone-488 User Manual
Grandstream Networks, Inc.
1 Welcome
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.
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2 Installation
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
RJ45
10M Ethernet
LAN - WAN
RJ11
FXS Port
(Phone)
+5V/1200mA
RJ11
FXO Port
(Phone Line)
BUTTON
RED LED
GREEN LED
Interconnection Diagram of the HandyTone-488:
Internet ADSL/Cable
Modem Ethernet
Analog Phone
WAN
FXS
FXO
PSTN
Cordless Phone
LAN
PC
PC
PC
Fax
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HandyTone-488 has one FXS port and one FXO port. The RJ11 jack next to the LAN port
is a FXS port. The RJ11 jack 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. Telephone calls can be originated or terminated from/to the PSTN network via FXO port.
Following are the steps to install a HandyTone-488:
1. Connect a standard touch-tone analog telephone to the FXS port.
2. Insert a standard RJ11 telephone cable into the FXO 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.
5. Insert the power adapter into the HandyTone-488 and connect it to a wall outlet.
Please follow the instructions in section 6.2.1 to configure the HandyTone-488.
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3 What 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.
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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4 Product 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 (a-law and u-law), G.723.1 (5.3K/6.3K),
G.726 (40K/32K/24K/16K), as well as G.728, G.729A/B, and iLBC.
• Support Caller ID/Name display or block, Hold, Call Waiting/Flash, Call Transfer, Call
Forward, in-band and out-of-band DTMF, Dial Plans, etc.
• Support fax pass through
• Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise
Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control)
• Support standard encryption and authentication (DIGEST using MD5 and MD5-sess)
• Support for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS)
• Support automated NAT traversal without manual manipulation of firewall/NAT
• Support device configuration via built-in IVR, Web browser or central configuration file
through TFTP or HTTP
• Support firmware upgrade via TFTP or HTTP with encrypted configuration files.
• 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 describes the difference among these models.
Model
HandyTone-488
LAN interface
WAN interface
FXS telephone port
FXO port
1xRJ45 10Base-T
1xRJ45 10Base-T
1xFXS
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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5 Basic Operations
5.1 Get Familiar with Voice Prompt
HandyTone-488 has stored a voice prompt menu for quick browsing and simple
configuration.
To enter this voice prompt menu, simply press the button or “***” from the analog phone.
Menu
Voice Prompt
User’s 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
Enter “9” to toggle the selection
01
02
“DHCP Mode”,
“Static IP Mode”
“IP Address “ + IP address
The current WAN IP address is
announced
Enter 12 digit new IP address if in Static IP
Mode
03
04
05
06
47
“Subnet “ + IP address
“Gateway “ + IP address
“DNS Server “ + IP address
“TFTP Server “ + IP address Same as menu 02
“Direct IP Calling”
Same as menu 02
Same as menu 02
Same as menu 02
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
86
99
“Voice Messages Pending”
“No Voice Messages”
“RESET”
Enter “9” to dial pre-configured phone
number to retrieve VM
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
• “*” always shifts down to the next menu option
• “#” always 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, the input will be processed.
• Key entry can not be deleted but the phone may prompt error once it is detected
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5.2 Make Phone Calls
5.2.1 Calling phone or extension numbers
To make a phone or extension number call:
a) Dial the number directly and wait for 4 seconds (default “No Key Entry Timeout”) .
Or
b) Dial the number directly, and press # (assuming that “Use # as dial key” is selected
in web configuration).
Other functions available during the call are call-waiting/flash, call-transfer, and call-forward.
5.2.2 Direct IP calls
Direct IP calling allows two phones, that is, a HandyTone with an analog phone and another
VoIP Device, to talk to each other in an ad hoc fashion without a SIP proxy. VoIP calls can be
made between two phones if:
• Both HandyTone ATA and the other VoIP device (i.e., another HandyTone ATA or
other SIP products) have public IP addresses, or
• Both HandyTone ATA and the other VoIP device (i.e., another HandyTone ATA or
other SIP produces) are on the same LAN using private or public IP addresses, or
• Both HandyTone ATA and the other VoIP device (i.e., another HandyTone ATA or
other SIP products) can be connected through a router using public or private IP
addresses.
To make a direct IP call, first pick up the analog phone or turn on the speakerphone on the
analog phone, then follow Section 5.1 with voice prompt 47, followed by the 12-digit target IP
address. Destination ports can be specified by using “*4” (encoding for “:”) followed by the
encoded port number. Following is a table of the encoding scheme for the most commonly
used characters:
Input
00
01
02
03
04
05
06
07
08
09
*0
Meaning
0
1
2
3
4
5
6
7
8
9
. (dot character)
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*4
: (column character)
Examples:
If the target IP address is 192.168.0.160, the dialing convention is
Voice Prompt with option 47, then 192168000160
followed by pressing the “#” key if it is configured as a send key or wait 4 seconds. In this
case, the default destination port 5060 is used if no port is specified.
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 Blind Transfer
Assuming that call party A and B are in conversation. A wants to Blind Transfer B to C:
1. A presses FLASH (on the analog phone, or Hook Flash for old model phones) to get a
dial tone.
2. Then A dials *87 then dials C’s number, and then #(or wait for 4 seconds)
3. A can hang up.
Note: Call Feature has to be set to YES.
A can hold on to the phone and wait for one of the three following behaviors:
• A quick confirmation tone (temporarily using the call waiting indication tone) followed
by a dial tone. This indicates the transfer is successful (transferee has received a 200
OK from transfer target). At this point, A can either hang up or make another call.
• 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 is just to indicate to the transferor that the transfer has
failed.
• Busy tone keeps playing. This means we have failed to receive the second NOTIFY
from the transferee and decided to time 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 will be the case. In bad network scenarios, this could also happen, although the
transfer may have been completed successfully.
5.2.4 Attended Transfer
Assuming that call party A and B are in conversation. A wants to Attend Transfer B to C:
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1. A presses FLASH (on the analog phone, or Hook Flash for old model phones) to get a
dial tone
2. A then dial C’s number then # (or wait for 4 seconds). A and C now in conversation.
3. A can hang up.
Note: When intended Transfer failed, if A hangs up, the HandyTone-488 will ring user A
again to remind A that B is still on the call, by pressing FLASH or Hook again will restore the
conversation between A and B.
5.2.5 Send and Receive PSTN Calls
Users can send and receive calls from PSTN. To receive PSTN calls, simply take the phone
off hook when the analog phone rings. To make a PSTN call, first press *00 (or your own
PSTN Access Code) to get the PSTN line dial tone and dial the PSTN number.
5.2.6 VoIP-to-PSTN Calls
To make a VoIP-to-PSTN call, users need to dial the FXO SIP account phone number first. A
ring tone is played once followed by a dial tone. At this time, users can dial a PSTN
telephone number or a mobile telephone number then # (or wait for 4 seconds). The call will
be established afterwards. If no PSTN number is entered after the dial tone, HandyTone-488
will hang up automatically in 10 seconds.
5.2.7 PSTN-to-VoIP Calls
To make a PSTN-to-VoIP call, PSTN callers need to originate a call to the FXO port
telephone number first. If no one answers the FXS phone after 4 ring tones, a dial tone is
played. At this time, users can dial a VoIP telephone number then # (or wait for 4 seconds).
The call will be established afterwards. If no VoIP number is entered after the dial tone,
HandyTone-488 will hang up automatically in 10 seconds.
5.3 Call Features
5.3.1 Call Features Table
Following table shows the call features of HandyTone-488.
Key
*30
*31
*67
*82
Call Features
Block Caller ID (for all subsequent calls)
Send Caller ID (for all subsequent calls)
Block Caller ID (per call)
Send Caller ID (per call)
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*50
*51
*70
*71
*72
Disable Call Waiting (for all subsequent calls)
Enable Call Waiting (for all subsequent calls)
Disable Call Waiting. (Per Call)
Enable Call Waiting (Per Call)
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
*90
Cancel Unconditional Call Forward
To cancel “Unconditional Call Forward”, dial “*73” and get
the dial tone, then hang up.
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
*92
Cancel Busy Call Forward
To cancel “Busy Call Forward”, dial “*91” and get the dial
tone, then hang up
Delayed Call Forward
To use this feature, dial “*92” 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
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.
Flash/Hook
5.3.2 PSTN Pass Through
When HandyTone-488 is out of power, the RJ 11 line jack on the HandyTone-488 side will
function as a pass through jack. Users will be able to use the same analog phone for PSTN
calls.
5.4 LED Light Pattern Indication
Following tables show the LED light pattern indication.
RED LED always indicates abnormal status
DHCP Failed or WAN No Cable
HandyTone-488 fails to register
Button flashes every 2 seconds (if DHCP is configured)
Button flashes every 2 seconds (if SIP server is
configured)
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GREEN LED always indicates normal working status
Message Waiting Indication
RINGING
RINGING INTERVAL
Button flashes every 2 seconds
Button flashes at 1/10 second
Button flashes every second
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6 Configuration 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 to set up HandyTone-488’s IP, Subnet Mask, Gateway
respectively.
6.2 Configuring HandyTone-488 with Web Browser
HandyTone-488 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 menu can be accessed via LAN or WAN port:
• From the LAN port use the default LAN gateway IP address:
• The WAN port HTML configuration option is disabled by default from factory. To
access the HTML configuration menu from the WAN port, first enable the “WAN side
HTTP access” option by accessing the configuration via LAN port. With the WAN side
HTTP access enabled, then get the WAN IP address of the HandyTone-488 through
section 5.1 with menu option 02. The HandyTone-488’s Web Configuration page can
be accessed by the following URI via WAN port:
where the HandyTone-IP-Address is the WAN IP address of the HandyTone-488.
6.2.2 End User Configuration
Once this HTTP request is entered and sent from a Web browser, the HandyTone-488 will
respond with the following login screen:
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Grandstream Device Configuration
Password
Login
All Rights Reserved Grandstream Networks, Inc. 2004
The password is case sensitive and the factory default password for End User is ‘123’.
After a correct password is entered in the login screen, the embedded Web server inside the
HandyTone-488 will respond with the Configuration page which is explained in details below.
Grandstream Device Configuration
STATUS BASIC SETTINGS ADVANCED SETTINGS
End User
Password:
(purposely not displayed for security protection)
IP Address:
dynamically assigned via DHCP (default) or
PPPoE
(will attempt PPPoE if DHCP fails and following is
non-blank)
PPPoE account ID:
PPPoE password:
0
0
0
0
Preferred DNS server:
.
.
.
statically configured as:
192
255
192
129
204
168
1
12
0
IP Address:
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
255
168
105
127
255
1
Subnet Mask:
Default Router:
DNS Server 1:
DNS Server 2:
1
5
1
198
4
current setting is " GMT-6:00 (US Central Time, Chicago)"
Time Zone:
Daylight Savings
Time:
No
Yes (if set to Yes, display time will be 1 hour ahead of normal
time)
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NAT/DHCP Server Information & Configuration:
Cloned WAN MAC
Addr:
(in hex format)
(default is 255.255.255.0)
(base IP for the LAN port, default is 192.168.2.1)
255.255.255.0
LAN Subnet Mask:
LAN DHCP Base
IP:
192.168.2.1
DHCP IP Lease
Time:
120
(in units of hours, default is 120 hours or 5 days)
DMZ IP:
0
0
0
0
0
0
0
0
0
0
0
0
0
0
" UDPOnly"
" UDPOnly"
" UDPOnly"
" UDPOnly"
" UDPOnly"
" UDPOnly"
" UDPOnly"
" UDPOnly"
WAN port
WAN port
WAN port
WAN port
WAN port
WAN port
WAN port
WAN port
LAN IP
LAN IP
LAN IP
LAN IP
LAN IP
LAN IP
LAN IP
LAN IP
LAN port
LAN port
LAN port
LAN port
LAN port
LAN port
LAN port
LAN port
Protocol
Protocol
Protocol
Protocol
Protocol
Protocol
Protocol
Protocol
Port Forwarding:
0
0
Update
All Rights Reserved Grandstream Networks, Inc. 2004
End User
Password
This contains the password to access the Web Configuration Menu.
This field is case sensitive.
IP Address
There are 2 modes under which the HandyTone ATA can operate:
- If DHCP mode is enabled, then all the field values for the Static IP
mode are not used (even though they are still saved in the Flash
memory.) The HandyTone ATA will acquire its IP address from the
first DHCP server it discovers from the LAN it is connected.
To use the PPPoE feature the PPPoE account settings need to be
set. The HandyTone will attempt to establish a PPPoE session if any
of the PPPoE fields is set.
- If Static IP mode is enabled, then the IP address, Subnet Mask,
Default Router IP address, DNS Server 1 (primary), DNS Server 2
(secondary) fields will need to be configured. These fields are reset to
zero by default.
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Time Zone
This parameter controls how the displayed date/time will be adjusted
according to the specified time zone.
Daylight Savings
Time
This parameter controls whether the displayed time will be daylight
savings time or not. If set to Yes, then the displayed time will be 1
hour ahead of normal time.
Cloned WAN MAC
Address:
Allow the user to set a specific MAC address. Set in Hex format
LAN Subnet Mask
Sets the LAN subnet mask. Default value is 255.255.255.0
LAN DHCP Base IP: Base IP for the LAN port which functions as a Gateway for the
subnet. Default value is 192.168.2.1
DHCP IP Lease
Time:
Value is set in units of hours. Default value is 120hr (5 Days.) The
time IP address are assigned to the LAN clients
DMZ IP:
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.
Port Forwarding:
Allow users to forward a matching (TCP/UDP) port to a specific LAN
IP address with a specific (TCP/UDP) port.
In addition to the Basic Settings configuration page, end users also have access to the device
Status page. The following is a snap shot of the device Status page. Details will be explained
next.
Grandstream Device Configuration
STATUS BASIC SETTINGS ADVANCED SETTINGS
MAC Address: 00.0B.82.02.B2.95
WAN IP Address: 192.168.1.12
Product Model: HT488
Software Version: Program-- 1.0.0.3 Bootloader-- 1.0.2.22 HTML-- 1.0.0.2 VOC-- 1.0.0.7
System Up Time: 0 day(s) 3 hour(s) 23 minute(s)
Registered: Yes
PPPoE Link Up: disabled
detected NAT type is full cone
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MAC Address
The device ID, in HEX format. This is a very important ID for ISP
troubleshooting.
WAN IP Address
Product Model
This field shows WAN port IP address.
This field contains the product model info.
Software Version
Program: This is the main software release, its number is always
used for firmware upgrade.
Bootloader: This is normally not changed.
HTML: This is the user interface, normally not changed.
VOC: This is the codec program, normally not changed.
System Up Time
This field indicates how long the device has been up since the last
reboot.
Registered
This field indicates whether the device is registered to the SIP server.
This field shows whether the PPPoE connection is enabled or not.
PPPoE Link Up
Detected NAT Type
This field shows what kind NAT the HandyTone is connected to via its
WAN port. It is based on STUN protocol.
6.2.3 Advanced User Configuration
To login to the Advanced User Configuration page, please follow the instructions in section
6.2.1 to get to the following login page. The password is case sensitive and the factory
default password for Advanced User is “admin”.
Grandstream Device Configuration
Password
Login
All Rights Reserved Grandstream Networks, Inc. 2004
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Advanced User configuration includes not only the end user configuration, but also advanced
configuration such as SIP configuration, Codec selection, NAT Traversal Setting and other
miscellaneous configuration. Following is a snap shot of the advanced configuration page.
Grandstream Device Configuration
STATUS BASIC SETTINGS ADVANCED SETTINGS
Admin Password:
(purposely not displayed for security protection)
FXS Port:
sip.mycompany.com
SIP Server:
(e.g., sip.mycompany.com, or IP address)
Outbound Proxy:
(e.g., proxy.myprovider.com, or IP address, if any)
3320
SIP User ID:
(the user part of an SIP address)
3320
Authenticate ID:
(can be identical to or different from SIP User ID)
Authenticate
(purposely not displayed for security protection)
Password:
John Doe
Name:
(optional, e.g., John Doe)
Register
Expiration:
60
(in minutes. default 1 hour, max 45 days)
5060
local SIP port:
(default 5060)
5004
local RTP port:
(1024-65535, default 5004)
Enable Call
No
Yes (if Yes, Call Forwarding & Call-Waiting-Disable are
Features:
supported locally)
Send DTMF:
in-audio
via RTP (RFC2833)
via SIP INFO
DTMF Payload
Type:
101
FXO Port:
sip.mycompany.com
SIP Server:
Outbound Proxy:
SIP User ID:
(e.g., sip.mycompany.com, or IP address)
(e.g., proxy.myprovider.com, or IP address, if any)
(the user part of an SIP address)
6602
6602
Authenticate ID:
(can be identical to or different from SIP User ID)
Authenticate
Password:
(purposely not displayed for security protection)
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John Doe
Name:
(optional, e.g., John Doe)
Register
Expiration:
60
(in minutes. default 1 hour, max 45 days)
5062
5008
local SIP port:
local RTP port:
(default 5062)
(1024-65535, default 5008)
Enable Call
Features:
No
Yes (if Yes, Call Forwarding & Call-Waiting-Disable are
supported locally)
Send DTMF:
in-audio
via RTP (RFC2833)
via SIP INFO
DTMF Payload
Type:
101
General
Settings:
Misc.
current setting is " PCMU"
current setting is " PCMA"
Preferred Vocoder:
(in listed order)
choice 1:
choice 2:
choice 3:
choice 4:
choice 5:
choice 6:
choice 7:
2
current setting is " G723"
current setting is " G729"
current setting is " G726-32"
current setting is " G728"
current setting is " iLBC"
Voice Frames per
TX:
(up to 10/20/32/64 for G711/G726/G723/other codecs respectively)
G723 rate:
iLBC frame size:
6.3kbps encoding rate
20ms 30ms
5.3kbps encoding rate
99
iLBC payload type:
Silence Suppression:
SIP Registration:
(between 96 and 127, default is 98)
No
Yes
No
Yes
Unregister On
Reboot:
Yes
No
48
Layer 3 QoS:
Layer 2 QoS:
Use DNS SRV:
(Diff-Serv or Precedence value)
0
0
802.1Q/VLAN Tag
No Yes
802.1p priority value
(0-7)
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User ID is phone
number:
No
Yes
Early Dial:
No
Yes (use "Yes" only if proxy supports 484 response)
(this prefix string is added to each dialed number)
Dial Plan Prefix:
No Key Entry
Timeout:
4
(in seconds, default is 4 seconds)
Use # as Dial Key:
Use random port:
No
No
No
Yes (if set to Yes, "#" will function as the "(Re-)Dial" key)
Yes
NAT Traversal:
stun.mycompany.com
Yes, STUN server is:
(URI or IP:port)
20
keep-alive interval:
Use NAT IP
(in seconds, default 20 seconds)
(if specified, this IP address is used in SIP/SDP message)
(if specified, the content will appear in Proxy-Require header)
Proxy-Require:
TFTP Upgrade
Server:
192
168
1
30
.
.
.
(for remote software upgrade and configuration)
HTTP Upgrade
Server:
(IP address or URL)
7
Auto Upgrade:
No
Yes, check for new firmware every
days (default 7 days)
SUBSCRIBE for
MWI:
No, do not send SUBSCRIBE for Message Waiting Indication
Yes, send periodical SUBSCRIBE for Message Waiting Indication
Offhook Auto-Dial:
(User ID/extension to dial automatically when offhook)
Send Flash Event:
FXS Impedance:
No
Yes (Flash will be sent as a DTMF event if set to Yes)
current setting is " 600 Ohm (North America)"
current setting is " Bellcore"
Caller ID Scheme:
Disable Call-
Waiting:
No
Yes
Polarity Reversal:
NTP Server:
No
Yes (reverse polarity upon call establishment and termination)
(URI or IP address)
time.nist.gov
Send Anonymous:
No
No
Yes (caller ID will be blocked if set to Yes)
Yes (configuration update via keypad is disabled if set to Yes)
Lock keypad update:
Special Feature:
current setting is "Standard"
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WAN side http
No
Yes (WAN side access to http server will be rejected if set to No)
access:
PSTN access code:
(key pattern to use the PSTN line, default is "*00")
Update
All Rights Reserved Grandstream Networks, Inc. 2004
Admin Password This contains the password to access the Advanced Web Configuration
page. This field is case sensitive.
SIP Server
SIP Server’s URI or IP address
Outbound Proxy SIP Outbound Proxy Server’s URI or IP address
SIP User ID
SIP service subscriber’s User ID
Authenticate ID
SIP service subscriber’s Authenticate ID. Can be identical to or different
from SIP User ID
Authenticate
Password
SIP service subscriber’s account password
Name
SIP service subscriber’s name which will be used for Caller ID display
Register
Expiration
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).
Local SIP port
Local RTP port
This parameter defines the local SIP port the HandyTone ATA will listen
and transmit. The default value for FXS port is 5060. The default value
for FXO port is 5062.
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. The
default value for FXO port is 5008.
Enable Call
Features
Default is No. If set to Yes, Call Forwarding & Do-Not-Disturb are
supported locally
Send DTMF
This parameter controls how DTMF events are transmitted. There are 3
ways: 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.
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DTMF Payload
Type
This parameter sets the payload type for DTMF using RFC2833
Preferred
Vocoder
The HandyTone ATA supports up to 7 different Vocoder types including
G.711 A-/U-law，G.723.1, G.726, G.728, G.729A/B, iLBC. Depending
on the product model, some of these Vocoders may not be provided in
standard release.
Users can configure Vocoders in a preference list that will be included
with the same preference order in SDP message. The first Vocoder in
this list can be entered by choosing the appropriate option in “Choice 1”.
Similarly, the last Vocoder in this list can be entered by choosing the
appropriate option in “Choice 7”.
Voice Frames per This field contains the number of voice frames to be transmitted in a
TX
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
above vocoder Preference List or the actual used payload type
negotiated between the 2 conversation parties at run time.
e.g., if the first vocoder is configured as G723 and the “Voice Frames
per TX” is set to be 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. Similarly, if this field is set to be 2 and if the first vocoder
chosen is G729 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.
G723 Rate:
This defines the encoding rate for G723 vocoder. By default, 6.3kbps
rate is chosen.
iLBC frame size: This sets the iLBC size in 20ms or 30ms
iLBC payload
type:
This defines payload time for iLBC. Default value is 98. The valid range
is between 96 and 127.
Silence
Suppression
This controls the silence suppression/VAD feature of G723 and G729. 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.
SIP Registration
This parameter controls whether the HandyTone ATA needs to send
REGISTER messages to the proxy server. The default setting is “Yes”.
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Unregister on
Reboot
Default is No. If set to yes, the SIP user will be unregistered on reboot.
Layer 3 QoS
Layer 2 QoS
Use DNS SRV:
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.
This contains the value used for layer 2 VLAN tag. Default setting is
blank.
Default is No. If set to Yes the client will use DNS SRV for server lookup
User ID is Phone
Number
If the HandyTone ATA has an assigned PSTN telephone number, this
field should be set to “Yes”. Otherwise, set it to “No”. If “Yes” is set, a
“user=phone” parameter will be attached to the “From” header in SIP
request
Early Dial
Default is No. Use only if proxy supports 484 response
Sets the prefix added to each dialed number
Default is 4 seconds.
Dial Plan Prefix
No Key Entry
Timeout
Use # as
Send Key
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. In this case, this key
is essentially equivalent to the “(Re)Dial” key. If set to “No”, this “#” key
will then be included as part of the dial string to be sent out.
Use Random Port This parameter, when set to Yes, will force random generation of both
the local SIP and RTP ports. This is usually necessary when multiple
HandyTone ATAs are behind the same NAT.
NAT Traversal
This parameter defines whether the HandyTone ATA NAT traversal
mechanism will be activated or not. If activated (by choosing “Yes”) and
a STUN server is also specified, then the HandyTone ATA will behave
according to the STUN client specification. Under this mode, the
embedded STUN client inside the HandyTone ATA will attempt to detect
if and what type of firewall/NAT it is sitting behind through
communication with the specified STUN server. If the detected NAT is a
Full Cone, Restricted Cone, or a Port-Restricted Cone, the HandyTone
ATA will attempt to use its mapped public IP address and port in all its
SIP and SDP messages. If the NAT Traversal field is set to “Yes” with
no specified STUN server, the HandyTone ATA will periodically (every
20 seconds or so) send a blank UDP packet (with no payload data) to
the SIP server to keep the “hole” on the NAT open.
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Keep-alive
interval
This parameter specifies how often the HandyTone ATA sends a blank
UDP packet to the SIP server in order to keep the “hole” on the NAT
open.
Use NAT IP:
NAT IP address used in SIP/SDP message. Default is blank.
Proxy-Require
SIP Extension to notify SIP server that the unit is behind the
NAT/Firewall.
TFTP Upgrade
Server
This is the IP address of the configured TFTP server. If it is non-zero or
not blank, the HandyTone ATA will attempt to retrieve new configuration
file or new code image from the specified TFTP server at boot time. It
will make up to 3 attempts before timeout and then it will start the boot
process using the existing code image in the Flash memory. If a TFTP
server is configured and a new code image is retrieved, the new
downloaded image will be verified and then saved into the Flash
memory.
HTTP Upgrade
Server
The URL for the HTTP server used for firmware upgrade and
configuration via HTTP. For example,
Here “:6688” is the specific TCP port that the HTTP server is listening
at, it can be omitted if using default port 80.
Note: Auto Upgrade has to be set to Yes to enable HTTP upgrade.
Auto Upgrade
Choose Yes to enable HTTP upgrade and provisioning.
In “Check for new firmware every” field, enter the number of days to
check the HTTP server for firmware upgrade or configuration.
Subscribe for
MWI:
Default is NO. When set to “Yes” a SUBSCRIBE for Message Waiting
Indication will be sent periodically.
Offhook
Auto-Dial
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.
Send Flash Event This parameter allows users to control whether to send an SIP NOTIFY
message indicating the Flash event, or just to switch to the voice
channel when users press the Flash key.
FXS Impedance
Selects the impedance of the analog telephone connected to the Phone
port.
Caller ID Scheme • Bellcore (North America)
• ETSI-FSK (France, Germany, Norway, Taiwan, UK-CCA)
• ETSI-DTMF (Finland, Sweden)
• DTMF (Denmark)
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Disable Call
Waiting
Default is No.
Polarity Reversal Default is No. If set to Yes, polarity will be reversed upon call
establishment and termination.
NTP server
This parameter defines the URI or IP address of the NTP server which
is used by the HandyTone ATA to display the current date/time.
Send Anonymous If this parameter is set to “Yes”, the “From” header in outgoing INVITE
message will be set to anonymous, essentially blocking the Caller ID
from displaying.
Lock keypad
update
If this parameter is set to “Yes”, the configuration update via keypad is
disabled.
Special Feature
• Standard
• Nortel MCS
• Broadsoft
• Howdy
WAN side http
access
If this parameter is set to “No”, the HTML configuration update via WAN
port is disabled.
PSTN access
code
This field allows users to customize their own code to access the PSTN
line. Default is “*00”.
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6.2.4 Saving the Configuration Changes
Once a change is made, users should press the “Update” button in the Configuration page.
The HandyTone-488 will then display the following screen to confirm that the changes have
been saved.
Grandstream Device Configuration
STATUS BASIC SETTINGS ADVANCED SETTINGS
Your configuration changes have been saved.
They will take effect on next reboot.
All Rights Reserved Grandstream Networks, Inc. 2004
Users are recommended to power cycle the HandyTone-488 after seeing the above
message.
6.2.5 Rebooting the HandyTone-488 from Remote
The administrator of the HandyTone-488 can remotely reboot the HandyTone ATA by
pressing the “Reboot” button at the bottom of the configuration page. Once done, the
following screen will be displayed to indicate that rebooting is underway.
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
At this point, the user can relogin to the HandyTone-488 after waiting for about 30 seconds.
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6.3 Configuration through a Central Server
Grandstream HandyTone ATAs can be automatically configured from a central provisioning
system.
When HandyTone ATA boot up, it will send TFTP or HTTP request to download configuration
files, there are two configuration files, one is “cfg.txt” and the other is “cfg000b82xxxxxx”,
where “000b82xxxxxx” is the MAC address of the HandyTone ATA. For more information
regarding configuration file format, please refer to the related Grandstream documentation.
The configuration file can be downloaded via TFTP or HTTP from the central server. A
service provider or an enterprise with large deployment of HandyTone ATAs can easily
manage the configuration and service provisioning of individual devices remotely and
automatically from a central server. 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 HandyTone ATA for firmware upgrade, remote
reboot, etc.
Grandstream provides a licensed provisioning system called GAPS that can be used to
support automated configuration of HandyTone ATA. To enable this feature on the
HandyTone ATA, a user just needs to enter the IP address of the GAPS server in the TFTP
server field of the configuration screen, or enter the HTTP provisioning Server URL in the
HTTP Upgrade Server field. Then reboot the HandyTone ATA.
For details on how GAPS works, please refer to the documentation of GAPS product.
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7 Software Upgrade
7.1 Upgrade through HTTP
To upgrade software, HandyTone-488 can be configured with an HTTP server where the new
code image file is located. For example, following URL in the HTTP Upgrade Server:
Where firmware.mycompany.com is the FQDN of the HTTP server, “:6688” is the TCP port
the HTTP server listening to, “/Grandstream/1.0.0.4” is the RELATIVE directory to the root dir
in HTTP server. Thus, you can put different firmware into different directory as well.
Note: To enable HTTP firmware upgrade, “Auto Upgrade” field has to be set to Yes. In
addition, the ATA will check the HTTP server in the number of days that is defined in “Check
for new firmware every” field.
7.2 Upgrade through TFTP
To upgrade software, HandyTone-488 can be configured with a TFTP server where the new
code image is located. The TFTP upgrade can work in either static IP or DHCP mode using
private or public IP address. It is recommended to set the TFTP server address in either a
public IP address or on the same LAN with the HandyTone-488.
There are two ways to set up the TFTP server to upgrade the firmware, namely through voice
menu prompt or via the HandyTone-488’s Web configuration interface. To configure the
TFTP server via voice prompt, follow section 5.1 with option 06, once set up the TFTP IP
address, power cycle the ATA, the firmware will be fetched once the ATA boots up.
To configure the TFTP server via the Web configuration interface, open up your browser to
point at the IP address of the HandyTone-488. Input the admin password to enter the
configuration screen. From there, enter the TFTP server address in the designated field
towards the bottom of the configuration screen.
Once the TFTP server is configured, power cycle the HandyTone-488.
TFTP checking is only performed during the initial power up. If the configured TFTP server is
found and a new code image is available, the HandyTone-488 will attempt to retrieve the new
image files by downloading them into the HandyTone-488’s SRAM. During this stage, the
HandyTone-488’s LED 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 fails
for any reason (e.g., TFTP server is not responding, there are no code image files available
for upgrade, or checksum test fails, etc), the HandyTone-488will stop the TFTP process and
simply boot using the existing code image in the flash.
TFTP process may take as long as 1 to 2 minutes over the Internet, or just 20+ seconds if it is
performed on a LAN. Users are recommended to conduct TFTP upgrade in a controlled LAN
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environment if possible. For those who do not have a local TFTP server, Grandstream
provides a NAT-friendly TFTP server on the public Internet for firmware upgrade. Please
check the Service or Support section of Grandstream’s Web site to obtain this TFTP server’s
IP address.
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8 Restore Factory Default Setting
Warning: Restore the Factory Default Setting will delete all configuration information of
the device.
Please disconnect network cable and power cycle the unit before trying to reset the unit to
factory default. The steps are as follows:
A. Find the MAC address of the device. It is a 12 digits number located on the bottom of the
unit.
B. Encode the MAC address. Please use the following mapping:
0-9: 0-9
A: 22
B: 222
C: 2222
D: 33
E: 333
F: 3333
For example, 000B82004D2A will be encoded to 0002228200433222.
C. To perform factory reset:
1. Press “***” or the LED button for voice prompt.
2. Enter “99” and get the voice prompt “Reset”.
3. Enter the encoded MAC address of the device. A "click" sound will be heard.
4. Wait for 15 seconds.
The device is now restored to the factory default setting.
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