Grandstream Networks HandyTone-486 User manual

User Manual
HandyTone-486
Analog Telephone Adaptor
User Manual 1.0 Version
Grandstream Networks, Inc.
www.grandstream.com

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Table of Contents
1WELCOME......................................................................................................... 3
2INSTALLATION ................................................................................................. 4
3WHAT IS INCLUDED IN THE PACKAGE ......................................................... 5
3.1 SAFETY COMPLIANCES .................................................................................... 5
4PRODUCT OVERVIEW...................................................................................... 6
4.1 KEY FEATURES ............................................................................................... 6
4.2 HARDWARE SPECIFICATION ............................................................................. 7
5BASIC OPERATIONS........................................................................................ 8
5.1 GET FAMILIAR WITH KEY PAD AND VOICE PROMPT............................................. 8
5.2 MAKE PHONE CALLS ....................................................................................... 9
5.2.1 Calling phone or extension numbers ...................................................... 9
5.2.2 Direct IP calls.......................................................................................... 9
5.3 CALL FEATURES............................................................................................ 11
5.3.1 Call Features Table .............................................................................. 11
5.3.2 PSTN Pass Through............................................................................. 12
5.4 LED LIGHT PATTERN INDICATION ................................................................... 12
6CONFIGURATION GUIDE............................................................................... 13
6.1 CONFIGURING HANDYTONE 486 WAN IP THROUGH VOICE PROMPT ................ 13
6.1.1 DHCP Mode ......................................................................................... 13
6.1.2 STATIC IP Mode................................................................................... 13
6.2 CONFIGURING HANDYTONE 486 WITH WEB BROWSER..................................... 13
6.2.1 Access the Web Configuration Menu.................................................... 13
6.2.2 Configuration Menu .............................................................................. 14
6.2.3 Saving the Configuration Changes....................................................... 24
6.2.4 Rebooting the HandyTone 486 from remotely...................................... 24
6.3 CONFIGURATION THROUGH A CENTRAL SERVER .............................................. 24
7SOFTWARE UPGRADE.................................................................................. 26
7.1 UPGRADE WITH TFTP ................................................................................... 26

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1 Welcome
Congratulations on becoming an owner of HandyTone-486. You made an excellent choice
and we hope you will enjoy all its capabilities.
Grandstream's award-wining HandyTone-486 is an all-in-one VoIP integrated access device
that features superb audio quality, rich functionalities, high level of integration, compactness
and ultra-affordability. The HandyTone-486 is fully compatible with SIP industry standard and
can interoperate with many other SIP compliant devices and software on the market.
Grandstream HandyTone-486 has been awarded the Best of Show product in 2004 Internet
Telephony Conference and Expo.

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2 Installation
HandyTone-486 Analog Telephone Adaptor is an all-in-one VoIP integrated device designed
to be a total solution for networks providing VoIP services.
The HandyTone-486 VoIP functionalities are available via a regular analog telephone.
The following photo illustrates the appearance of a HandyTone-486.
Top View Side Views
Interconnection Diagram of the HandyTone-486:
Fax
Cordless Phone
Analog Phone
Internet ADSL/Cable
Modem Ethernet
PSTN
PC PCPC
LAN
WAN
LINE PHONE
BUTTON
RED LED
GREEN LED
RJ45
10M Ethernet
LAN - WAN
RJ11
Telephone
(Phone)
+5V/1200mA
RJ11
Phone Line
(Line)

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3 What is Included in the Package
The HandyTone-486 package contains:
1) One HandyTone-486
2) One universal power adaptor
3) One Ethernet cable
3.1 Safety Compliances
The HandyTone-486 is compliant with various safety standards including FCC/CE and C-tick.
Its power adaptor is compliant with UL standard. The HandyTone-486 should only operate
with the universal power adaptor provided in the package.
Warning: Please do not attempt to use a different power adaptor. Using other power
adaptor may damage the HandyTone-486 and will void the manufacturer warranty.
Caution: Changes or modifications to this product not expressly approved by
Grandstream, or operation of this product in any way other than as detailed by this
User Manual, could void your manufacturer warranty.
Information in this document is subject to change without notice. No part of this document may be
reproduced or transmitted in any form or by any means, electronic or mechanical, for any purpose without the
express written permission of Grandstream Networks, Inc..

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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
•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 (alaw 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
•Support firmware upgrade via TFTP
•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-486
LAN interface 1xRJ45 10Base-T
WAN interface 1xRJ45 10Base-T
FXS telephone port 1xFXS
PSTN port 1x life line port
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 Key Pad and Voice Prompt
HandyTone-486 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 Main Option” Enter ‘*’ to menu_01
Enter 00-06, 99 menu option
01 “DHCP Mode”,
“Static IP Mode”
Enter ‘9’ to toggle the selection
02 “IP Address “ + IP address It will prompt you with the current WAN IP
address.
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 prompt a dialtone,
then enter 12 digit IP address
This menu can be also entered by
pressing the button again
86 “Voice Messages Pending”
“No Voice Messages”
Enter 9 to dial pre-configured phone
number to retrieve VM
99 “RESET” Enter ‘9’ to confirm the RESET
Enter MAC address to restore factory
default setting
“Invalid Entry” Automatically return 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 state, it jumps to “Direct IP
Calling” option and dial tone plays in this state
•‘*’ functions similar to ‘↓’ key of BT-100 phone to select the next menu option
•‘#’ returns back to main MENU
•‘9’ is similar to ENTER key in many cases to confirm an option
•All entered digit sequences have known lengths - 2 digits for menu option and 12 digits
for IP address. Once all digits are accumulated, it automatically processes them.
•Key entry cannot 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
There are currently two methods to make a extension number call:
a) Dial the numbers number 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).
Other functions available during the call are call-waiting/flash, call-transfer, and call-
forwarding.
5.2.2 Direct IP calls
Direct IP calling allows 2 phones, that is, a HandyTone with a 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 other VoIP Device(i.e., another HandyTone ATA or other
HandyTone ATA) have public IP addresses, or
•both HandyTone ATA and other VoIP Device(i.e., another HandyTone ATA or other
HandyTone ATA) are on the same LAN using private or public IP addresses, or
•both HandyTone ATA and other VoIP Device(i.e., another HandyTone ATA or other
HandyTone ATA) can be connected through a router using public or private IP
addresses.
To make a direct IP calling, 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. If there is a user part, again follow Section 4.1 with voice prompt 47, and then the
encoded user part, followed by *3 (encoding for “@”) and then followed by the 12-digit target
IP address. Destination ports can also be specified using *4 (encoding for “:”) followed by the
encoded port number.

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The follow is a table of the encoding scheme for the most commonly used characters:
F
00 0
01 1
02 2
03 3
04 4
05 5
06 6
07 7
08 8
09 9
*0 . (dot character)
*1 _ (underscore character)
*2 - (hyphen character)
*3 @
*4 : (column character)
21 A
22 B
23 C
31 D
32 E
33 F
41 G
42 H
43 I
51 J
52 K
53 L
61 M
62 N
63 O
71 P
72 Q
73 R
74 S
81 T
82 U
83 V
91 W
92 X
93 Y
94 Z

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The rule of thumb to remember these encoding is: “a” is the first letter on button “1” so its
encoding is “11”. “b” is the 2nd letter on button “1” and its encoding is “12”. “c” is the 3rd letter
on button “1” and its encoding is “13”. Likewise, “d” is the first letter on button “2” and its
encoding is “21”. This pattern and rule applies to all other alphabetic encoding.
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 for more than 5
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 5 seconds.
If the target address is j[email protected]:5062, then the dialing convention would be:
Voice Prompt with option 47, then 51634262*3192168001100*45062
followed by pressing the “#” key if it is configured as a send key or wait for 5 seconds.
5.3 Call Features
5.3.1 Call Features Table
Following table shows the call features of HandyTone-486.
Key Call Features
*70 Do not disturb, it will give caller busy tone
when called.
*72 Unconditional Call Forward
*73 Cancel Unconditional Call Forward
*90 Busy Call Forward
*91 Cancel Busy Call Forward
*92 Delayed Call Forward
*93 Cancel Delayed Call Forward
Flash/Hook When in conversation, this action will
switch to the new incoming call if user
heard the call waiting sound.
When in conversation and no incoming
call heard, this action will switch to a new
channel for a new call.

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5.3.2 PSTN Pass Through
When HandyTone-486 is out of power, the RJ 11 line jack on the HandyTone-486 side will
function as a pass through jack, user will be able to use the same analog phone for PSTN
calls.
5.4 LED Light Pattern Indication
Following are the LED light pattern indication.
RED LED always indicates not normal status
DHCP Failed or WAN No Cable flash every 2 seconds (if DHCP is configured)
HandyTone-486 fails to register flash every 2 seconds (if SIP server is configured)
GREEN LED always for normal working status
Message Waiting Indication Button flashes every 2 seconds
RINGING Button flashes at 1/10 second
RINGING INTERVAL Button flashes every second

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6 Configuration Guide
6.1 Configuring HandyTone-486 WAN IP through Voice Prompt
6.1.1 DHCP Mode
Follow section 5.1 with voice menu option 01 to enable HandyTone-486 to use DHCP.
6.1.2 STATIC IP Mode
Follow section 5.1 with voice menu option 01 to enable HandyTone-486 to use STATIC IP
mode, then use option 02, 03, 04 to set up HandyTone’s IP, Subnet Mask, Gateway
respectively.
6.2 Configuring HandyTone-486 with Web Browser
HandyTone-486 series ATA has an embedded Web server that will respond to HTTP
GET/POST requests. It also has embedded HTML pages that allow a user to configure the
HandyTone through a Web browser such as Microsoft’s IE and AOL’s Netscape.
6.2.1 Access the Web Configuration Menu
The HandyTone-486 HTML configuration menu can be accessed via LAN or WAN port:
•From the LAN port use the default LAN gateway IP address:
http:/192.168.2.1
•The WAN port HTML configuration menu option is disabled 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 enable, then get the WAN IP address of the
HandyTone through section 4.1 with menu option 02. Then the HandyTone’s Web
Configuration Menu can be accessed by the following URI via WAN port:
http://HandyTone-IP-Address,
where the HandyTone-IP-Address is the WAN IP address of the HandyTone-486.
Once this request is entered and sent from a Web browser, the HandyTone-486 will respond
with the following login screen:

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Welcome to Grandstream HandyTone-486
Password
Login
The password is case sensitive and the factory default password is lower case ‘admin’.
6.2.2 Configuration Menu
After the correct password is entered in the login screen, the embedded Web server inside
the HandyTone-486 will respond with the Configuration Menu screen which is explained in
details below.
The definitions for all the configuration parameters in the Configuration Menu are following a
sample of the actual configuration page:

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Grandstream HandyTone-486 Configuration
MAC Address: 00.0B.82.00.XX.XX
Product Model: HT486
Software Version: Program--1.0.X.X Bootloader—1.0.0.XX HTML--1.0.0.XX VOC--
1.0.0.5
Admin Password: ***** (password to configure this IP phone)
WAN 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:
Use this DNS server (if specified): 0.0.0.0
statically configured as:
IP Address: 192 .168 .0.160
Subnet Mask: 0.0.0.0
Default Router: 0.0.0.0
DNS Server 1: 0.0.0.0
DNS Server 2: 0.0.0.0
SIP Server: (e.g., sip.mycompany.com, or IP address)
Outbound Prox
y
: (e.g., proxy.myprovider.com, or IP address, if any)
SIP User ID: (the user part of an SIP address)
Authenticate ID: (can be identical to or different from SIP User ID)
Authenticate
Password:
Name: (optional, e.g., John Doe)

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Admin
Password This contains the password to access the Web Configuration Menu.
This field is case sensitive.
WAN 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.
To use the PPPoE feature set the PPPoE account settings. The
Handytone will attempt to establish a PPPoE session if any of the
PPPoE fieds is set.
- If Static IP mode is selected, then the IP address, Subnet Mask,
Default Router IP address, DNS Server 1 (primary), DNS Server 2
(secondary) fields will need to be configured. These fields are reset to
zero by default.
SIP Server 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-voip-provider.com, or sip:my-company-sip-
server.com, or 192.168.1.200:5066
Outbound Proxy This field contains the URI string or the IP address (and port, if different
from 5060) of the outbound proxy. If there is no outbound proxy, this
field SHOULD be left blank. If not blank, all outgoing requests will be
sent to this outbound proxy.
SIP User ID This field contains the user part of the SIP address for this HandyTone
ATA. e.g., if the SIP address is: sip:my_user_id@my_provider.com,
then the SIP User ID is: my_user_id. Please do NOT include the
preceding “sip:” scheme or the host portion of the SIP address in this
field.
SIP User ID is
Phone Number If the HandyTone ATA has an assigned PSTN telephone number, then
this field will 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.
SIP Login ID This field contains the login ID used for SIP authentication. Typically,
this is the account number on an SIP server for this HandyTone ATA. It
can be the same as or different from the above SIP User ID, depending
on whether a separate account ID is used for authentication.
SIP Password This field contains the password used for SIP authentication. It is used
together with the above SIP Login ID

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Advanced Options:
A
dvanced Options:
Preferred Vocoder:
(in listed order) choice 1:
choice 2:
choice 3:
choice 4:
choice 5:
choice 6:
choice 7:
choice 8:
G723 rate: 6.3kbps encoding rate 5.3kbps encoding rate
iLBC frame size: 20ms 30ms
iLBC payload type: 98 (between 96 and 127, default is 98)
Silence Suppression: No Yes
Voice Frames per TX: 2(up to 10/20/32/64 for G711/G726/G723/other codecs
respectively)
Layer 3 QoS: 48 (Diff-Serv or Precedence value)
Layer 2 QoS: 802.1Q/VLAN Tag 0802.1p priority value 0(0-7)
Use DNS SRV: No Yes
User ID is phone
number: No Yes
SIP Registration: Yes No
Unregister On Reboot: Yes No
R
egister Expiration: 60 (in minutes. default 1 hour, max 45 days)
Early Dial: No Yes (use "Yes" only if proxy supports 484 response)
Dial Plan Prefix: (this prefix string is added to each dialed number)

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No Key Entry
Timeout: 4(in seconds, default is 4 seconds)
Use # as Dial Key: No Yes (if set to Yes, "#" will function as the "(Re-)Dial" key)
local SIP port: 5060 (default 5060)
Local RTP port: 5004 (1024-65535, default 5004)
Use random port: No Yes
N
AT Traversal: No
Yes, STUN server is: (URI or IP:port)
keep-alive interval: 20 (in seconds, default 20 seconds)
Use NAT IP (if specified, this IP address is used in SIP/SDP
message)
TFTP Server: 67 .153 .142 .66 (for remote software upgrade and configuration)
SUBSCRIBE for
M
WI: No, do not send SUBSCRIBE for Message Waiting Indication
Yes, send periodical SUBSCRIBE for Message Waiting Indication
Offhook Auto-
D
ial: (User ID/extension to dial automatically when offhook)
E
nable Call
Features No Yes (if Yes, Call Forwarding & Do-Not-Disturb are supported
locally)
Disable Call-
Waiting: No Yes
Send DTMF: in-audio via RTP (RFC2833) via SIP INFO
D
TMF Payload
Type: 101
Send Flash Event: No Yes (Flash will be sent as a DTMF event if set to Yes)
F
XS Impedance:
N
TP Server: time.nist.gov (URI or IP address)
Time Zone:
Daylight Savings
Time: No Yes (if set to Yes, display time will be 1 hour ahead of normal
time)
Send Anonymous: No Yes (caller ID will be blocked if set to Yes)
Lock keypad
update: No Yes (configuration update via keypad is disabled if set to Yes)
WAN side http
access: No Yes (WAN side access to http server will be rejected if set to No)

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Preferred
Vocoder The BudgeTone HandyTone ATA supports up to 6 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.
A user 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 6”.
G723 Rate: This defines the encoding rate for G723 vocoder. By default, 6.3kbps
rate is chose.
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.
Voice Frames per
TX 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
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.
Layer 3 QoS 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 This contains the value used for layer 2 VLAN tag. Default setting is
blank.

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Use DNS SRV: Default is No. If set to Yes the
User ID is Phone
Number If the HandyTone ATA has an assigned PSTN telephone number,
then this field will 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
SIP Registration This parameter controls whether the HandyTone ATA needs to send
REGISTER messages to the proxy server. The default setting is “Yes”.
Unregister on
Reboot Default is No. If set to yes, the SIP user will be unregistered on reboot.
Register
Expiration This parameter allows the user to specify the time frequency (in
minutes) the HandyTone ATA will refresh its registration with the
specified registrar. The default interval is 60 minutes (or 1 hour). The
maximum interval is 65535 minutes (about 45 days).
Early Dial Default is No. Use only if proxy supports 484 response
Dial Plan Prefix Sets the prefix added to each dialed number
No Key Entry
timeout Default is 4 seconds.
Use # as
Send Key This parameter allows the user to configure the “#” key to be used as
the “Send”(or “Dial”) key. Once set to “Yes”, pressing this key will
immediately trigger the sending of dialed string collected so far. 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.
Local SIP port This parameter defines the local SIP port the HandyTone ATA will listen
and transmit on. The default value is 5060.
Local RTP port This parameter defines the local RTP-RTCP port pair the HandyTone
ATA will listen and transmit on. It is the base RTP port for channel 0.
When configured, channel 0 will use this port_value for RTP and the
port_value+1 for its RTCP; channel 1 will use port_value+2 for RTP and
port_value+3 for its RTCP. The default value is 5004.
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.