User Manual
HandyTone-496
Analog Telephone Adaptor
For SW Release Version 1.0.3.44
Grandstream Networks, Inc.
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HandyTone-496 User Manual
Grandstream Networks, Inc.
8
9
RESTORE FACTORY DEFAULT SETTING.......................................................... 37
GLOSSARY OF TERMS............................................................................................. 38
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HandyTone-496 User Manual
Grandstream Networks, Inc.
1 Welcome
Congratulations on becoming an owner of HandyTone-496. You made an excellent choice and we
hope you enjoy all of its capabilities.
Grandstream's HandyTone-496 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-496 is fully compatible with SIP industry standard and can interoperate with many other
SIP compliant devices and software on the market.
Grandstream HandyTone-496 is a new addition to the popular HandyTone product family. The new
HandyTone-496 features two FXS ports each with independent SIP accounts.
This document is subject to changes without notice. The latest electronic version of this user manual is
available for download from the following location:
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2 Installation
HandyTone-496 Analog Telephone Adaptor is an all-in-one VoIP integrated device designed to be a
total solution for networks providing VoIP services.
The HandyTone-496 VoIP functionalities are available via regular analog telephones.
The following photo illustrates the appearance of a HandyTone-496.
Top View
Side Views
RJ45
10M Ethernet
LAN - WAN
RJ11
FXS Port
(Phone)
+5V/1200mA
RJ11
FXS Port
(Phone)
BUTTON
RED LED
GREEN LED
Interconnection Diagram of the HandyTone-496:
Internet ADSL/Cable
Modem Ethernet
Analog Phone
Analog Phone
WAN
FXS 1
FXS 2
Cordless Phone
Cordless Phone
LAN
PC
PC
PC
Fax
Fax
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HandyTone-496 has two FXS ports. The PHONE1 port next to the LAN is FXS port 1 and the
PHONE2 on the side of the HandyTone-496 is FXS port 2. Each FXS port can have a separate SIP
account. This is a key feature of HandyTone-496. Both ports can make calls concurrently.
Following are the steps to install a HandyTone-496:
1. Connect a standard touch-tone analog telephone (or fax machine) to PHONE1 port.
2. Connect another standard touch-tone analog telephone (or fax machine) to PHONE2 port.
3. Insert the Ethernet cable into the WAN port of HandyTone-496 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-496.
5. Insert the power adapter into the HandyTone-496 and connect it to a wall outlet.
Please follow the instructions in section 6.2.1 to configure the HandyTone-496.
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3 What is Included in the Package
The HandyTone-496 package contains:
1) One HandyTone-496
2) One universal power adaptor
3) One Ethernet cable
3.1 Safety Compliances
The HandyTone-496 is compliant with various safety standards including FCC/CE and C-tick. Its
power adaptor is compliant with UL standard. The HandyTone-496 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-496 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
• Device bridge mode support
• Supports dual SIP accounts via dual FXS ports
• 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
(32K), as well as G.729A, and iLBC.
• Support Caller ID/Name display or block, Hold, Call Waiting/Flash, Call Transfer, Call
Forward, 3-way conferencing, Call Waiting Caller ID, in-band and out-of-band DTMF, Dial
Plans, etc.
• Support fax pass through (for PCMU and PCMA) and T.38 FoIP (Fax over IP)
• Support syslog
• Support volume amplification
• Support configurable Call Progress Tones
• 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
Grandstream Networks, Inc.
The table below lists the hardware specification of HandyTone-496.
Model
HandyTone-496
LAN interface
WAN interface
FXS telephone port
Button
1xRJ45 10Base-T
1xRJ45 10Base-T
2xFXS
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
FCC/CE/C-Tick
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5 Basic Operations
5.1 Get Familiar with Voice Prompt
HandyTone-496 stores a voice prompt menu (Interactive Voice Response or IVR) for quick browsing
and simple configuration. Currently, the voice prompt menu and the LED button are designed for FXS
port 1 only.
To enter this voice prompt menu, simply pick up the phone and press the button on the HandyTone-
496 or pick up the phone and dial “***”. 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
Enter “9” to toggle the selection
If user selects “Static IP Mode”, user need
configure the all IP address information
through menu 02 to 05. If user selects
“Dynamic IP Mode”, the device will retrieve
all IP address information from DHCP server
automatically when user reboots the device.
The current WAN IP address is announced
Enter 12 digit new IP address if in Static IP
Mode
01
02
“DHCP Mode”,
“Static IP Mode”
“IP Address “ + IP address
03
04
05
06
47
“Subnet “ + IP address
“Gateway “ + IP address
“DNS Server “ + IP address
“TFTP Server “ + IP address
“Direct IP Calling”
Same as menu 02
Same as menu 02
Same as menu 02
Same as menu 02
When entered, user will be prompted a dial
tone, dial a 12 digit IP address to make a direct
IP call.
(For details, see “4.2.2 Make a Direct IP
Call”.)
99
“RESET”
Enter “9” to reboot the device; or
Enter encoded MAC address to restore factory
default setting (For details, see section 8.)
Automatically returns to main menu
“Invalid Entry”
NOTES:
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• Once the button is pressed, it enters the voice prompt main menu. If the button is pressed again,
while it is already in the voice prompt menu, it jumps to “Direct IP Call” option and a dial
tone is prompted
• “*” shifts down to the next menu option
• “#” returns to the main menu
• “9” functions as the ENTER key in many cases to confirm an option
• All entered digit sequences have known lengths - 2 digits for menu option and 12 digits for IP
address. 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
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).
Examples:
To dial another extension on the same proxy, such as 1008, simply pick up the attached phone,
dial 1008 and then press the # or wait for 4 seconds.
To dial a PSTN number such as 6266667890, you 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 #
or wait for 4 seconds.
5.2.2 Direct IP Calls
Direct IP calling allows two parties, 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. 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).
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To make a direct IP call, first pick up the analog phone or turn on the speakerphone on the analog
phone, then access the voice menu prompt by dial “***” or press the button on the HandyTone-496,
and dials “47” to access the direct IP call menu. 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:
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 Call Hold
While in conversation, pressing the “flash” button on the attached phone will put the remote end on
hold. Pressing the “flash” button again will release the previously held party and the bi-directional
media will resume.
5.2.4 Call Waiting
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 toggles between two active calls.
5.2.5 Call Transfer
5.2.5.1 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: “Enable Call Feature” has to be set to “Yes” in web configuration page.
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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.
5.2.5.2 Attended Transfer
Assuming that call party A and B are in conversation. A wants to Attend Transfer B to C:
1. A presses FLASH (on the analog phone, or Hook Flash for old model phones) to get a dial tone
2. A then dial C’s number then # (or wait for 4 seconds).
3. If C answers the call, A and C are in conversation. Then A can hang up to complete transfer.
4. If C does not answer the call, A can press “flash” back to talk to B.
NOTE:
• When Attended Transfer failed, if A hangs up, the HandTone-496 will ring user A again to
remind A that B is still on the call. A can pick up the phone to restore conversation with B.
5.2.6 3-way Conferencing
HandyTone-496 supports 3-way conference in two styles: star code style or Bellcore style.
5.2.6.1 Star Code Style 3-way Conference
Assuming that call party A and B are in conversation. A wants to bring C in a conference:
1. A presses FLASH (on the analog phone, or Hook Flash for old model phones) to get a dial
tone.
2. A dials *23 then C’s number then # (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.
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5.2.6.2 Bellcore Style 3-way Conference
Bellcore style 3-way conference is also supported. To do this, user needs to enable “Use Bell-style 3-
way Conference” in FXS1 or FXS2 web configuration.
Assuming that call party A and B are in conversation. A wants to bring C in a conference:
1. A presses FLASH (on the analog phone, or Hook Flash for old model phones) to get a dial
tone.
2. A dials C’s number then # (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.
5.3 Call Features
Following table shows the call features of HandyTone-496.
Key
*23
*87
*30
*31
*67
*82
*50
*51
*70
*71
*72
Call Features
3-way conference
Blind Transfer
Block Caller ID (for all subsequent calls)
Send Caller ID (for all subsequent calls)
Block Caller ID (per call)
Send Caller ID (per call)
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.
Cancel Unconditional Call Forward
To cancel “Unconditional Call Forward”, dial “*73” and get
the dial tone, then hang up.
*73
*90
*91
*92
*93
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.
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.
Cancel Delayed Call Forward
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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-496 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, user is recommended to select all the Preferred Codecs to be PCMU/PCMA.
5.5 LED Light Pattern Indication
Following tables show the LED light pattern indication. The LED shows PHONE1 status only.
RED LED always indicates not abnormal status
DHCP Failed or WAN No Cable
HandyTone-496 fails to register
Firmware Upgrading
Button flashes every 2 seconds (if DHCP is configured)
Button flashes every 2 seconds (if SIP server is configured)
Button flashes every 2 seconds
Device Malfunctions
Red light steady on
GREEN LED mostly indicates normal working status
Message Waiting Indication
RINGING
RINGING INTERVAL
In Conversation
Button flashes every 2 seconds
Button flashes at 1/10 second
Button flashes every second
Green light steady on
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6 Configuration Guide
6.1 Configuring HandyTone-496 WAN IP through Voice Prompt
6.1.1 DHCP Mode
Follow section 5.1 with voice menu option 01 to enable HandyTone-496 to use DHCP.
6.1.2 STATIC IP Mode
Follow section 5.1 with voice menu option 01 to enable HandyTone-496 to use STATIC IP mode, then
use option 02, 03, 04, 05 to set up IP address, 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 firmware and
configuration file server.
6.2 Configuring HandyTone-496 with Web Browser
HandyTone-496 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-496 through a Web browser
such as Microsoft’s IE and AOL’s Netscape.
6.2.1 Access the Web Configuration Menu
The HandyTone-496 HTML configuration menu can be accessed via LAN or WAN port:
• From the LAN port:
ƒ
ƒ
ƒ
ƒ
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
ƒ
Open a web browser, type in the default gateway IP address. You will see the login
page of the device.
• 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”
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option by accessing the configuration via LAN port. With the WAN side HTTP access enabled,
then get the WAN IP address of the HandyTone-496 through section 5.1 with menu option 02.
The HandyTone-496’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-496.
NOTE:
• To type IP address into browser to get into the configuration page, please strip out the leading
“0” as the browser will parse in octet. e.g.: if the IP address is: 192.168.001.014, please type in:
192.168.1.14.
6.2.2 End User Configuration
Once this HTTP request is entered and sent from a Web browser, the HandyTone-496 will respond
with the following login screen:
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 and administrator is
“123” and “admin” respectively. Only administrator can get access to the “ADVANCED SETTING”
configuration page.
NOTE:
If you cannot log into the configuration page by using default password, please check with the VoIP
service provider. Most likely the service provider has provisioned the device and configured for you.
After a correct password is entered in the login screen, the embedded Web server inside the
HandyTone-496 will respond with the Configuration page, which is explained in details below.
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Grandstream Device Configuration
STATUS BASIC SETTINGS ADVANCED SETTINGS FXS PORT1 FXS PORT2
End User
Password:
(purposely not displayed for security protection)
80
Web Port:
(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:
DHCP vendor class ID:
PPPoE account ID:
PPPoE password:
PPPoE Service Name:
0
0
0
0
Preferred DNS server:
.
.
.
statically configured as:
IP Address:
192 168
1
16
0
.
.
.
.
.
.
.
255 255
255
1
Subnet Mask:
Default Router:
DNS Server 1:
DNS Server 2:
.
.
.
.
.
192 168
1
.
204 181
101
0
4
.
0
0
0
.
GMT-5:00 (US Eastern Time, New York)
Time Zone:
Daylight
Savings Time:
4,1,7,2,0;10,-1,7,2,0;60
No
Yes Optional Rule:
NAT/DHCP Server Information & Configuration:
Device Mode:
NAT Router
Bridge
Cloned WAN MAC
Addr:
(in hex format)
255.255.255.0
192.168.2.1
LAN Subnet Mask:
(default is 255.255.255.0)
LAN DHCP Base
IP:
(base IP for the LAN port, default is 192.168.2.1)
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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
0
0
UDP Only
UDP Only
UDP Only
UDP Only
UDP Only
UDP Only
UDP Only
UDP Only
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
Update
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:
Cancel
Reboot
All Rights Reserved Grandstream Networks, Inc. 2005
End User
Password
This contains the password to access the Web Configuration Menu. This
field is case sensitive.
Web Port
This is the device’s internal HTTP server port. Default is 80.
There are 2 modes under which the HandyTone ATA can operate:
IP Address
- 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.
DHCP hostname
DHCP domain
This option specifies the name of the client. This field is optional but may be
required by some Internet Service Providers. Default is blank.
This option specifies the domain name that client should use when resolving
hostnames via the Domain Name System. Default is blank.
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DHCP vendor class ID This option is used by clients and servers to exchange vendor-specific
information. Default is blank.
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” and the Optional Rule is empty, then the
displayed time will be 1 hour ahead of normal time.
The “Automatic Daylight Saving Time Rule” shall have the following
syntax:
start-time;end-time;saving
Both
start-time
and
end-time
have
the
same
syntax:
month,day,weekday,hour,minute
month: 1,2,3,..,12 (for Jan, Feb, .., Dec)
day: [+|-]1,2,3,..,31
weekday: 1, 2, 3, .., 7 (for Mon, Tue, .., Sun), or 0 which means the daylight
saving rule is not based on week days but based on the day of the month.
hour: hour (0-23),
minute: minute (0-59)
If “weekday” is 0, it means the date to start or end daylight saving is at
exactly the given date. In that case, the “day” value must not be negative. If
“weekday” is not zero and “day” is positive, then the daylight saving starts
on the first “day”th iteration of the weekday (1st Sunday, 3rd Tuesday etc).
If “weekday” us not zero and “day” is negative, then the daylight saving
starts on the last “day”th iteration of the weekday (last Sunday, 3rd last
Tuesday etc).
The saving is in the unit of minutes. The saving time may also be preceded
by a negative (-) sign if subtraction is desired instead of addition.
The default value for “Automatic Daylight Saving Time Rule” shall be set to
“04,01,7,02,00;10,-1,7,02,00;60” which is the rule for US.
Examples
US/Canada where daylight saving time is applicable:
04,01,7,02,00;10,-1,7,02,00;60
This means the daylight saving time starts from the first Sunday of April at
2AM and ends the last Sunday of October at 2AM. The saving is 60 minutes
(1hour).
PPPoE account ID
PPPoE password
PPPoE username. Fill this field if your ISP requires you to use a PPPoE
(Point to Point Protocol over Ethernet) connection.
PPPoE account password.
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PPPoE Service Name This field is optional. If your ISP uses a service name for the PPPoE
connection, enter the service name here. Default is blank.
Device Mode
To use the device as a router or a bridge.
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-496 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 FXS PORT1 FXS PORT2
MAC Address: 00.0B.82.03.C8.AF
WAN IP Address: 192.168.1.187
Product Model: HT496
Software Version: Program-- 1.0.3.44 Bootloader-- 1.0.8.11 HTML-- 1.0.3.44 VOC-- 1.0.0.10
System Up Time: 2 day(s) 0 hour(s) 9 minute(s)
Registered: Yes
PPPoE Link Up: disabled
NAT: 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 web user interface, normally not changed.
VOC: This is the codec program, normally not changed.
System Up Time
Registered
This field indicates how long the device has been up since the last reboot.
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
NAT
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
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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.
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Grandstream Device Configuration
STATUS BASIC SETTINGS ADVANCED SETTINGS FXS PORT1 FXS PORT2
Admin Password:
(purposely not displayed for security protection)
Home NPA:
Layer 3 QoS:
48
(Diff-Serv or Precedence value)
0
0
Layer 2 QoS:
802.1Q/VLAN Tag
802.1p priority value
(URI or IP:port)
(0-7)
stun.mycompany.com
STUN server is :
keep-alive interval:
Use NAT IP:
20
(in seconds, default 20 seconds)
(used in SIP/SDP message if specified)
Firmware Upgrade and
Provisioning:
Upgrade Via
TFTP
HTTP
192.168.1.193
Firmware Server Path:
192.168.1.193
Config Server Path:
Firmware File Prefix:
Config File Prefix:
Firmware File Postfix:
Config File Postfix:
Automatic Upgrade:
10080
No
Yes, check for upgrade every
minutes (default 7 days)
Always Check for New Firmware
Check New Firmware only when F/W pre/suffix changes
Firmware Key:
(in Hexadecimal Representation)
800 ms
Onhook Threshold:
600 Ohm (North America)
FXS Impedance:
Caller ID Scheme:
Bellcore (North America)
36V
Onhook Voltage:
Polarity Reversal:
NTP Server:
No
Yes (reverse polarity upon call establishment and termination)
time.nist.gov
(URI or IP address)
Reply to ICMP on WAN port:
No
Yes (Unit will not respond to PING from WAN side if set to No)
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No
Yes (WAN side access to http server will be rejected if set to
WAN side http access:
No)
Syslog Server:
Syslog Level:
NONE
Update
Cancel
Reboot
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Two FXS SIP accounts each has its own configuration page. Their configurations are identical. The
following is a screen shot of FXS Port 1 Account settings.
Grandstream Device Configuration
STATUS BASIC SETTINGS ADVANCED SETTINGS FXS PORT1 FXS PORT2
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)
3123320
SIP User ID:
(the user part of an SIP address)
3123320
Authenticate ID:
(can be identical to or different from SIP User ID)
Authenticate Password:
(purposely not displayed for security protection)
John Doe
Name:
(optional, e.g., John Doe)
Use DNS SRV:
No
No
No
No
Yes
Yes
User ID is phone number:
SIP Registration:
Yes
Unregister On Reboot:
Register Expiration:
local SIP port:
Yes
60
(in minutes. default 1 hour, max 45 days)
(default 5060)
5060
5004
local RTP port:
(1024-65535, default 5004)
Yes
Use random port:
No
101
DTMF Payload Type:
Send DTMF:
in-audio
via RTP (RFC2833)
via SIP INFO
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Send Flash Event:
No
No
No
Yes (Flash will be sent as a DTMF event if set to Yes)
Enable Call Features:
Yes (if Yes, Call Forwarding & Call-Waiting-Disable are supported locally)
Yes (if Yes, *23 will be disabled)
Use Bell-style
3-way Conference:
Offhook Auto-Dial:
(User ID/extension to dial automatically when offhook)
Proxy-Require:
Disable Call-Waiting:
No
No
Yes
NAT Traversal (STUN):
No Key Entry Timeout:
Yes
4
(in seconds, default is 4 seconds)
Preferred Vocoder:
(in listed order)
current setting is " PCMU"
current setting is " PCMA"
current setting is " G723"
current setting is " G729"
current setting is " G726-32"
current setting is " iLBC"
choice 1:
choice 2:
choice 3:
choice 4:
choice 5:
choice 6:
2
Voice Frames per TX:
G723 rate:
(up to 10/20/32/64 for G711/G726/G723/other codecs respectively)
6.3kbps encoding rate
20ms 30ms
5.3kbps encoding rate
iLBC frame size:
iLBC payload type:
97
(between 96 and 127, default is 97)
Yes
Silence Suppression:
Fax Mode:
No
T.38 (Auto Detect)
Pass-Through
Early Dial:
No
Yes (use "Yes" only if proxy supports 484 response)
(this prefix string is added to each dialed number)
Yes (if set to Yes, "#" will function as the "(Re-)Dial" key)
Dial Plan Prefix:
Use # as Dial Key:
No
SUBSCRIBE for MWI:
No, do not send SUBSCRIBE for Audible Ringing Indication
Yes, send periodical SUBSCRIBE for Audible Ringing Indication
Send Anonymous:
No
No
Yes (caller ID will be blocked if set to Yes)
Lock keypad update:
Yes (configuration update via keypad is disabled if set to Yes)
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Special Feature:
Standard
Frequency1(Hz) Frequency2(Hz) ON (x10ms)
OFF (x10ms) Alert-info
350
350
350
350
440
480
480
1400
440
440
440
440
440
440
20
440
440
440
440
480
620
620
2600
440
440
440
440
440
440
20
0
0
Dial Tone
10
10
Recall Dial Tone
10
10
Message Waiting
10
10
Confirmation
200
50
400
Audible Ringing
50
Busy Tone
25
25
Reorder Tone
10
10
Receiver offhook
100
100
100
100
100
100
80
100
Ring1
100
Distinct Ring Tones:
Ring2
100
Ring3
100
Ring4
100
Ring5
100
Ring6
40
Ring7
20
20
40
20
Ring8
20
20
30
20
Ring9
20
20
50
800
Ring10
20
20
200
200
400
Ring11
20
20
400
Ring12
0dB default
0dB default
Reboot
Volume Amplification:
TX
RX
Update
Cancel
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Admin Password
This contains the password to access the Advanced Web Configuration page.
This field is case sensitive.
Home NPA
Local area code for North American Dial Plan.
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.
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Layer 2 QoS
This setting includes two fields. The 802.1Q/VLAN Tag contains the value
used for layer 2 VLAN tag. Default setting is blank. And 802.1p priority value
contains the value of the priority value.
STUN Server
STUN server’s IP or FQDN
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 is used in SIP/SDP message. Default is blank.
Firmware Upgrade Default method is HTTP. Firmware upgrade may take up to 10 minutes
and Provisioning
depending on network environment. Do not interrupt the firmware upgrading
process.
Firmware Server
Path
IP address or domain name of firmware server.
Config Server Path IP address or domain name of configuration server.
Firmware File
Prefix
Default is blank. If configured, HandyTone-496 will request the firmware file
with the prefix. This setting is useful for ITSPs. End user should keep it blank.
Firmware File
Postfix
Default is blank. End user should keep it blank.
Config File Prefix
Default is blank. End user should keep it blank.
Config File Postfix Default is blank. End user should keep it blank.
Automatic Upgrade Default is “Yes”.
Firmware Key
For firmware encryption. It should be 32 digit in Hexadecimal Representation.
End user should keep it blank.
Onhook Threshold The amount of time the hookflash is pressed that will cause the device to
onhook. Default is 800ms.
FXS Impedance
Selects the impedance of the analog telephone connected to the Phone port.
Caller ID Scheme
• Bellcore (North America)
• CID-Canada
• DTMF-Brazil
• DTMF-Sewden
• DTMF (Denmark)
• ETSI-DTMF (Finland, Sweden)
• ETSI-FSK (France, Germany, Norway, Taiwan, UK-CCA)
Onhook Voltage
Select the onhook voltage to suit different area or PBX.
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Polarity Reversal
NTP server
Default is No. If set to Yes, polarity will be reversed upon call establishment
and termination.
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.
Reply to ICMP on When this parameter is set to “No”, the HandyTone ATA will not respond to
WAN port
PING from WAN side.
WAN side http
access
If this parameter is set to “No”, the HTML configuration update via WAN port
is disabled.
Syslog Server
The IP address or URL of System log server. This feature is especially useful
for ITSP (Internet Telephone Service Provider)
Syslog Level
Select the ATA to report the log level. Default is NONE. The level is one of
DEBUG, INFO, WARNING or ERROR. Syslog messages are sent based on
the following events:
•
•
•
•
•
•
•
•
•
•
product model/version on boot up (INFO level)
NAT related info (INFO level)
sent or received SIP message (DEBUG level)
SIP message summary (INFO level)
inbound and outbound calls (INFO level)
registration status change (INFO level)
negotiated codec (INFO level)
Ethernet link up (INFO level)
SLIC chip exception (WARNING and ERROR levels)
memory exception (ERROR level)
The Syslog uses USER facility. In addition to standard Syslog payload, it
contains the following components:
GS_LOG: [device MAC address][error code] error message
Here is an example: May 19 02:40:38 192.168.1.14 GS_LOG:
[00:0b:82:00:a1:be][000] Ethernet link is up
Individual Account Settings
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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
SIP User ID
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.
This field contains the user part of the SIP address for this phone. e.g., if the
SIP address is: sip:my_user_id@my_provider.com, then the SIP User ID is:
my_user_id. Please do NOT include the preceding “sip:” scheme or the host
portion of the SIP address in this field.
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.
Default is No. If set to Yes the client will use DNS SRV for server lookup.
Use DNS SRV:
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.
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’s registration information will be
cleared on reboot.
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
This parameter defines the local SIP port the HandyTone ATA will listen and
transmit. The default value for FXS port 1 is 5060. The default value for FXS
port 2 is 5062.
Local RTP port
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 1 is 5004. The default value for FXS port 2 is 5008.
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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.
DTMF Payload
Type
This parameter sets the payload type for DTMF using RFC2833.
Send DTMF
This parameter specifies the mechanism to transmit DTMF digit. There are 3
modes supported: in audio which means DTMF is combined in audio signal
(not very reliable with low-bit-rate codec), via RTP (RFC2833), or via SIP
INFO. Multiple selections of DTMF method are supported.
Send Flash Event
Default is NO. If set to yes, flash will be sent as DTMF event.
Enable Call
Features
Default is No. If set to Yes, Call Forwarding & Do-Not-Disturb are supported
locally
Use Bell-style
3-way Conference
If this parameter is set to “Yes”, user will be able to make Bellcore style 3-way
conference. *23 will be disabled.
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.
Proxy-Require
SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.
Default is No.
Disable Call
Waiting
NAT Traversal
(STUN)
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.
No Key Entry
Timeout
Default is 4 seconds.
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Preferred Vocoder The HandyTone ATA supports up to 6 different Vocoder types including G.711
A-/U-law，G.723.1, G.726, 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 6”.
G723 Rate:
This defines the encoding rate for G723 vocoder. By default, 6.3kbps rate is
chosen.
iLBC frame size:
This sets the iLBC frame size in 20ms or 30ms
iLBC payload type: This defines payload time for iLBC. Default value is 97. The valid range is
between 96 and 127.
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.
Silence Suppression This controls the silence suppression/VAD feature of G723. If set to “Yes”,
when a silence is detected, small quantity of VAD packets (instead of audio
packets) will be sent during the period of no talking. If set to “No”, this feature
is disabled.
Fax Mode
T.38 (Auto Detect) FoIP by default, or Pass-Through (must use codec
PCMU/PCMA)
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Early Dial
This parameter controls whether the phone will attempt to send an early
INVITE each time a key is pressed when a user dials a number. If set to “Yes”,
an INVITE is sent using the dial-number collected thus far; Otherwise, no
INVITE is sent until the “(Re-)Dial” button is pressed or after about 5 seconds
have elapsed if the user forgets to press the “(Re-)Dial” button.
The “Yes” option should be used ONLY if there is a SIP proxy configured and
the proxy server supports 484 Incomplete Address response. Otherwise, the call
will most likely be rejected by the proxy (with a 404 Not Found error).
Please note that this feature is NOT designed to work with and should NOT be
enabled for direct IP-to-IP calling.
Dial Plan Prefix
Sets the prefix added to each dialed number
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.
Subscribe for MWI: Default is No. When set to “Yes” a SUBSCRIBE for Message Waiting
Indication will be sent periodically.
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
Default is Standard. Choose the selection to meet some special requirements
from Soft Switch vendors like Lucent FS5000 Simple Endpoint, CBCOM, etc.
Distinct Ring Tones Using these settings, user can configure ring or tone frequencies according to
their preference. By default they are set to North American frequencies.
Frequencies should be configured with known values to avoid uncomfortable
high pitch sounds. ON is the period of ringing (“On time” in ‘ms’) while OFF
is the period of silence. In order to set a continuous ring, OFF should be zero.
Otherwise it will ring ON ms and a pause of OFF ms and then repeat the
pattern.
Ring 1 - 6 are reserved for distinct call waiting tones through SIP Alert-Info.
Ring 7 – 12 are distinct ring tones through SIP Alert-Info. Both client and
Server need to support this to be a useful feature. Currently, distinct ring tone
feature is only supported under Lucent Special Feature.
Volume
Amplification
Handset volume adjustment. RX is for receiving volume, TX is for
transmission volume. Default values are 0dB for both parameters. +6dB
generates the highest volume and -6dB generates the lowest volume.
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6.2.4 Saving the Configuration Changes
Once a change is made, users should click on the “Update” button in the Configuration page. The
HandyTone-496 will then display the following screen to confirm that the changes have been saved.
Grandstream Device Configuration
STATUS BASIC SETTINGS ADVANCED SETTINGS FXS PORT1 FXS PORT2
Your configuration changes have been saved.
They will take effect on next reboot.
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Users are recommended to power cycle the HandyTone-496 after seeing the above message.
6.2.5 Rebooting the HandyTone-496 from Remote
The administrator of the HandyTone-496 can remotely reboot the HandyTone ATA by clicking on the
“Reboot” button at the bottom of the configuration page. Once done, the following screen will be
displayed to indicate that rebooting is underway.
Grandstream Device Configuration
The device is rebooting now...
You may relogin by clicking on the link below in 30 seconds.
Click to relogin
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At this point, the user can relogin to the HandyTone-496 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 file,
“cfg000b82xxxxxx”, where “000b82xxxxxx” is the MAC address of the HandyTone ATA.
The configuration files can be downloaded via TFTP or HTTP from the central server. A service
provider or an enterprise with large deployment of HandyTone ATA can easily manage the
configuration and service provisioning of individual devices remotely from a central server.
Grandstream provides a licensed provisioning system called GAPS that can be used to support
automated configuration of HandyTone ATA. 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 provide GAPS (Grandstream Automated Provisioning System) service to VoIP service
providers. It could be either simple redirection or with certain special provisioning settings. Initially
upon booting up, Grandstream devices by default point to Grandstream provisioning server GAPS,
based on the unique MAC address of each device, GAPS provision the devices with redirection
settings so that they will be redirected to customer’s TFTP or HTTP server for further provisioning.
Grandstream also provide GAPSLite software package which contains our NAT friendly TFTP server
and a configuration tool to facilitate the task of generating device configuration files.
The GAPSLite configuration tool is now free to end users. The tool and configuration template are
available for download from http://www.grandstream.com/DOWNLOAD/Configuration_Tool/.
For details on how GAPS works, please contact Grandstream and refer to the documentation of GAPS
product provided.
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7 Software Upgrade
Software upgrade can be done via either TFTP or HTTP. The corresponding configuration settings are
in the ADVANCED SETTINGS configuration page.
7.1
Firmware Upgrade through TFTP/HTTP
To upgrade via TFTP or HTTP, the “Firmware Upgrade and Provisioning upgrade via” field needs to
be set to TFTP or HTTP, respectively. “Firmware Server Path” needs to be set to a valid URL of a
TFTP or HTTP server, server name can be in either FQDN or IP address format. Here are examples of
some valid URL.
e.g. firmware.mycompany.com:6688/Grandstream/1.0.3.44
e.g. 168.75.215.190
NOTES:
• TFTP server in IP address format can be configured via IVR. Please refer to section 6.1.3 for
instructions. If TFTP server is in FQDN format, it must be set via web configuration interface.
• Once a “Firmware Server Path” is set, user needs to update the settings and reboot the device.
If the configured firmware server is found and a new code image is available, the HandyTone
ATA will attempt to retrieve the new image files by downloading them into the HandyTone
ATA’s SRAM. During this stage, the HandyTone ATA’s LEDs will blink until the
checking/downloading process is completed. Upon verification of checksum, the new code
image will then be saved into the Flash. If TFTP/HTTP fails for any reason (e.g., TFTP/HTTP
server is not responding, there are no code image files available for upgrade, or checksum test
fails, etc), the HandyTone ATA will stop the TFTP/HTTP process and simply boot using the
existing code image in the flash.
• Firmware upgrade may take as long as 1 to 20 minutes over Internet, or just 20+ seconds if it is
performed on a LAN. It is recommended to conduct firmware upgrade in a controlled LAN
environment if possible. For users who do not have a local firmware upgrade server,
Grandstream provides a NAT-friendly TFTP server on the public Internet for firmware
upgrade. Please check the Services section of Grandstream’s Web site to obtain our public
TFTP server’s IP address.
• Alternatively, user can download a free TFTP or HTTP server and conduct local firmware
upgrade. A free windows version TFTP server is available for download from
them under the root directory of the TFTP server. Put the PC running the TFTP server and the
HandyTone ATA in the same LAN segment. Please go to File -> Configure -> Security to
change the TFTP server's default setting from "Receive Only" to "Transmit Only" for the
firmware upgrade. Start the TFTP server, in the HandyTone ATA’s web configuration page,
configure the Firmware Server Path with the IP address of the PC, update the change and
reboot the unit. Please be advised that our client will pull out firmware from the WAN side, if
the TFTP server is connected to the device’s LAN port, the firmware upgrade will not work by
design.
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7.2 Configuration File Download
Grandstream Networks, Inc.
Grandstream SIP Device can be configured via Web Interface as well as via Configuration File
through TFTP or HTTP. “Config Server Path” is the TFTP or HTTP server path for configuration file.
It needs to be set to a valid URL, either in FQDN or IP address format. The “Config Server Path” can
be same or different from the “Firmware Server Path”.
A configuration parameter is associated with each particular field in the web configuration page. A
parameter consists of a Capital letter P and 2 to 3 (Could be extended to 4 in the future) digit numeric
numbers. i.e., P2 is associated with “Admin Password” in the ADVANCED SETTINGS page. For a
detailed parameter list, please refer to the corresponding firmware release configuration template.
When Grandstream Device boots up or reboots, it will issue request for configuration file named
“cfgxxxxxxxxxxxx”, where “xxxxxxxxxxxx” is the MAC address of the device, i.e.,
“cfg000b820102ab”. The configuration file name should be in lower cases.
7.3
Firmware and Configuration File Prefix and Postfix
Starting from firmware version 1.0.3.18 for HandyTone-496, adding prefix and postfix for both
firmware and configuration file is supported.
Firmware Prefix and Postfix allows device to download the firmware name with the matching Prefix
and Postfix. This makes it the possible to store ALL of the firmware with different version in one
single directory. Similarly, Config File Prefix and Postfix allows device to download the configuration
file with the matching Prefix and Postfix. Thus multiple configuration files for the same device can be
stored in one directory.
In addition, when the field “Check New Firmware only when F/W pre/suffix changes” is set to “Yes”,
the device will only issue firmware upgrade request if there are changes in the firmware Prefix or
Postfix.
7.4
Managing Firmware and Configuration File Download
When “Automatic Upgrade” is set to “Yes”, Service Provider can use P193 (Auto Check Interval, in
minutes, default and minimum is 60 minutes) to have the devices periodically check with either
Firmware Server or Config Server, whenever they are defined. This allows the device periodically
check if there are any new changes need to be taken on a scheduled time. By defining different
intervals in P193 for different devices, Server Provider can spread the Firmware or Configuration File
download in minutes to reduce the Firmware or Provisioning Server load at any given time.
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8 Restore Factory Default Setting
Warning !!!
Restore the Factory Default Setting will DELETE all configuration information of the device.
Please backup or print out all the settings before you approach to following steps. Grandstream will
not take any responsibility if you lose all the parameters of setting and cannot connect to your
service provider.
Please disconnect network cable and power cycle the unit before trying to reset the unit to factory
default. The steps are as follows:
Step 1:
Find the MAC address of the device. It is a 12 digits HEX number located on the bottom of the
unit.
Step 2:
Encode the MAC address. Please use the following mapping:
0-9: 0-9
A: 22
B: 222
C: 2222
D: 33
E: 333
F: 3333
For example, if the MAC address is 000b8200e395, it should be encoded as
“0002228200333395”.
Step 3:
To perform factory reset:
a. Press “***” or the LED button for voice prompt.
b. Enter “99” and get the voice prompt “Reset”.
c. Enter the encoded MAC address of the device.
d. Wait for 15 seconds.
The device will reboot automatically and restore to factory default setting.
NOTE:
• Please be aware by default the HandyTone-496 WAN side HTTP access is disabled. After a
factory reset, the device’s web configuration page can be accessed only from its LAN port,
please refer to instructions in section 6.2.1 for details.
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9 Glossary of Terms
ADSL
AGC
ARP
Asymmetric Digital Subscriber Line: Modems attached to twisted pair copper wiring that
transmit from 1.5 Mbps to 9 Mbps downstream (to the subscriber) and from 16 kbps to 800
kbps upstream, depending on line distance.
Automatic Gain Control, is an electronic system found in many types of devices. Its purpose is
to control the gain of a system in order to maintain some measure of performance over a
changing range of real world conditions.
Address Resolution Protocol is a protocol used by the Internet Protocol (IP) [RFC826],
pecifically IPv4, to map IP network addresses to the hardware addresses used by a data link
protocol. The protocol operates below the network layer as a part of the interface between the
OSI network and OSI link layer. It is used when IPv4 is used over Ethernet
ATA
Analogue Telephone Adapter. Covert analogue telephone to be used in data network for VoIP,
like Grandstream HT series products.
CODEC
Abbreviation for Coder-Decoder. It's an analog-to-digital (A/D) and digital-to-analog (D/A)
converter for translating the signals from the outside world to digital, and back again.
CNG
Comfort Noise Generator, geneate artificial background noise used in radio and wireless
communications to fill the silent time in a transmission resulting from voice activity detection.
DATAGRAM
A data packet carrying its own address information so it can be independently routed from its
source to the destination computer
DECIMATE
To discard portions of a signal in order to reduce the amount of information to be encoded or
compressed. Lossy compression algorithms ordinarily decimate while subsampling.
DECT
Digital Enhanced Cordless Telecommunications: A standard developed by the European
Telecommunication Standard Institute from 1988, governing pan-European digital mobile
telephony. DECT covers wireless PBXs, telepoint, residential cordless telephones, wireless
access to the public switched telephone network, Closed User Groups (CUGs), Local Area
Networks, and wireless local loop. The DECT Common Interface radio standard is a
multicarrier time division multiple access, time division duplex (MC-TDMA-TDD) radio
transmission technique using ten radio frequency channels from 1880 to 1930 MHz, each
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divided into 24 time slots of 10ms, and twelve full-duplex accesses per carrier, for a total of
120 possible combinations. A DECT base station (an RFP, Radio Fixed Part) can transmit all
12 possible accesses (time slots) simultaneously by using different frequencies or using only
one frequency. All signaling information is transmitted from the RFP within a multiframe (16
frames). Voice signals are digitally encoded into a 32 kbit/s signal using Adaptive Differential
Pulse Code Modulation.
DNS
DID
Short for Domain Name System (or Service or Server), an Internet service that translates
domain names into IP addresses
Direct Inward Dialing
Direct Inward Dialing. The ability for an outside caller to dial to a PBX extension without
going through an attendant or auto-attendant.
DSP
Digital Signal Processing. Using computers to process signals such as sound, video, and other
analog signals which have been converted to digital form.
Digital Signal Processor. A specialized CPU used for digital signal processing.
Grandstream products all have DSP chips built inside.
DTMF
Dual Tone Multi Frequency
The standard tone-pairs used on telephone terminals for dialing using in-band signaling. The
standards define 16 tone-pairs (0-9, #, * and A-F) although most terminals support only 12 of
them (0-9, * and #).
FQDN
FXO
Fully Qualified Domain Name
A FQDN consists of a host and domain name, including top-level domain. For example,
www.grandstream.com is a fully qualified domain name. www is the host, grandstream is the
second-level domain, and.com is the top level domain.
Foreign eXchange Office
An FXO device can be an analog phone, answering machine, fax, or anything that handles a
call from the telephone company like AT&T. They should also operate the same way when
connected to an FXS interface.
An FXO interface will accept calls from FXS or PSTN interfaces. All countries and regions
have their own standards.
FXO is complimentary to FXS (and the PSTN).
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FXS
Foreign eXchange Station
An FXS device has hardware to generate the ring signal to the FXO extension (usually an
analog phone).
An FXS device will allow any FXO device to operate as if it were connected to the phone
company. This makes your PBX the POTS+PSTN for the phone.
The FXS Interface connects to FXO devices (by an FXO interface, of course).
DHCP
The Dynamic Host Configuration Protocol (DHCP) is an Internet protocol for automating the
configuration of computers that use TCP/IP. DHCP can be used to automatically assign IP
addresses, to deliver TCP/IP stack configuration parameters such as the subnet mask and
default router, and to provide other configuration information such as the addresses for printer,
time and news servers.
ECHO CANCELLATION
Echo Cancellation is used in telephony to describe the process of removing echo from a voice
communication in order to improve voice quality on a telephone call. In addition to improving
quality, this process improves bandwidth savings achieved through silence suppression by
preventing echo from traveling across a network.
There are two types of echo of relevance in telephony: acoustic echo and hybrid echo. Speech
compression techniques and digital processing delay often contribute to echo generation in
telephone networks.
H.323
HTTP
A suite of standards for multimedia conferences on traditional packet-switched networks.
Hyper Text Transfer Protocol; the World Wide Web protocol that performs the request and
retrieve functions of a server
IP
Internet Protocol. A packet-based protocol for delivering data across networks.
IP-PBX
IP-based Private Branch Exchange
IP Telephony
(Internet Protocol telephony, also known as Voice over IP Telephony) A general term for the
technologies that use the Internet Protocol's packet-switched connections to exchange voice,
fax, and other forms of information that have traditionally been carried over the dedicated
circuit-switched connections of the public switched telephone network (PSTN). The basic steps
involved in originating an IP Telephony call are conversion of the analog voice signal to digital
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format and compression/translation of the signal into Internet protocol (IP) packets for
transmission over the Internet or other packet-switched networks; the process is reversed at the
receiving end. The terms IP Telephony and Internet Telephony are often used to mean the
same; however, they are not 100 per cent interchangeable, since Internet is only a subcase of
packet-switched networks. For users who have free or fixed-price Internet access, IP Telephony
software essentially provides free telephone calls anywhere in the world. However, the
challenge of IP Telephony is maintaining the quality of service expected by subscribers.
Session border controllers resolve this issue by providing quality assurance comparable to
legacy telephone systems.
IVR
IVR is a software application that accepts a combination of voice telephone input and touch-
tone keypad selection and provides appropriate responses in the form of voice, fax, callback, e-
mail and perhaps other media.
MTU
A Maximum Transmission Unit (MTU) is the largest size packet or frame, specified in octets
(eight-bit bytes), that can be sent in a packet- or frame-based network such as the Internet. The
maximum for Ethernet is 1500 byte.
NAT
NTP
Network Address Translation
Network Time Protocol, a protocol to exchange and synchronize time over networks
The port used is UDP 123
Grandstream products using NTP to get time from Internet
OBP/SBC
Outbound Proxy or another name Session Border Controller. A device used in VoIP networks.
OBP/SBCs are put into the signaling and media path between calling and called party. The
OBP/SBC acts as if it was the called VoIP phone and places a second call to the called party.
The effect of this behaviour is that not only the signaling traffic, but also the media traffic
(voice, video etc) crosses the OBP/SBC. Without an OBP/SBC, the media traffic travels
directly between the VoIP phones. Private OBP/SBCs are used along with firewalls to enable
VoIP calls to and from a protected enterprise network. Public VoIP service providers use
OBP/SBCs to allow the use of VoIP protocols from private networks with internet connections
using NAT.
PPPoE
PSTN
Point-to-Point Protocol over Ethernet, is a network protocol for encapsulating PPP frames in
Ethernet frames. It is used mainly with cable modem and DSL services.
Public Switched Telephone Network
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i.e. the phone service we use for every ordinary phone call, or called POT (Plain Old
Telephone), or circuit switched network.
RTCP
RTP
Real-time Transport Control Protocol, defined in RFC 3550, a sister protocol of the Real-time
Transport Protocol (RTP), It partners RTP in the delivery and packaging of multimedia data,
but does not transport any data itself. It is used periodically to transmit control packets to
participants in a streaming multimedia session. The primary function of RTCP is to provide
feedback on the quality of service being provided by RTP.
Real-time Transport Protocol defines a standardized packet format for delivering audio and
video over the Internet. It was developed by the Audio-Video Transport Working Group of the
IETF and first published in 1996 as RFC 1889
SDP
SIP
Session Description Protocol, is a format for describing streaming media initialization
parameters. It has been published by the IETF as RFC 2327.
Session Initiation Protocol, An IP telephony signaling protocol developed by the IETF
(RFC3261). SIP is a text-based protocol suitable for integrated voice-data applications. SIP is
designed for voice transmission and uses fewer resources and is considerably less complex than
H.323.
All Grandstream products are SIP based
STUN
TCP
Simple Traversal of UDP over NATs, is a network protocol allowing clients behind NAT (or
multiple NATs) to find out its public address, the type of NAT it is behind and the internet side
port associated by the NAT with a particular local port. This information is used to set up UDP
communication between two hosts that are both behind NAT routers. The protocol is defined in
RFC 3489. STUN will usually work good with non-symmetric NAT routers.
Transmission Control Protocol, is one of the core protocols of the Internet protocol suite. Using
TCP, applications on networked hosts can create connections to one another, over which they
can exchange data or packets. The protocol guarantees reliable and in-order delivery of sender
to receiver data.
TFTP
UDP
Trivial File Transfer Protocol, is a very simple file transfer protocol, with the functionality of a
very basic form of FTP; It uses UDP (port 69) as its transport protocol.
User Datagram Protocol (UDP) is one of the core protocols of the Internet protocol suite. Using
UDP, programs on networked computers can send short messages known as datagrams to one
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another. UDP does not provide the reliability and ordering guarantees that TCP does;
datagrams may arrive out of order or go missing without notice. However, as a result, UDP is
faster and more efficient for many lightweight or time-sensitive purposes.
VAD
Voice Activity Detection or Voice Activity Detector is an algorithm used in speech processing
wherein, the presence or absence of human speech is detected from the audio samples.
VLAN
VoIP
A virtual LAN, known as a VLAN, is a logically-independent network. Several VLANs can co-
exist on a single physical switch. It is usually refer to the IEEE 802.1Q tagging protocol.
Voice over IP
VoIP encompasses many protocols. All the protocols do some form of signalling of call
capabilities and transport of voice data from one point to another. e.g: SIP, H.323, etc.
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