User Manual
HandyTone-386
Analog Telephone Adaptor
For SW Release Version 1.0.2.5
Grandstream Networks, Inc.
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HandyTone-386 User Manual
Grandstream Networks, Inc.
1 Welcome
Congratulations on becoming an owner of HandyTone-386. You made an excellent choice and we
hope you enjoy all of its capabilities.
Grandstream's HandyTone-386 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-386 is fully compatible with SIP industry standard and can interoperate with many other
SIP compliant devices and software on the market.
Grandstream HandyTone-386 is a new addition to the popular HandyTone product family. The new
HandyTone-386 features two FXS ports each with independent SIP accounts.
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2 Installation
HandyTone-386 Analog Telephone Adaptor is an all-in-one VoIP integrated device designed to be a
total solution for networks providing VoIP services.
The HandyTone-386 VoIP functionalities are available via regular analog telephones.
The following photo illustrates the appearance of a HandyTone-386.
Top View
Side Views
RJ11
Line Port
(Line )
+5V/1200mA
RJ45
10M Ethernet
WAN
BUTTON
RED LED
GREEN LED
RJ11
FXS Port
(Phone)
RJ11
FXS Port
(Fax)
Interconnection Diagram of the HandyTone-386:
Internet ADSL/Cable
Modem Ethernet
Analog Phone
Analog Phone
WAN
FXS 1
FXS 2
Cordless Phone
Cordless Phone
LINE
Phone Line
Fax
Fax
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HandyTone-386 has two FXS ports. The RJ11 jack next to the WAN is called FXS port 2 and the
RJ11 jack on the corner is called FXS port 1. The RJ11 jack on the side on of the HandyTone-386 is a
LINE port. Each FXS port can have a separate SIP account. This is a key feature of HandyTone-386.
Both ports can make calls concurrently.
Following are the steps to install a HandyTone-386:
1. Connect a standard touch-tone analog telephone (or fax machine) to FXS port 1.
2. Connect another standard touch-tone analog telephone (or fax machine) to FXS port 2.
3. Insert a standard telephone cable into the LINE port of HandyTone-386 and connect the other
end of the telephone cable to a wall jack.
4. Insert the Ethernet cable into the WAN port of HandyTone-386 and connect the other end of
the Ethernet cable to an uplink port (a router or a modem, etc.)
5. Insert the power adapter into the HandyTone-386 and connect it to a wall outlet.
Please follow the instructions in section 6.2.1 to configure the HandyTone-386.
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3 What is Included in the Package
The HandyTone-386 package contains:
1) One HandyTone-386
2) One universal power adaptor
3) One Ethernet cable
3.1 Safety Compliances
The HandyTone-386 is compliant with various safety standards including FCC/CE and C-tick. Its
power adaptor is compliant with UL standard. The HandyTone-386 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-386 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.
• 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
(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, Call Waiting Caller ID, in-band and out-of-band DTMF, Dial Plans, etc.
• Support fax pass through and T.38
• 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 (pending) with encrypted configuration files.
• Ultra compact (wallet size) and lightweight design, great companion for travelers
• Compact, lightweight Universal Power adapter.
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4.2 Hardware Specification
The table below lists the hardware specification of HandyTone-386.
Model
HandyTone-386
WAN interface
FXS telephone port
PSTN Port
Button
1xRJ45 10Base-T
2xFXS
1x PSTN pass-through or life line port
1
LED
Green and red color
Universal Switching
Power Adaptor
Input: 100-240VAC 50-60 Hz
Output: +5VDC, 1200mA,
UL certified
Dimension
70mm (W)
130mm (D)
27mm (H)
Weight
0.6lbs (0.3kg)
Temperature
40 - 130oF
5 – 45oC
Humidity
10% - 90%
(non-condensing)
Compliance
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5 Basic Operations
5.1 Get Familiar with Voice Prompt
HandyTone-386 has stored a voice prompt menu 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 press the button or “***” from the analog phone.
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
01
02
“DHCP Mode”,
“Static IP Mode”
“IP Address “ + IP address
The current WAN IP address is announced
Enter 12 digit new IP address if in Static IP
Mode
03
04
05
06
47
“Subnet “ + IP address
“Gateway “ + IP address
“DNS Server “ + IP address
“TFTP Server “ + IP address
“Direct IP Calling”
Same as menu 02
Same as menu 02
Same as menu 02
Same as menu 02
When entered, you will be prompted a dial
tone, then enter 12 digit IP address
This menu can also be entered by pressing the
button again
86
99
“Voice Messages Pending”
“No Voice Messages”
“RESET”
Enter “9” to dial pre-configured phone number
to retrieve VM
Enter “9” to reboot the phone
Enter encoded MAC address to restore factory
default setting
“Invalid Entry”
Automatically returns to main menu
Notes:
• Once the button is pressed, it enters the voice prompt main menu. If the button is pressed again,
while it is already in the voice prompt menu, it jumps to “Direct IP Call” option and a dial tone
is prompted
• “*” 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
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5.2 Make Phone Calls
5.2.1 Calling phone or extension numbers
To make a phone or extension number call:
a) Dial the number directly and wait for 4 seconds (default “No Key Entry Timeout”). Or
b) Dial the number directly, and press # (assuming that “Use # as dial key” is selected in web
configuration).
Other functions available during the call are call-waiting/flash, call-transfer, and call-forward.
5.2.2 Direct IP calls
Direct IP calling allows two phones, that is, a HandyTone with an analog phone and another VoIP
Device, to talk to each other in an ad hoc fashion without a SIP proxy. VoIP calls can be made between
two phones if:
• Both HandyTone ATA and the other VoIP device (i.e., another HandyTone ATA or other SIP
products) have public IP addresses, or
• Both HandyTone ATA and the other VoIP device (i.e., another HandyTone ATA or other SIP
produces) are on the same LAN using private or public IP addresses, or
• Both HandyTone ATA and the other VoIP device (i.e., another HandyTone ATA or other SIP
products) can be connected through a router using public or private IP addresses.
To make a direct IP call, first pick up the analog phone or turn on the speakerphone on the analog
phone, follow Section 5.1 with voice prompt 47, followed by the 12-digit target IP address. Destination
ports can be specified by using “*4” (encoding for “:”) followed by the encoded port number.
Following is a table of the encoding scheme for the most commonly used characters:
Input
Meaning
00
01
02
03
04
05
06
07
08
09
*0
*4
0
1
2
3
4
5
6
7
8
9
. (dot character)
: (column character)
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Examples:
If the target IP address is 192.168.0.160, the dialing convention is
Voice Prompt with option 47, then 192168000160
followed by pressing the “#” key if it is configured as a send key or wait 4 seconds. In this case, the
default destination port 5060 is used if no port is specified.
If the target IP address/port is 192.168.1.20:5062, then the dialing convention would be:
Voice Prompt with option 47, then 192168001020*45062 followed by pressing the “#” key if it is
configured as a send key or wait for 4 seconds.
5.2.3 Blind Transfer
Assuming that call party A and B are in conversation. A wants to Blind Transfer B to C:
1. A presses FLASH (on the analog phone, or Hook Flash for old model phones) to get a dial
tone.
2. Then A dials *87 then dials C’s number, and then #(or wait for 4 seconds)
3. A can hang up.
Note: Call Feature has to be set to YES.
A can hold on to the phone and wait for one of the three following behaviors:
• A quick confirmation tone (temporarily using the call waiting indication tone) followed by a
dial tone. This indicates the transfer is successful (transferee has received a 200 OK from
transfer target). At this point, A can either hang up or make another call.
• A quick busy tone followed by a restored call (on supported platforms only). This means the
transferee has received a 4xx response for the INVITE and we will try to recover the call. The
busy tone is just to indicate to the transferor that the transfer has failed.
• Busy tone keeps playing. This means we have failed to receive the second NOTIFY from the
transferee and decided to time out. Note: this does not indicate the transfer has been
successful, nor does it indicate the transfer has failed. When transferee is a client that does not
support the second NOTIFY (such as our own earlier firmware), this will be the case. In bad
network scenarios, this could also happen, although the transfer may have been completed
successfully.
5.2.4 Attended Transfer
Assuming that call party A and B are in conversation. A wants to Attend Transfer B to C:
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). A and C are now in conversation.
3. A can hang up.
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Note: When intended Transfer failed, if A hangs up, the HandyTone-386 will ring user A again to
remind A that B is still on the call, by pressing FLASH or Hook again will restore the conversation
between A and B.
5.2.5 3-way Conferencing
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). A and C are now in conversation.
3. A presses FLASH again to begin conference.
5.2.6 Send and Receive PSTN Calls Through PSTN Line Port
User can send and receive calls from PSTN. To receive PSTN calls, simply make phone off hook when
the analog phone rings. To make a PSTN call, simply press *00 (default, can be configured in web
configuration page) to switch to the PSTN line and get dial tone and call out using the PSTN line.
5.3 Call Features
5.3.1 Call Features Table
Following table shows the call features of HandyTone-386.
Key
*30
*31
*67
*82
*50
*51
*70
*71
*72
Call Features
Block Caller ID (for all subsequent calls)
Send Caller ID (for all subsequent calls)
Block Caller ID (per call)
Send Caller ID (per call)
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
Busy Call Forward
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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
*91
*92
*93
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
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.
5.3.2 PSTN Pass Through / Life Line
When HandyTone-386 is out of power, the RJ 11 line jack on the HandyTone-486 side will function as
a pass through jack. The user will be able to use the same analog phone for PSTN calls.
5.4 FAX Support
HandyTone-286 supports FAX in two modes: T.38 (Fax over IP) and fax pass through. T.38 is the
preferred method because it is more reliable and works well in most network conditions. If the service
provider supports T.38, please use this method by selecting Fax mode to be T.38. If the service
provider does not support T.38, pass-through mode may be used. To send or receive faxes in fax pass
through mode, users will need to select all the Preferred Codecs to be PCMU/PCMA.
5.5 LED Light Pattern Indication
Following tables show the LED light pattern indication.
RED LED indicates abnormal status
DHCP Failed or WAN No Cable
HandyTone-386 fails to register
Button flashes every 2 seconds (if DHCP is configured)
Button flashes every 2 seconds (if SIP server is configured)
GREEN LED indicates normal working status
Message Waiting Indication
RINGING
RINGING INTERVAL
Button flashes every 2 seconds
Button flashes at 1/10 second
Button flashes every second
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6 Configuration Guide
6.1 Configuring HandyTone-386 WAN IP through Voice Prompt
6.1.1 DHCP Mode
Follow section 5.1 with voice menu option 01 to enable HandyTone-386 to use DHCP.
6.1.2 STATIC IP Mode
Follow section 5.1 with voice menu option 01 to enable HandyTone-386 to use STATIC IP mode, then
use option 02, 03, 04 to set up HandyTone-386’s IP, Subnet Mask, Gateway respectively.
6.1.3 TFTP Server Address
Follow section 5.1 with voice menu option 06 to configure the IP address of the TFTP server.
6.2 Configuring HandyTone-386 with Web Browser
HandyTone-386 series ATA has an embedded Web server that will respond to HTTP GET/POST
requests. It also has embedded HTML pages that allow users to configure the HandyTone-386 through
a Web browser such as Microsoft’s IE and AOL’s Netscape.
6.2.1 Access the Web Configuration Menu
First, get the IP address of the HandyTone-386 through section 5.1 with menu option 02. Then access
the HandyTone-386’s Web Configuration Menu using the following URI:
where the Phone-IP-Address is the IP address of the phone.
6.2.2 End User Configuration
Once this HTTP request is entered and sent from a Web browser, the HandyTone-386 will respond
with the following login screen:
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Grandstream Device Configuration
Password
Login
All Rights Reserved Grandstream Networks, Inc. 2005
The password is case sensitive and the factory default password for End User is “123”.
After a correct password is entered in the login screen, the embedded Web server inside the
HandyTone-386 will respond with the Configuration page which is explained in details below.
Grandstream Device Configuration
STATUS BASIC SETTINGS ADVANCED SETTINGS
End User Password:
(purposely not displayed for security protection)
IP Address:
dynamically assigned via DHCP (default) or PPPoE
(will attempt PPPoE if DHCP fails and following is non-blank)
PPPoE account ID:
PPPoE password:
0
0
0
0
Preferred DNS server:
.
.
.
statically configured as:
IP Address:
192
255
192
19
168
255
168
105
127
1
12
0
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
255
1
Subnet Mask:
Default Router:
DNS Server 1:
DNS Server 2:
1
5
1
24
198
4
Time Zone:
current setting is " GMT-6:00 (US Central Time, Chicago)"
Daylight Savings
Time:
No
Yes (if set to Yes, display time will be 1 hour ahead of normal time)
Cancel
Update
Reboot
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End User
Password
This contains the password to access the Web Configuration Menu. This
field is case sensitive.
IP Address
There are 2 modes under which the HandyTone ATA can operate:
- If DHCP mode is enabled, then all the field values for the Static IP mode
are not used (even though they are still saved in the Flash memory.) The
HandyTone ATA will acquire its IP address from the first DHCP server it
discovers from the LAN it is connected.
To use the PPPoE feature the PPPoE account settings need to be set. The
HandyTone will attempt to establish a PPPoE session if any of the PPPoE
fields is set.
- If Static IP mode is enabled, then the IP address, Subnet Mask, Default
Router IP address, DNS Server 1 (primary), DNS Server 2 (secondary) fields
will need to be configured. These fields are reset to zero by default.
Time Zone
This parameter controls how the displayed date/time will be adjusted
according to the specified time zone.
Daylight Savings Time This parameter controls whether the displayed time will be daylight savings
time or not. If set to Yes, then the displayed time will be 1 hour ahead of
normal time.
In addition to the Basic Settings configuration page, end users also have access to the device Status
page. The following is a snap shot of the device Status page. Details will be explained next.
Grandstream Device Configuration
STATUS BASIC SETTINGS ADVANCED SETTINGS
MAC Address: 00.0B.82.00.00.00
WAN IP Address:
Product Model: HT386
Software Version: Program-- 1.0.2.5 Bootloader-- 1.0.1.0 HTML-- 1.0.0.5 VOC-- 1.0.0.10
System Up Time: 0 day(s) 4 hour(s) 43 minute(s)
Registered: Yes
PPPoE Link Up: disabled
detected NAT type is symmetric NAT
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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
Detected NAT Type
This field shows what kind NAT the HandyTone is connected to via its
WAN port. It is based on STUN protocol.
6.2.3 Advanced User Configuration
To login to the Advanced User Configuration page, please follow the instructions in section 6.2.1 to
get to the following login page. The password is case sensitive and the factory default password for
Advanced User is “admin”.
Grandstream Device Configuration
Password
Login
All Rights Reserved Grandstream Networks, Inc. 2004
Advanced User configuration includes not only the end user configuration, but also advanced
configuration such as SIP configuration, Codec selection, NAT Traversal Setting and other
miscellaneous configuration. Following is a snap shot of the advanced configuration page.
Grandstream Device Configuration
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STATUS BASIC SETTINGS ADVANCED SETTINGS
Admin Password:
FXS Port 1:
(purposely not displayed for security protection)
sip.mycompany.com
SIP Server:
Outbound Proxy:
SIP User ID:
(e.g., sip.mycompany.com, or IP address)
(e.g., proxy.myprovider.com, or IP address, if any)
(the user part of an SIP address)
3123320
3123320
Authenticate ID:
(can be identical to or different from SIP User ID)
Authenticate
Password:
(purposely not displayed for security protection)
(optional, e.g., John Doe)
Name:
Register Expiration:
local SIP port:
60
(in minutes. default 1 hour, max 45 days)
(default 5060)
(1024-65535, default 5004)
Yes (if Yes, Call Forwarding & Call-Waiting-Disable are supported
5060
5004
local RTP port:
Enable Call Features:
No
locally)
Send DTMF:
in-audio
via RTP (RFC2833)
via SIP INFO
101
DTMF Payload Type:
FXS Port 2:
sip.mycompany.com
SIP Server:
(e.g., sip.mycompany.com, or IP address)
(e.g., proxy.myprovider.com, or IP address, if any)
(the user part of an SIP address)
Outbound Proxy:
SIP User ID:
3125250
3125250
Authenticate ID:
(can be identical to or different from SIP User ID)
Authenticate
Password:
(purposely not displayed for security protection)
(optional, e.g., John Doe)
Name:
Register Expiration:
local SIP port:
60
(in minutes. default 1 hour, max 45 days)
(default 5062)
(1024-65535, default 5008)
Yes (if Yes, Call Forwarding & Call-Waiting-Disable are supported
5062
5008
local RTP port:
Enable Call Features:
No
locally)
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Send DTMF:
in-audio
via RTP (RFC2833)
via SIP INFO
101
DTMF Payload Type:
General Misc. Settings:
current setting is " PCMU"
Preferred Vocoder:
(in listed order)
choice 1:
current setting is " PCMA"
current setting is " G723"
current setting is " G729"
current setting is " G726-32"
current setting is " iLBC"
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 5.3kbps encoding rate
20ms 30ms
iLBC frame size:
iLBC payload type:
Silence Suppression:
SIP Registration:
97
(between 96 and 127, default is 97)
No
Yes
No
Yes
Yes
Unregister On Reboot:
Fax Mode:
No
T.38 (Auto Detect)
Pass-Through
48
Layer 3 QoS:
Layer 2 QoS:
Use DNS SRV:
(Diff-Serv or Precedence value)
0
0
802.1Q/VLAN Tag
802.1p priority value
(0-7)
No
No
No
Yes
Yes
User ID is phone
number:
Early Dial:
Dial Plan Prefix:
Yes (use "Yes" only if proxy supports 484 response)
(this prefix string is added to each dialed number)
(in seconds, default is 4 seconds)
4
No Key Entry Timeout:
Use # as Dial Key:
Use random port:
No
No
No
Yes (if set to Yes, "#" will function as the "(Re-)Dial" key)
Yes
NAT Traversal:
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stun.mycompany.com
Yes, STUN server is:
(URI or IP:port)
20
keep-alive interval:
Use NAT IP
(in seconds, default 20 seconds)
(if specified, this IP address is used in SIP/SDP message)
(if specified, the content will appear in Proxy-Require
Proxy-Require:
header)
192
168
1
30
TFTP Upgrade Server:
SUBSCRIBE for MWI:
.
.
.
(for remote software upgrade and configuration)
No, do not send SUBSCRIBE for Message Waiting Indication
Yes, send periodical SUBSCRIBE for Message Waiting Indication
Offhook Auto-Dial:
(User ID/extension to dial automatically when offhook)
Yes (Flash will be sent as a DTMF event if set to Yes)
Send Flash Event:
FXS Impedance:
No
current setting is " 600 Ohm (North America)"
current setting is " Bellcore"
Caller ID Scheme:
Onhook Voltage (FXS
Port):
current setting is " 48V"
Disable Call-Waiting:
No
Yes
Polarity Reversal:
NTP Server:
No
Yes (reverse polarity upon call establishment and termination)
(URI or IP address)
time.nist.gov
Send Anonymous:
Lock keypad update:
PSTN access code:
Syslog Server:
No
No
Yes (caller ID will be blocked if set to Yes)
Yes (configuration update via keypad is disabled if set to Yes)
(key pattern to use the PSTN line, default is "*00")
current setting is "NONE"
Cancel
Syslog Level:
Update
Reboot
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Admin Password
This contains the password to access the Advanced Web Configuration page.
This field is case sensitive.
SIP Server
SIP Server’s URI or IP address
Outbound Proxy
SIP Outbound Proxy Server’s URI or IP address
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SIP User ID
SIP service subscriber’s User ID
Authenticate ID
SIP service subscriber’s Authenticate ID. Can be identical to or different from
SIP User ID
Authenticate
Password
SIP service subscriber’s account password
Name
SIP service subscriber’s name which will be used for Caller ID display
Register Expiration This parameter allows the user to specify the time frequency (in minutes) the
HandyTone ATA refreshes its registration with the specified registrar. The
default interval is 60 minutes (or 1 hour). The maximum interval is 65535
minutes (about 45 days).
Local SIP port
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.
Enable Call
Features
Default is No. If set to Yes, Call Forwarding & Do-Not-Disturb are supported
locally
Send DTMF
This parameter controls how DTMF events are transmitted. There are 3 ways:
in audio which means DTMF is combined in audio signal (not very reliable
with low-bit-rate codec), via RTP (RFC2833), or via SIP INFO.
DTMF Payload
Type
This parameter sets the payload type for DTMF using RFC2833
Preferred Vocoder The HandyTone ATA supports up to 7 different Vocoder types including G.711
A-/U-law，G.723.1, G.726, G.728, G.729A/B, iLBC. Depending on the
product model, some of these Vocoders may not be provided in standard
release.
Users can configure Vocoders in a preference list that will be included with the
same preference order in SDP message. The first Vocoder in this list can be
entered by choosing the appropriate option in “Choice 1”. Similarly, the last
Vocoder in this list can be entered by choosing the appropriate option in
“Choice 7”.
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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.
G723 Rate:
This defines the encoding rate for G723 vocoder. By default, 6.3kbps rate is
chosen.
iLBC frame size:
This sets the iLBC size in 20ms or 30ms
iLBC payload type: This defines payload time for iLBC. Default value is 98. The valid range is
between 96 and 127.
Silence Suppression This controls the silence suppression/VAD feature of G723 and G729. If set to
“Yes”, when a silence is detected, small quantity of VAD packets (instead of
audio packets) will be sent during the period of no talking. If set to “No”, this
feature is disabled.
SIP Registration
This parameter controls whether the HandyTone ATA needs to send
REGISTER messages to the proxy server. The default setting is “Yes”.
Unregister on
Reboot
Default is No. If set to yes, the SIP user will be unregistered on reboot.
Fax Mode
T.38 (Auto Detect) FoIP by default, or Pass-Through (must use codec
PCMU/PCMA)
Layer 3 QoS
Layer 2 QoS
Use DNS SRV:
This field defines the layer 3 QoS parameter which can be the value used for IP
Precedence or Diff-Serv or MPLS. Default value is 48.
This contains the value used for layer 2 VLAN tag. Default setting is
blank.
Default is No. If set to Yes the client will use DNS SRV for server lookup
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User ID is Phone
Number
If the HandyTone ATA has an assigned PSTN telephone number, this field
should be set to “Yes”. Otherwise, set it to “No”. If “Yes” is set, a
“user=phone” parameter will be attached to the “From” header in SIP request
Early Dial
Default is No. Use only if proxy supports 484 response
Sets the prefix added to each dialed number
Default is 4 seconds.
Dial Plan Prefix
No Key Entry
Timeout
Use # as
Send Key
This parameter allows users to configure the “#” key to be used as the “Send”
(or “Dial”) key. If set to “Yes”, pressing this key will immediately trigger the
sending of dialed string collected so far. In this case, this key is essentially
equivalent to the “(Re)Dial” key. If set to “No”, this “#” key will then be
included as part of the dial string to be sent out.
Use Random Port
NAT Traversal
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.
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.
Keep-alive interval This parameter specifies how often the HandyTone ATA sends a blank UDP
packet to the SIP server in order to keep the “hole” on the NAT open.
Use NAT IP:
NAT IP address used in SIP/SDP message. Default is blank.
Proxy-Require
SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.
TFTP Upgrade
Server
This is the IP address of the configured TFTP server. If it is non-zero or not
blank, the HandyTone ATA will attempt to retrieve new configuration file or
new code image from the specified TFTP server at boot time. It will make up to
3 attempts before timeout and then it will start the boot process using the
existing code image in the Flash memory. If a TFTP server is configured and a
new code image is retrieved, the new downloaded image will be verified and
then saved into the Flash memory.
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Subscribe for MWI: Default is No. When set to “Yes” a SUBSCRIBE for Message Waiting
Indication will be sent periodically.
Offhook
Auto-Dial
This parameter allows users to configure a User ID or extension number to be
automatically dialed upon offhook. Please note that only the user part of a SIP
address needs to be entered here. The HandyTone ATA will automatically
append the “@” and the host portion of the corresponding SIP address.
Send Flash Event
This parameter allows users to control whether to send an SIP NOTIFY
message indicating the Flash event, or just to switch to the voice channel when
users press the Flash key.
FXS Impedance
Selects the impedance of the analog telephone connected to the Phone port.
Caller ID Scheme
• Bellcore (North America)
• ETSI-FSK (France, Germany, Norway, Taiwan, UK-CCA)
• ETSI-DTMF (Finland, Sweden)
• DTMF (Denmark)
Onhook Voltage
Select the onhook voltage to suit different area or PBX.
Default is No.
Disable Call
Waiting
Polarity Reversal
Default is No. If set to Yes, polarity will be reversed upon call establishment
and termination.
NTP server
This parameter defines the URI or IP address of the NTP server which is used
by the HandyTone ATA to display the current date/time.
Send Anonymous
If this parameter is set to “Yes”, the “From” header in outgoing INVITE
message will be set to anonymous, essentially blocking the Caller ID from
displaying.
Lock keypad
update
If this parameter is set to “Yes”, the configuration update via keypad is
disabled.
Syslog Server
The IP address or URL of System log server. This feature is especially useful
for ITSP (Internet Telephone Service Provider)
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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
PSTN Access Code Default is “*00”. By pressing the code, user can switch the phone connected to
ATA to PSTN and call out using PSTN. User can change this access code in
the configuration. This is very useful for “911” calls when emergency happens.
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-386 will then display the following screen to confirm that the changes have been saved.
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Grandstream Device Configuration
STATUS BASIC SETTINGS ADVANCED SETTINGS
Your configuration changes have been saved.
They will take effect on next reboot.
All Rights Reserved Grandstream Networks, Inc. 2004
Users are recommended to power cycle the HandyTone-386 after seeing the above message.
6.2.5 Rebooting the HandyTone-386 from Remote
The administrator of the HandyTone-386 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
All Rights Reserved Grandstream Networks, Inc. 2004
At this point, the user can relogin to the HandyTone-386 after waiting for about 30 seconds.
6.3 Configuration through a Central Server
Grandstream HandyTone ATAs can be automatically configured from a central provisioning system.
When HandyTone ATA boot up, it will send TFTP or HTTP request to download configuration files,
there are two configuration files, one is “cfg.txt” and the other is “cfg000b82xxxxxx”, where
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“000b82xxxxxx” is the MAC address of the HandyTone ATA. For more information regarding
configuration file format, please refer to the related Grandstream documentation.
The configuration file can be downloaded via TFTP or HTTP from the central server. A service
provider or an enterprise with large deployment of HandyTone ATAs can easily manage the
configuration and service provisioning of individual devices remotely and automatically from a central
server. GAPS (Grandstream Automated Provisioning System) uses enhanced (NAT friendly) TFTP or
HTTP (thus no NAT issues) and other communication protocols to communicate with each individual
HandyTone ATA for firmware upgrade, remote reboot, etc.
Grandstream provides a licensed provisioning system called GAPS that can be used to support
automated configuration of HandyTone ATA. To enable this feature on the HandyTone ATA, a user
just needs to enter the IP address of the GAPS server in the TFTP server field of the configuration
screen, or enter the HTTP provisioning Server URL in the HTTP Upgrade Server field. Then reboot
the HandyTone ATA.
For details on how GAPS works, please refer to the documentation of GAPS product.
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7 Software Upgrade
7.1 Upgrade through HTTP (pending)
To upgrade software, HandyTone-386 can be configured with an HTTP server where the new code
image file is located. For example, following URL in the HTTP Upgrade Server:
Where firmware.mycompany.com is the FQDN of the HTTP server, “:6688” is the TCP port the HTTP
server listening to, “/Grandstream/1.0.0.4” is the RELATIVE directory to the root directory in HTTP
server. Thus, you can put different firmware into different directory as well.
Note: For HTTP firmware upgrade, if “Auto Upgrade” field is set to “No”, HTTP firmware upgrade
will be performed once. If “Auto Upgrade” is set to “Yes”, the ATA will check the HTTP server in
the number of days that is defined in “Check for new firmware every” field.
7.2 Upgrade through TFTP
To upgrade software, HandyTone-386 can be configured with a TFTP server where the new code
image is located. The TFTP upgrade can work in either static IP or DHCP mode using private or public
IP address. It is recommended to set the TFTP server address in either a public IP address or on the
same LAN with the HandyTone-386.
There are two ways to set up the TFTP server to upgrade the firmware, namely through voice menu
prompt or via the HandyTone-386’s Web configuration interface. To configure the TFTP server via
voice prompt, follow section 5.1 with option 06, once set up the TFTP IP address, power cycle the
ATA, the firmware will be fetched once the ATA boots up.
To configure the TFTP server via the Web configuration interface, open up your browser to point at
the IP address of the HandyTone-386. Input the admin password to enter the configuration screen.
From there, enter the TFTP server address in the designated field towards the bottom of the
configuration screen.
Once the TFTP server is configured, please power cycle the HandyTone-386.
TFTP checking is only performed during the initial power up. If the configured TFTP server is found
and a new code image is available, the HandyTone-386 will attempt to retrieve the new image files by
downloading them into the HandyTone-386’s SRAM. During this stage, the HandyTone-386’s LED
will blink until the checking/downloading process is completed. Upon verification of checksum, the
new code image will be saved into the Flash. If TFTP fails for any reason (e.g., TFTP server is not
responding, there are no code image files available for upgrade, or checksum test fails, etc), the
HandyTone-386will stop the TFTP process and simply boot using the existing code image in the flash.
TFTP process may take as long as 1 to 20 minutes over the Internet, or just 20+ seconds if it is
performed on a LAN. Users are recommended to conduct TFTP upgrade in a controlled LAN
environment if possible. For those who do not have a local TFTP server, Grandstream provides a
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NAT-friendly TFTP server on the public Internet for firmware upgrade. Please check the Service
section of Grandstream’s Web site to obtain this TFTP server’s IP address.
NOTES:
• When HandyTone ATA boot up, it will send TFTP or HTTP request to download configuration
files, there are two configuration files, one is “cfg.txt” and the other is “cfg000b82xxxxxx”,
where “000b82xxxxxx” is the MAC address of the HandyTone ATA. These two files are for
initial automatically provisioning purpose only, for normal TFTP or HTTP firmware upgrade,
the following error messages in a TFTP or HTTP server log can be ignored.
TFTP Error from [IP ADRESS] requesting cfg000b82023dd4 : File does not exist
TFTP Error from [IP ADRESS] requesting cfg.txt : File does not exist
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8 Restore Factory Default Setting
Warning: Restore the Factory Default Setting will delete all configuration information of the
device.
Please disconnect network cable and power cycle the unit before trying to reset the unit to factory
default. The steps are as follows:
A. Find the MAC address of the device. It is a 12 digits number located on the bottom of the unit.
B. Encode the MAC address. Please use the following mapping:
0-9: 0-9
A: 22
B: 222
C: 2222
D: 33
E: 333
F: 3333
For example, 000B82004D2A will be encoded to 0002228200433222.
C. To perform factory reset:
1. Press “***” or the LED button for voice prompt.
2. Enter “99” and get the voice prompt “Reset”.
3. Enter the encoded MAC address of the device. A "click" sound will be heard.
4. Wait for 15 seconds.
The device is now restored to the factory default setting.
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