Grandstream Networks Grandstream HandyTone HandyTone 488 User Manual

User Manual  
HandyTone-488  
Analog Telephone Adaptor  
For SW Release Version 1.0.0.4  
Grandstream Networks, Inc.  
HandyTone-488 User Manual  
Grandstream Networks, Inc.  
1 Welcome  
Congratulations on becoming an owner of HandyTone-488. You made an excellent choice  
and we hope you enjoy all of its capabilities.  
Grandstream's HandyTone-488 is an all-in-one VoIP integrated access device that features  
superb audio quality, rich functionalities, high level of integration, compactness and ultra-  
affordability. The HandyTone-488 is fully compatible with SIP industry standard and can  
interoperate with many other SIP compliant devices and software on the market.  
Grandstream HandyTone-488 is a new addition to the popular HandyTone product family. It  
is an enhanced model compared to the award-winning HandyTone-486 in that it allows call  
origination and termination from/to the PSTN network (via FXO port) remotely and automated  
emergency call routing through PSTN network.  
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2 Installation  
HandyTone-488 Analog Telephone Adaptor is an all-in-one VoIP integrated device designed  
to be a total solution for networks providing VoIP services.  
The HandyTone-488 VoIP functionalities are available via a regular analog telephone.  
The following photo illustrates the appearance of a HandyTone-488.  
Top View  
Side Views  
RJ45  
10M Ethernet  
LAN - WAN  
RJ11  
FXS Port  
(Phone)  
+5V/1200mA  
RJ11  
FXO Port  
(Phone Line)  
BUTTON  
RED LED  
GREEN LED  
Interconnection Diagram of the HandyTone-488:  
Internet ADSL/Cable  
Modem Ethernet  
Analog Phone  
WAN  
FXS  
FXO  
PSTN  
Cordless Phone  
LAN  
PC  
PC  
PC  
Fax  
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HandyTone-488 has one FXS port and one FXO port. The RJ11 jack next to the LAN port  
is a FXS port. The RJ11 jack on the side of the HandyTone-488 is a FXO port. Both the FXS  
port and the FXO port can have a separate SIP account. This is a key feature of HandyTone-  
488. Telephone calls can be originated or terminated from/to the PSTN network via FXO port.  
Following are the steps to install a HandyTone-488:  
1. Connect a standard touch-tone analog telephone to the FXS port.  
2. Insert a standard RJ11 telephone cable into the FXO port and connect the other end of  
the telephone cable to a wall jack.  
3. Insert the Ethernet cable into the WAN port of HandyTone-488 and connect the other  
end of the Ethernet cable to an uplink port (a router or a modem, etc.)  
4. Connect a PC to the LAN port of HandyTone-488.  
5. Insert the power adapter into the HandyTone-488 and connect it to a wall outlet.  
Please follow the instructions in section 6.2.1 to configure the HandyTone-488.  
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3 What is Included in the Package  
The HandyTone-488 package contains:  
1) One HandyTone-488  
2) One universal power adaptor  
3) One Ethernet cable  
3.1 Safety Compliances  
The HandyTone-488 is compliant with various safety standards including FCC/CE and C-tick.  
Its power adaptor is compliant with UL standard. The HandyTone-488 should only operate  
with the universal power adaptor provided in the package.  
Warning: Please do not attempt to use a different power adaptor. Using other power  
adaptor may damage the HandyTone-488 and will void the manufacturer warranty.  
Caution: Changes or modifications to this product not expressly approved by  
Grandstream, or operation of this product in any way other than as detailed by this  
User Manual, could void your manufacturer warranty.  
Information in this document is subject to change without notice. No part of this document may be  
reproduced or transmitted in any form or by any means, electronic or mechanical, for any purpose without the  
express written permission of Grandstream Networks, Inc..  
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4 Product Overview  
4.1 Key Features  
Supports SIP 2.0(RFC 3261), TCP/UDP/IP, RTP/RTCP, HTTP, ICMP, ARP/RARP,  
DNS, DHCP (both client and server), NTP, PPPoE, STUN, TFTP, etc.  
Built-in router, NAT, Gateway and DMZ port forwarding  
Supports call origination and termination from/to the PSTN network (via FXO port)  
Powerful digital signal processing (DSP) to ensure superb audio quality; advanced  
adaptive jitter control and packet loss concealment technology  
Support various vocoders including G.711 (a-law and u-law), G.723.1 (5.3K/6.3K),  
G.726 (40K/32K/24K/16K), as well as G.728, G.729A/B, and iLBC.  
Support Caller ID/Name display or block, Hold, Call Waiting/Flash, Call Transfer, Call  
Forward, in-band and out-of-band DTMF, Dial Plans, etc.  
Support fax pass through  
Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise  
Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control)  
Support standard encryption and authentication (DIGEST using MD5 and MD5-sess)  
Support for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS)  
Support automated NAT traversal without manual manipulation of firewall/NAT  
Support device configuration via built-in IVR, Web browser or central configuration file  
through TFTP or HTTP  
Support firmware upgrade via TFTP or HTTP with encrypted configuration files.  
Ultra compact (wallet size) and lightweight design, great companion for travelers  
Compact, lightweight Universal Power adapter.  
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4.2 Hardware Specification  
The table below describes the difference among these models.  
Model  
HandyTone-488  
LAN interface  
WAN interface  
FXS telephone port  
FXO port  
1xRJ45 10Base-T  
1xRJ45 10Base-T  
1xFXS  
1xFXO  
Button  
1
LED  
Green and red color  
Universal Switching  
Power Adaptor  
Input: 100-240VAC 50-60 Hz  
Output: +5VDC, 1200mA,  
UL certified  
Dimension  
70mm (W)  
130mm (D)  
27mm (H)  
Weight  
0.6lbs (0.3kg)  
Temperature  
40 - 130oF  
5 – 45oC  
Humidity  
10% - 90%  
(non-condensing)  
Compliance  
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5 Basic Operations  
5.1 Get Familiar with Voice Prompt  
HandyTone-488 has stored a voice prompt menu for quick browsing and simple  
configuration.  
To enter this voice prompt menu, simply press the button or “***” from the analog phone.  
Menu  
Voice Prompt  
User’s Options  
Main Menu “Enter a Menu Option”  
Enter “*” for the next menu option  
Enter “#” to return to the main menu  
Enter 01-06, 47, 86, 99 menu option  
Enter “9” to toggle the selection  
01  
02  
“DHCP Mode”,  
“Static IP Mode”  
“IP Address “ + IP address  
The current WAN IP address is  
announced  
Enter 12 digit new IP address if in Static IP  
Mode  
03  
04  
05  
06  
47  
“Subnet “ + IP address  
“Gateway “ + IP address  
“DNS Server “ + IP address  
“TFTP Server “ + IP address Same as menu 02  
“Direct IP Calling”  
Same as menu 02  
Same as menu 02  
Same as menu 02  
When entered, you will be prompted a dial  
tone, then enter 12 digit IP address  
This menu can also be entered by  
pressing the button again  
86  
99  
“Voice Messages Pending”  
“No Voice Messages”  
“RESET”  
Enter “9” to dial pre-configured phone  
number to retrieve VM  
Enter “9” to reboot the phone  
Enter encoded MAC address to restore  
factory default setting  
“Invalid Entry”  
Automatically returns to main menu  
Notes:  
Once the button is pressed, it enters the voice prompt main menu. If the button is  
pressed again, while it is already in the voice prompt menu, it jumps to “Direct IP Call”  
option and a dial tone is prompted  
“*” always shifts down to the next menu option  
“#” always returns to the main menu  
“9” functions as the ENTER key in many cases to confirm an option  
All entered digit sequences have known lengths - 2 digits for menu option and 12 digits  
for IP address. Once all of the digits are collected, the input will be processed.  
Key entry can not be deleted but the phone may prompt error once it is detected  
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5.2 Make Phone Calls  
5.2.1 Calling phone or extension numbers  
To make a phone or extension number call:  
a) Dial the number directly and wait for 4 seconds (default “No Key Entry Timeout”) .  
Or  
b) Dial the number directly, and press # (assuming that “Use # as dial key” is selected  
in web configuration).  
Other functions available during the call are call-waiting/flash, call-transfer, and call-forward.  
5.2.2 Direct IP calls  
Direct IP calling allows two phones, that is, a HandyTone with an analog phone and another  
VoIP Device, to talk to each other in an ad hoc fashion without a SIP proxy. VoIP calls can be  
made between two phones if:  
Both HandyTone ATA and the other VoIP device (i.e., another HandyTone ATA or  
other SIP products) have public IP addresses, or  
Both HandyTone ATA and the other VoIP device (i.e., another HandyTone ATA or  
other SIP produces) are on the same LAN using private or public IP addresses, or  
Both HandyTone ATA and the other VoIP device (i.e., another HandyTone ATA or  
other SIP products) can be connected through a router using public or private IP  
addresses.  
To make a direct IP call, first pick up the analog phone or turn on the speakerphone on the  
analog phone, then follow Section 5.1 with voice prompt 47, followed by the 12-digit target IP  
address. Destination ports can be specified by using “*4” (encoding for “:”) followed by the  
encoded port number. Following is a table of the encoding scheme for the most commonly  
used characters:  
Input  
00  
01  
02  
03  
04  
05  
06  
07  
08  
09  
*0  
Meaning  
0
1
2
3
4
5
6
7
8
9
. (dot character)  
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*4  
: (column character)  
Examples:  
If the target IP address is 192.168.0.160, the dialing convention is  
Voice Prompt with option 47, then 192168000160  
followed by pressing the “#” key if it is configured as a send key or wait 4 seconds. In this  
case, the default destination port 5060 is used if no port is specified.  
If the target IP address/port is 192.168.1.20:5062, then the dialing convention would be:  
Voice Prompt with option 47, then 192168001020*45062 followed by pressing the “#” key if  
it is configured as a send key or wait for 4 seconds.  
5.2.3 Blind Transfer  
Assuming that call party A and B are in conversation. A wants to Blind Transfer B to C:  
1. A presses FLASH (on the analog phone, or Hook Flash for old model phones) to get a  
dial tone.  
2. Then A dials *87 then dials C’s number, and then #(or wait for 4 seconds)  
3. A can hang up.  
Note: Call Feature has to be set to YES.  
A can hold on to the phone and wait for one of the three following behaviors:  
A quick confirmation tone (temporarily using the call waiting indication tone) followed  
by a dial tone. This indicates the transfer is successful (transferee has received a 200  
OK from transfer target). At this point, A can either hang up or make another call.  
A quick busy tone followed by a restored call (on supported platforms only). This  
means the transferee has received a 4xx response for the INVITE and we will try to  
recover the call. The busy tone is just to indicate to the transferor that the transfer has  
failed.  
Busy tone keeps playing. This means we have failed to receive the second NOTIFY  
from the transferee and decided to time out. Note: this does not indicate the transfer  
has been successful, nor does it indicate the transfer has failed. When transferee is a  
client that does not support the second NOTIFY (such as our own earlier firmware),  
this will be the case. In bad network scenarios, this could also happen, although the  
transfer may have been completed successfully.  
5.2.4 Attended Transfer  
Assuming that call party A and B are in conversation. A wants to Attend Transfer B to C:  
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1. A presses FLASH (on the analog phone, or Hook Flash for old model phones) to get a  
dial tone  
2. A then dial C’s number then # (or wait for 4 seconds). A and C now in conversation.  
3. A can hang up.  
Note: When intended Transfer failed, if A hangs up, the HandyTone-488 will ring user A  
again to remind A that B is still on the call, by pressing FLASH or Hook again will restore the  
conversation between A and B.  
5.2.5 Send and Receive PSTN Calls  
Users can send and receive calls from PSTN. To receive PSTN calls, simply take the phone  
off hook when the analog phone rings. To make a PSTN call, first press *00 (or your own  
PSTN Access Code) to get the PSTN line dial tone and dial the PSTN number.  
5.2.6 VoIP-to-PSTN Calls  
To make a VoIP-to-PSTN call, users need to dial the FXO SIP account phone number first. A  
ring tone is played once followed by a dial tone. At this time, users can dial a PSTN  
telephone number or a mobile telephone number then # (or wait for 4 seconds). The call will  
be established afterwards. If no PSTN number is entered after the dial tone, HandyTone-488  
will hang up automatically in 10 seconds.  
5.2.7 PSTN-to-VoIP Calls  
To make a PSTN-to-VoIP call, PSTN callers need to originate a call to the FXO port  
telephone number first. If no one answers the FXS phone after 4 ring tones, a dial tone is  
played. At this time, users can dial a VoIP telephone number then # (or wait for 4 seconds).  
The call will be established afterwards. If no VoIP number is entered after the dial tone,  
HandyTone-488 will hang up automatically in 10 seconds.  
5.3 Call Features  
5.3.1 Call Features Table  
Following table shows the call features of HandyTone-488.  
Key  
*30  
*31  
*67  
*82  
Call Features  
Block Caller ID (for all subsequent calls)  
Send Caller ID (for all subsequent calls)  
Block Caller ID (per call)  
Send Caller ID (per call)  
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*50  
*51  
*70  
*71  
*72  
Disable Call Waiting (for all subsequent calls)  
Enable Call Waiting (for all subsequent calls)  
Disable Call Waiting. (Per Call)  
Enable Call Waiting (Per Call)  
Unconditional Call Forward.  
To use this feature, dial “*72” and get the dial tone. Then  
dial the forward number and “#” for a dial tone, then hang  
up.  
*73  
*90  
Cancel Unconditional Call Forward  
To cancel “Unconditional Call Forward”, dial “*73” and get  
the dial tone, then hang up.  
Busy Call Forward  
To use this feature, dial “*90” and get the dial tone. Then  
dial the forward number and “#” for a dial tone, then hang  
up.  
*91  
*92  
Cancel Busy Call Forward  
To cancel “Busy Call Forward”, dial “*91” and get the dial  
tone, then hang up  
Delayed Call Forward  
To use this feature, dial “*92” and get the dial tone. Then  
dial the forward number and “#” for a dial tone, then hang  
up.  
*93  
Cancel Delayed Call Forward  
To cancel this Forward, dial “*93” and get the dial tone,  
then hang up  
When in conversation, this action will switch to the new  
incoming call if there is a call waiting indication.  
When in conversation without an incoming call, this action  
will switch to a new channel for a new call.  
Flash/Hook  
5.3.2 PSTN Pass Through  
When HandyTone-488 is out of power, the RJ 11 line jack on the HandyTone-488 side will  
function as a pass through jack. Users will be able to use the same analog phone for PSTN  
calls.  
5.4 LED Light Pattern Indication  
Following tables show the LED light pattern indication.  
RED LED always indicates abnormal status  
DHCP Failed or WAN No Cable  
HandyTone-488 fails to register  
Button flashes every 2 seconds (if DHCP is configured)  
Button flashes every 2 seconds (if SIP server is  
configured)  
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GREEN LED always indicates normal working status  
Message Waiting Indication  
RINGING  
RINGING INTERVAL  
Button flashes every 2 seconds  
Button flashes at 1/10 second  
Button flashes every second  
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6 Configuration Guide  
6.1 Configuring HandyTone-488 WAN IP through Voice Prompt  
6.1.1 DHCP Mode  
Follow section 5.1 with voice menu option 01 to enable HandyTone-488 to use DHCP.  
6.1.2 STATIC IP Mode  
Follow section 5.1 with voice menu option 01 to enable HandyTone-488 to use STATIC IP  
mode, then use option 02, 03, 04 to set up HandyTone-488’s IP, Subnet Mask, Gateway  
respectively.  
6.2 Configuring HandyTone-488 with Web Browser  
HandyTone-488 series ATA has an embedded Web server that will respond to HTTP  
GET/POST requests. It also has embedded HTML pages that allow users to configure the  
HandyTone-488 through a Web browser such as Microsoft’s IE and AOL’s Netscape.  
6.2.1 Access the Web Configuration Menu  
The HandyTone-488 HTML configuration menu can be accessed via LAN or WAN port:  
From the LAN port use the default LAN gateway IP address:  
The WAN port HTML configuration option is disabled by default from factory. To  
access the HTML configuration menu from the WAN port, first enable the “WAN side  
HTTP access” option by accessing the configuration via LAN port. With the WAN side  
HTTP access enabled, then get the WAN IP address of the HandyTone-488 through  
section 5.1 with menu option 02. The HandyTone-488’s Web Configuration page can  
be accessed by the following URI via WAN port:  
where the HandyTone-IP-Address is the WAN IP address of the HandyTone-488.  
6.2.2 End User Configuration  
Once this HTTP request is entered and sent from a Web browser, the HandyTone-488 will  
respond with the following login screen:  
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Grandstream Device Configuration  
Password  
Login  
All Rights Reserved Grandstream Networks, Inc. 2004  
The password is case sensitive and the factory default password for End User is ‘123’.  
After a correct password is entered in the login screen, the embedded Web server inside the  
HandyTone-488 will respond with the Configuration page which is explained in details below.  
Grandstream Device Configuration  
STATUS BASIC SETTINGS ADVANCED SETTINGS  
End User  
Password:  
(purposely not displayed for security protection)  
IP Address:  
dynamically assigned via DHCP (default) or  
PPPoE  
(will attempt PPPoE if DHCP fails and following is  
non-blank)  
PPPoE account ID:  
PPPoE password:  
0
0
0
0
Preferred DNS server:  
.
.
.
statically configured as:  
192  
255  
192  
129  
204  
168  
1
12  
0
IP Address:  
.
.
.
.
.
.
.
.
.
.
.
.
.
.
.
255  
168  
105  
127  
255  
1
Subnet Mask:  
Default Router:  
DNS Server 1:  
DNS Server 2:  
1
5
1
198  
4
current setting is " GMT-6:00 (US Central Time, Chicago)"  
Time Zone:  
Daylight Savings  
Time:  
No  
Yes (if set to Yes, display time will be 1 hour ahead of normal  
time)  
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NAT/DHCP Server Information & Configuration:  
Cloned WAN MAC  
Addr:  
(in hex format)  
(default is 255.255.255.0)  
(base IP for the LAN port, default is 192.168.2.1)  
255.255.255.0  
LAN Subnet Mask:  
LAN DHCP Base  
IP:  
192.168.2.1  
DHCP IP Lease  
Time:  
120  
(in units of hours, default is 120 hours or 5 days)  
DMZ IP:  
0
0
0
0
0
0
0
0
0
0
0
0
0
0
" UDPOnly"  
" UDPOnly"  
" UDPOnly"  
" UDPOnly"  
" UDPOnly"  
" UDPOnly"  
" UDPOnly"  
" UDPOnly"  
WAN port  
WAN port  
WAN port  
WAN port  
WAN port  
WAN port  
WAN port  
WAN port  
LAN IP  
LAN IP  
LAN IP  
LAN IP  
LAN IP  
LAN IP  
LAN IP  
LAN IP  
LAN port  
LAN port  
LAN port  
LAN port  
LAN port  
LAN port  
LAN port  
LAN port  
Protocol  
Protocol  
Protocol  
Protocol  
Protocol  
Protocol  
Protocol  
Protocol  
Port Forwarding:  
0
0
Update  
All Rights Reserved Grandstream Networks, Inc. 2004  
End User  
Password  
This contains the password to access the Web Configuration Menu.  
This field is case sensitive.  
IP Address  
There are 2 modes under which the HandyTone ATA can operate:  
- If DHCP mode is enabled, then all the field values for the Static IP  
mode are not used (even though they are still saved in the Flash  
memory.) The HandyTone ATA will acquire its IP address from the  
first DHCP server it discovers from the LAN it is connected.  
To use the PPPoE feature the PPPoE account settings need to be  
set. The HandyTone will attempt to establish a PPPoE session if any  
of the PPPoE fields is set.  
- If Static IP mode is enabled, then the IP address, Subnet Mask,  
Default Router IP address, DNS Server 1 (primary), DNS Server 2  
(secondary) fields will need to be configured. These fields are reset to  
zero by default.  
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Time Zone  
This parameter controls how the displayed date/time will be adjusted  
according to the specified time zone.  
Daylight Savings  
Time  
This parameter controls whether the displayed time will be daylight  
savings time or not. If set to Yes, then the displayed time will be 1  
hour ahead of normal time.  
Cloned WAN MAC  
Address:  
Allow the user to set a specific MAC address. Set in Hex format  
LAN Subnet Mask  
Sets the LAN subnet mask. Default value is 255.255.255.0  
LAN DHCP Base IP: Base IP for the LAN port which functions as a Gateway for the  
subnet. Default value is 192.168.2.1  
DHCP IP Lease  
Time:  
Value is set in units of hours. Default value is 120hr (5 Days.) The  
time IP address are assigned to the LAN clients  
DMZ IP:  
Forward all WAN IP traffic to a specific IP address if no matching port  
is used by HandyTone-488 itself or in the defined port forwarding.  
Port Forwarding:  
Allow users to forward a matching (TCP/UDP) port to a specific LAN  
IP address with a specific (TCP/UDP) port.  
In addition to the Basic Settings configuration page, end users also have access to the device  
Status page. The following is a snap shot of the device Status page. Details will be explained  
next.  
Grandstream Device Configuration  
STATUS BASIC SETTINGS ADVANCED SETTINGS  
MAC Address: 00.0B.82.02.B2.95  
WAN IP Address: 192.168.1.12  
Product Model: HT488  
Software Version: Program-- 1.0.0.3 Bootloader-- 1.0.2.22 HTML-- 1.0.0.2 VOC-- 1.0.0.7  
System Up Time: 0 day(s) 3 hour(s) 23 minute(s)  
Registered: Yes  
PPPoE Link Up: disabled  
detected NAT type is full cone  
All Rights Reserved Grandstream Networks, Inc. 2004  
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MAC Address  
The device ID, in HEX format. This is a very important ID for ISP  
troubleshooting.  
WAN IP Address  
Product Model  
This field shows WAN port IP address.  
This field contains the product model info.  
Software Version  
Program: This is the main software release, its number is always  
used for firmware upgrade.  
Bootloader: This is normally not changed.  
HTML: This is the user interface, normally not changed.  
VOC: This is the codec program, normally not changed.  
System Up Time  
This field indicates how long the device has been up since the last  
reboot.  
Registered  
This field indicates whether the device is registered to the SIP server.  
This field shows whether the PPPoE connection is enabled or not.  
PPPoE Link Up  
Detected NAT Type  
This field shows what kind NAT the HandyTone is connected to via its  
WAN port. It is based on STUN protocol.  
6.2.3 Advanced User Configuration  
To login to the Advanced User Configuration page, please follow the instructions in section  
6.2.1 to get to the following login page. The password is case sensitive and the factory  
default password for Advanced User is “admin”.  
Grandstream Device Configuration  
Password  
Login  
All Rights Reserved Grandstream Networks, Inc. 2004  
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Advanced User configuration includes not only the end user configuration, but also advanced  
configuration such as SIP configuration, Codec selection, NAT Traversal Setting and other  
miscellaneous configuration. Following is a snap shot of the advanced configuration page.  
Grandstream Device Configuration  
STATUS BASIC SETTINGS ADVANCED SETTINGS  
Admin Password:  
(purposely not displayed for security protection)  
FXS Port:  
sip.mycompany.com  
SIP Server:  
(e.g., sip.mycompany.com, or IP address)  
Outbound Proxy:  
(e.g., proxy.myprovider.com, or IP address, if any)  
3320  
SIP User ID:  
(the user part of an SIP address)  
3320  
Authenticate ID:  
(can be identical to or different from SIP User ID)  
Authenticate  
(purposely not displayed for security protection)  
Password:  
John Doe  
Name:  
(optional, e.g., John Doe)  
Register  
Expiration:  
60  
(in minutes. default 1 hour, max 45 days)  
5060  
local SIP port:  
(default 5060)  
5004  
local RTP port:  
(1024-65535, default 5004)  
Enable Call  
No  
Yes (if Yes, Call Forwarding & Call-Waiting-Disable are  
Features:  
supported locally)  
Send DTMF:  
in-audio  
via RTP (RFC2833)  
via SIP INFO  
DTMF Payload  
Type:  
101  
FXO Port:  
sip.mycompany.com  
SIP Server:  
Outbound Proxy:  
SIP User ID:  
(e.g., sip.mycompany.com, or IP address)  
(e.g., proxy.myprovider.com, or IP address, if any)  
(the user part of an SIP address)  
6602  
6602  
Authenticate ID:  
(can be identical to or different from SIP User ID)  
Authenticate  
Password:  
(purposely not displayed for security protection)  
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John Doe  
Name:  
(optional, e.g., John Doe)  
Register  
Expiration:  
60  
(in minutes. default 1 hour, max 45 days)  
5062  
5008  
local SIP port:  
local RTP port:  
(default 5062)  
(1024-65535, default 5008)  
Enable Call  
Features:  
No  
Yes (if Yes, Call Forwarding & Call-Waiting-Disable are  
supported locally)  
Send DTMF:  
in-audio  
via RTP (RFC2833)  
via SIP INFO  
DTMF Payload  
Type:  
101  
General  
Settings:  
Misc.  
current setting is " PCMU"  
current setting is " PCMA"  
Preferred Vocoder:  
(in listed order)  
choice 1:  
choice 2:  
choice 3:  
choice 4:  
choice 5:  
choice 6:  
choice 7:  
2
current setting is " G723"  
current setting is " G729"  
current setting is " G726-32"  
current setting is " G728"  
current setting is " iLBC"  
Voice Frames per  
TX:  
(up to 10/20/32/64 for G711/G726/G723/other codecs respectively)  
G723 rate:  
iLBC frame size:  
6.3kbps encoding rate  
20ms 30ms  
5.3kbps encoding rate  
99  
iLBC payload type:  
Silence Suppression:  
SIP Registration:  
(between 96 and 127, default is 98)  
No  
Yes  
No  
Yes  
Unregister On  
Reboot:  
Yes  
No  
48  
Layer 3 QoS:  
Layer 2 QoS:  
Use DNS SRV:  
(Diff-Serv or Precedence value)  
0
0
802.1Q/VLAN Tag  
No Yes  
802.1p priority value  
(0-7)  
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User ID is phone  
number:  
No  
Yes  
Early Dial:  
No  
Yes (use "Yes" only if proxy supports 484 response)  
(this prefix string is added to each dialed number)  
Dial Plan Prefix:  
No Key Entry  
Timeout:  
4
(in seconds, default is 4 seconds)  
Use # as Dial Key:  
Use random port:  
No  
No  
No  
Yes (if set to Yes, "#" will function as the "(Re-)Dial" key)  
Yes  
NAT Traversal:  
stun.mycompany.com  
Yes, STUN server is:  
(URI or IP:port)  
20  
keep-alive interval:  
Use NAT IP  
(in seconds, default 20 seconds)  
(if specified, this IP address is used in SIP/SDP message)  
(if specified, the content will appear in Proxy-Require header)  
Proxy-Require:  
TFTP Upgrade  
Server:  
192  
168  
1
30  
.
.
.
(for remote software upgrade and configuration)  
HTTP Upgrade  
Server:  
(IP address or URL)  
7
Auto Upgrade:  
No  
Yes, check for new firmware every  
days (default 7 days)  
SUBSCRIBE for  
MWI:  
No, do not send SUBSCRIBE for Message Waiting Indication  
Yes, send periodical SUBSCRIBE for Message Waiting Indication  
Offhook Auto-Dial:  
(User ID/extension to dial automatically when offhook)  
Send Flash Event:  
FXS Impedance:  
No  
Yes (Flash will be sent as a DTMF event if set to Yes)  
current setting is " 600 Ohm (North America)"  
current setting is " Bellcore"  
Caller ID Scheme:  
Disable Call-  
Waiting:  
No  
Yes  
Polarity Reversal:  
NTP Server:  
No  
Yes (reverse polarity upon call establishment and termination)  
(URI or IP address)  
time.nist.gov  
Send Anonymous:  
No  
No  
Yes (caller ID will be blocked if set to Yes)  
Yes (configuration update via keypad is disabled if set to Yes)  
Lock keypad update:  
Special Feature:  
current setting is "Standard"  
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WAN side http  
No  
Yes (WAN side access to http server will be rejected if set to No)  
access:  
PSTN access code:  
(key pattern to use the PSTN line, default is "*00")  
Update  
All Rights Reserved Grandstream Networks, Inc. 2004  
Admin Password This contains the password to access the Advanced Web Configuration  
page. This field is case sensitive.  
SIP Server  
SIP Server’s URI or IP address  
Outbound Proxy SIP Outbound Proxy Server’s URI or IP address  
SIP User ID  
SIP service subscriber’s User ID  
Authenticate ID  
SIP service subscriber’s Authenticate ID. Can be identical to or different  
from SIP User ID  
Authenticate  
Password  
SIP service subscriber’s account password  
Name  
SIP service subscriber’s name which will be used for Caller ID display  
Register  
Expiration  
This parameter allows the user to specify the time frequency (in  
minutes) the HandyTone ATA refreshes its registration with the  
specified registrar. The default interval is 60 minutes (or 1 hour). The  
maximum interval is 65535 minutes (about 45 days).  
Local SIP port  
Local RTP port  
This parameter defines the local SIP port the HandyTone ATA will listen  
and transmit. The default value for FXS port is 5060. The default value  
for FXO port is 5062.  
This parameter defines the local RTP-RTCP port pair the HandyTone  
ATA will listen and transmit. It is the base RTP port for channel 0. When  
configured, channel 0 will use this port _value for RTP and the  
port_value+1 for its RTCP; channel 1 will use port_value+2 for RTP and  
port_value+3 for its RTCP. The default value for FXS port is 5004. The  
default value for FXO port is 5008.  
Enable Call  
Features  
Default is No. If set to Yes, Call Forwarding & Do-Not-Disturb are  
supported locally  
Send DTMF  
This parameter controls how DTMF events are transmitted. There are 3  
ways: in audio which means DTMF is combined in audio signal (not very  
reliable with low-bit-rate codec), via RTP (RFC2833), or via SIP INFO.  
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DTMF Payload  
Type  
This parameter sets the payload type for DTMF using RFC2833  
Preferred  
Vocoder  
The HandyTone ATA supports up to 7 different Vocoder types including  
G.711 A-/U-lawG.723.1, G.726, G.728, G.729A/B, iLBC. Depending  
on the product model, some of these Vocoders may not be provided in  
standard release.  
Users can configure Vocoders in a preference list that will be included  
with the same preference order in SDP message. The first Vocoder in  
this list can be entered by choosing the appropriate option in “Choice 1”.  
Similarly, the last Vocoder in this list can be entered by choosing the  
appropriate option in “Choice 7”.  
Voice Frames per This field contains the number of voice frames to be transmitted in a  
TX  
single packet. When setting this value, the user should be aware of the  
requested packet time (used in SDP message) as a result of configuring  
this parameter. This parameter is associated with the first vocoder in the  
above vocoder Preference List or the actual used payload type  
negotiated between the 2 conversation parties at run time.  
e.g., if the first vocoder is configured as G723 and the “Voice Frames  
per TX” is set to be 2, then the “ptime” value in the SDP message of an  
INVITE request will be 60ms because each G723 voice frame contains  
30ms of audio. Similarly, if this field is set to be 2 and if the first vocoder  
chosen is G729 or G711 or G726, then the “ptime” value in the SDP  
message of an INVITE request will be 20ms.  
If the configured voice frames per TX exceeds the maximum allowed  
value, the HandyTone ATA will use and save the maximum allowed  
value for the corresponding first vocoder choice. The maximum value for  
PCM is 10(x10ms) frames; for G726, it is 20 (x10ms) frames; for G723,  
it is 32 (x30ms) frames; for G729/G728, 64 (x10ms) and 64 (x2.5ms)  
frames respectively.  
G723 Rate:  
This defines the encoding rate for G723 vocoder. By default, 6.3kbps  
rate is chosen.  
iLBC frame size: This sets the iLBC size in 20ms or 30ms  
iLBC payload  
type:  
This defines payload time for iLBC. Default value is 98. The valid range  
is between 96 and 127.  
Silence  
Suppression  
This controls the silence suppression/VAD feature of G723 and G729. If  
set to “Yes”, when a silence is detected, small quantity of VAD packets  
(instead of audio packets) will be sent during the period of no talking. If  
set to “No”, this feature is disabled.  
SIP Registration  
This parameter controls whether the HandyTone ATA needs to send  
REGISTER messages to the proxy server. The default setting is “Yes”.  
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Unregister on  
Reboot  
Default is No. If set to yes, the SIP user will be unregistered on reboot.  
Layer 3 QoS  
Layer 2 QoS  
Use DNS SRV:  
This field defines the layer 3 QoS parameter which can be the value  
used for IP Precedence or Diff-Serv or MPLS. Default value is 48.  
This contains the value used for layer 2 VLAN tag. Default setting is  
blank.  
Default is No. If set to Yes the client will use DNS SRV for server lookup  
User ID is Phone  
Number  
If the HandyTone ATA has an assigned PSTN telephone number, this  
field should be set to “Yes”. Otherwise, set it to “No”. If “Yes” is set, a  
“user=phone” parameter will be attached to the “From” header in SIP  
request  
Early Dial  
Default is No. Use only if proxy supports 484 response  
Sets the prefix added to each dialed number  
Default is 4 seconds.  
Dial Plan Prefix  
No Key Entry  
Timeout  
Use # as  
Send Key  
This parameter allows users to configure the “#” key to be used as the  
“Send” (or “Dial”) key. If set to “Yes”, pressing this key will immediately  
trigger the sending of dialed string collected so far. In this case, this key  
is essentially equivalent to the “(Re)Dial” key. If set to “No”, this “#” key  
will then be included as part of the dial string to be sent out.  
Use Random Port This parameter, when set to Yes, will force random generation of both  
the local SIP and RTP ports. This is usually necessary when multiple  
HandyTone ATAs are behind the same NAT.  
NAT Traversal  
This parameter defines whether the HandyTone ATA NAT traversal  
mechanism will be activated or not. If activated (by choosing “Yes”) and  
a STUN server is also specified, then the HandyTone ATA will behave  
according to the STUN client specification. Under this mode, the  
embedded STUN client inside the HandyTone ATA will attempt to detect  
if and what type of firewall/NAT it is sitting behind through  
communication with the specified STUN server. If the detected NAT is a  
Full Cone, Restricted Cone, or a Port-Restricted Cone, the HandyTone  
ATA will attempt to use its mapped public IP address and port in all its  
SIP and SDP messages. If the NAT Traversal field is set to “Yes” with  
no specified STUN server, the HandyTone ATA will periodically (every  
20 seconds or so) send a blank UDP packet (with no payload data) to  
the SIP server to keep the “hole” on the NAT open.  
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Keep-alive  
interval  
This parameter specifies how often the HandyTone ATA sends a blank  
UDP packet to the SIP server in order to keep the “hole” on the NAT  
open.  
Use NAT IP:  
NAT IP address used in SIP/SDP message. Default is blank.  
Proxy-Require  
SIP Extension to notify SIP server that the unit is behind the  
NAT/Firewall.  
TFTP Upgrade  
Server  
This is the IP address of the configured TFTP server. If it is non-zero or  
not blank, the HandyTone ATA will attempt to retrieve new configuration  
file or new code image from the specified TFTP server at boot time. It  
will make up to 3 attempts before timeout and then it will start the boot  
process using the existing code image in the Flash memory. If a TFTP  
server is configured and a new code image is retrieved, the new  
downloaded image will be verified and then saved into the Flash  
memory.  
HTTP Upgrade  
Server  
The URL for the HTTP server used for firmware upgrade and  
configuration via HTTP. For example,  
Here “:6688” is the specific TCP port that the HTTP server is listening  
at, it can be omitted if using default port 80.  
Note: Auto Upgrade has to be set to Yes to enable HTTP upgrade.  
Auto Upgrade  
Choose Yes to enable HTTP upgrade and provisioning.  
In “Check for new firmware every” field, enter the number of days to  
check the HTTP server for firmware upgrade or configuration.  
Subscribe for  
MWI:  
Default is NO. When set to “Yes” a SUBSCRIBE for Message Waiting  
Indication will be sent periodically.  
Offhook  
Auto-Dial  
This parameter allows users to configure a User ID or extension number  
to be automatically dialed upon offhook. Please note that only the user  
part of a SIP address needs to be entered here. The HandyTone ATA  
will automatically append the “@” and the host portion of the  
corresponding SIP address.  
Send Flash Event This parameter allows users to control whether to send an SIP NOTIFY  
message indicating the Flash event, or just to switch to the voice  
channel when users press the Flash key.  
FXS Impedance  
Selects the impedance of the analog telephone connected to the Phone  
port.  
Caller ID Scheme • Bellcore (North America)  
• ETSI-FSK (France, Germany, Norway, Taiwan, UK-CCA)  
• ETSI-DTMF (Finland, Sweden)  
• DTMF (Denmark)  
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Disable Call  
Waiting  
Default is No.  
Polarity Reversal Default is No. If set to Yes, polarity will be reversed upon call  
establishment and termination.  
NTP server  
This parameter defines the URI or IP address of the NTP server which  
is used by the HandyTone ATA to display the current date/time.  
Send Anonymous If this parameter is set to “Yes”, the “From” header in outgoing INVITE  
message will be set to anonymous, essentially blocking the Caller ID  
from displaying.  
Lock keypad  
update  
If this parameter is set to “Yes”, the configuration update via keypad is  
disabled.  
Special Feature  
• Standard  
• Nortel MCS  
• Broadsoft  
• Howdy  
WAN side http  
access  
If this parameter is set to “No”, the HTML configuration update via WAN  
port is disabled.  
PSTN access  
code  
This field allows users to customize their own code to access the PSTN  
line. Default is “*00”.  
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6.2.4 Saving the Configuration Changes  
Once a change is made, users should press the “Update” button in the Configuration page.  
The HandyTone-488 will then display the following screen to confirm that the changes have  
been saved.  
Grandstream Device Configuration  
STATUS BASIC SETTINGS ADVANCED SETTINGS  
Your configuration changes have been saved.  
They will take effect on next reboot.  
All Rights Reserved Grandstream Networks, Inc. 2004  
Users are recommended to power cycle the HandyTone-488 after seeing the above  
message.  
6.2.5 Rebooting the HandyTone-488 from Remote  
The administrator of the HandyTone-488 can remotely reboot the HandyTone ATA by  
pressing the “Reboot” button at the bottom of the configuration page. Once done, the  
following screen will be displayed to indicate that rebooting is underway.  
Grandstream Device Configuration  
The device is rebooting now...  
You may relogin by clicking on the link below in 30 seconds.  
Click to relogin  
All Rights Reserved Grandstream Networks, Inc. 2004  
At this point, the user can relogin to the HandyTone-488 after waiting for about 30 seconds.  
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6.3 Configuration through a Central Server  
Grandstream HandyTone ATAs can be automatically configured from a central provisioning  
system.  
When HandyTone ATA boot up, it will send TFTP or HTTP request to download configuration  
files, there are two configuration files, one is “cfg.txt” and the other is “cfg000b82xxxxxx”,  
where “000b82xxxxxx” is the MAC address of the HandyTone ATA. For more information  
regarding configuration file format, please refer to the related Grandstream documentation.  
The configuration file can be downloaded via TFTP or HTTP from the central server. A  
service provider or an enterprise with large deployment of HandyTone ATAs can easily  
manage the configuration and service provisioning of individual devices remotely and  
automatically from a central server. GAPS (Grandstream Automated Provisioning System)  
uses enhanced (NAT friendly) TFTP or HTTP (thus no NAT issues) and other communication  
protocols to communicate with each individual HandyTone ATA for firmware upgrade, remote  
reboot, etc.  
Grandstream provides a licensed provisioning system called GAPS that can be used to  
support automated configuration of HandyTone ATA. To enable this feature on the  
HandyTone ATA, a user just needs to enter the IP address of the GAPS server in the TFTP  
server field of the configuration screen, or enter the HTTP provisioning Server URL in the  
HTTP Upgrade Server field. Then reboot the HandyTone ATA.  
For details on how GAPS works, please refer to the documentation of GAPS product.  
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7 Software Upgrade  
7.1 Upgrade through HTTP  
To upgrade software, HandyTone-488 can be configured with an HTTP server where the new  
code image file is located. For example, following URL in the HTTP Upgrade Server:  
Where firmware.mycompany.com is the FQDN of the HTTP server, “:6688” is the TCP port  
the HTTP server listening to, “/Grandstream/1.0.0.4” is the RELATIVE directory to the root dir  
in HTTP server. Thus, you can put different firmware into different directory as well.  
Note: To enable HTTP firmware upgrade, “Auto Upgrade” field has to be set to Yes. In  
addition, the ATA will check the HTTP server in the number of days that is defined in “Check  
for new firmware every” field.  
7.2 Upgrade through TFTP  
To upgrade software, HandyTone-488 can be configured with a TFTP server where the new  
code image is located. The TFTP upgrade can work in either static IP or DHCP mode using  
private or public IP address. It is recommended to set the TFTP server address in either a  
public IP address or on the same LAN with the HandyTone-488.  
There are two ways to set up the TFTP server to upgrade the firmware, namely through voice  
menu prompt or via the HandyTone-488’s Web configuration interface. To configure the  
TFTP server via voice prompt, follow section 5.1 with option 06, once set up the TFTP IP  
address, power cycle the ATA, the firmware will be fetched once the ATA boots up.  
To configure the TFTP server via the Web configuration interface, open up your browser to  
point at the IP address of the HandyTone-488. Input the admin password to enter the  
configuration screen. From there, enter the TFTP server address in the designated field  
towards the bottom of the configuration screen.  
Once the TFTP server is configured, power cycle the HandyTone-488.  
TFTP checking is only performed during the initial power up. If the configured TFTP server is  
found and a new code image is available, the HandyTone-488 will attempt to retrieve the new  
image files by downloading them into the HandyTone-488’s SRAM. During this stage, the  
HandyTone-488’s LED will blink until the checking/downloading process is completed. Upon  
verification of checksum, the new code image will then be saved into the Flash. If TFTP fails  
for any reason (e.g., TFTP server is not responding, there are no code image files available  
for upgrade, or checksum test fails, etc), the HandyTone-488will stop the TFTP process and  
simply boot using the existing code image in the flash.  
TFTP process may take as long as 1 to 2 minutes over the Internet, or just 20+ seconds if it is  
performed on a LAN. Users are recommended to conduct TFTP upgrade in a controlled LAN  
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environment if possible. For those who do not have a local TFTP server, Grandstream  
provides a NAT-friendly TFTP server on the public Internet for firmware upgrade. Please  
check the Service or Support section of Grandstream’s Web site to obtain this TFTP server’s  
IP address.  
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8 Restore Factory Default Setting  
Warning: Restore the Factory Default Setting will delete all configuration information of  
the device.  
Please disconnect network cable and power cycle the unit before trying to reset the unit to  
factory default. The steps are as follows:  
A. Find the MAC address of the device. It is a 12 digits number located on the bottom of the  
unit.  
B. Encode the MAC address. Please use the following mapping:  
0-9: 0-9  
A: 22  
B: 222  
C: 2222  
D: 33  
E: 333  
F: 3333  
For example, 000B82004D2A will be encoded to 0002228200433222.  
C. To perform factory reset:  
1. Press “***” or the LED button for voice prompt.  
2. Enter “99” and get the voice prompt “Reset”.  
3. Enter the encoded MAC address of the device. A "click" sound will be heard.  
4. Wait for 15 seconds.  
The device is now restored to the factory default setting.  
32  

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