Grandstream Networks Telephone Accessories HT 502 User Manual

Grandstream Networks, Inc.  
HT– 502  
Dual FXS Port  
Analog Telephone Adaptor  
HT–502 User Manual  
Firmware Version 1.0.0.39  
Download from Www.Somanuals.com. All Manuals Search And Download.  
TABLE OF FIGURES  
HT–502 USER MANUAL  
FIGURE 1: CONNECTING THE HT–502 .............................................................................................................5  
FIGURE 2: CONFIGURING THE HT–502 ............................................................................................................6  
FIGURE 3: SCREENSHOT OF CONFIGURATION LOG IN PAGE ............................................................................16  
FIGURE 4: SCREENSHOT OF SAVE CONFIGURATION PAGE...............................................................................27  
TABLE OF TABLES  
HT–502 USER MANUAL  
TABLE 1: DEFINITIONS OF THE HT–502 CONNECTORS ....................................................................................... 5  
TABLE 2: DEFINITIONS OF THE HT–502 LEDS.................................................................................................... 6  
TABLE3: HT–502 TECHNICAL SPECIFICATIONS .................................................................................................. 8  
TABLE 4: HT–502 HARDWARE SPECIFICATIONS................................................................................................. 9  
TABLE 5: HT–502 IVR MENU DEFINITIONS ........................................................................................................ 9  
TABLE 6: HT–502 CALL FEATURE DEFINITIONS ............................................................................................... 13  
TABLE 7: HT–502 BASIC CONFIGURATION SETTINGS DEFINITIONS ................................................................... 18  
TABLE 8: HT–502 DEVICE STATUS PAGE DEFINITIONS..................................................................................... 21  
TABLE 9: HT–502 ADVANCED CONFIGURATION PAGE DEFINITIONS .................................................................. 21  
TABLE 10: HT–502 INDIVIDUAL ACCOUNT SETTINGS DEFINITIONS .................................................................... 23  
CONFIGURATION GUI INTERFACE EXAMPLES  
HT–502 USER MANUAL  
(http://www.grandstream.com/user_manuals/GUI/GUI_HT502.rar)  
1. SCREENSHOT OF ADVANCED USER CONFIGURATION PAGE  
2. SCREENSHOT OF BASIC SETTINGS CONFIGURATION PAGE  
3. SCREENSHOT OF FXS PORT 1 CONFIGURATION LOGIN PAGE  
4. SCREENSHOT OF FXS PORT 2 CONFIGURATION PAGE  
5. SCREENSHOT OF STATUS CONFIGURATION LOG IN PAGE  
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WELCOME  
Thank you for purchasing Grandstream’s HT–502, the affordable, feature rich Analog Telephone Adaptor.  
Grandstream HandyTone-502 is a new addition to the popular HandyTone ATA product family. It features  
the rich audio quality, a broad range of voice codecs, and functionality of the HT–502, including two (2)  
FXS ports each with independent SIP accounts.  
This manual will help you learn how to operate and manage your HandyTone-502 Analog Telephone  
Adaptor and make the best use of its many upgraded features including simple and quick installation, 3-  
way conferencing, and direct IP-IP Calling. This HT–502 is very easy to manage and configure, and is  
specifically designed to be an easy to use and affordable VoIP solution for both the residential user and the  
tele-worker.  
SAFETY COMPLIANCES  
The HT–502 phone complies with FCC/CE and various safety standards. The HT–502 power adaptor is  
compliant with UL standard. Only use the universal power adapter provided with the HT–502 package.  
The manufacturer’s warranty does not cover damages to the phone caused by unsupported power  
adaptors.  
WARRANTY  
If you purchased your HT–502 from a reseller, please contact the company where you purchased your  
phone 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  
before you return the product. Grandstream reserves the right to remedy warranty policy without prior  
notification.  
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. Please do not use a different power adaptor with the HT–502 as it may cause damage to the  
products and void the manufacturer warranty.  
This document is contains links to Grandstream GUI Interfaces. Please download these examples  
This document is subject to change without notice. The latest electronic version of this user manual is  
Reproduction or transmittal of the entire or any part, in any form or by any means, electronic or print,  
for any purpose is not permitted without the express written permission of Grandstream Networks, Inc.  
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CONFIGURE YOUR HT–502  
Configuring your HT–502 and connecting the unit to the VoIP network is very simple. The HT–502 is easy to  
configure using the embedded GUI pages and the following five (5) steps outlined below. Examples of the  
Before you begin, please verify the contents of the HT–502 package.  
EQUIPMENT PACKAGING  
Unpack and check all accessories. Equipment included in the package:  
one 12V universal power adapter  
one Ethernet cable  
one device unit  
one HT502 stand  
CONNECT YOUR ATA  
First, familiarize yourself with the features of the HT–502. The HT–502 has two FXS ports. Each FXS port  
can have a separate SIP account. This is a key feature of HT–502. Both ports can make calls  
concurrently.  
FIGURE 1: CONNECTING THE HT–502  
HT-502  
HT-502  
Front View  
Back View  
RJ-45 Ports  
10/100 Mbps  
RJ-11 FXS Ports  
(Phone)  
Display LEDs  
(green)  
Reset  
Power  
Supply  
(12V)  
TABLE 1: DEFINITIONS OF THE HT–502 CONNECTORS  
Power Cable  
Power adapter connection  
Connect to the internal LAN network or router.  
Connect the LAN port with an Ethernet cable to your PC.  
WAN Port (RJ-45)  
LAN Port (RJ-45)  
RESET  
Factory Reset button. Press for 7 seconds to reset factory default settings.  
FXS port to be connected to analog phones / fax machines.  
FXS port to be connected to analog phones / fax machines.  
PHONE1 (RJ-11)  
PHONE2 (RJ-11)  
There are five (5) LED buttons that help you manage the status of your HandyTone.  
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TABLE 2: DEFINITIONS OF THE HT–502 LEDS  
Power LED  
WAN LED  
LAN LED  
Indicates Power. Remains ON when Power is connected and turned ON.  
Indicates LAN (or WAN) port activity.  
Indicates PC (or LAN) port activity.  
PHONE1 /  
Indicate status of the respective FXS Ports-PHONE1 / PHONE2 on the back  
PHONE2 LED  
panel  
Busy – ON (Solid Green)  
Available – OFF  
Slow blinking FXS LEDs indicates voicemail for that port.  
NOTE: All LEDs display green when ON. Slow blinking of Power, WAN and LAN LED together indicate  
the product in firmware upgrading or provision state  
CONFIGURE YOUR ATA  
The HT–502 is easy to configure using the embedded GUI pages and the following five (5) steps.  
FIVE EASY STEPS TO CONFIGURE THE HT–502  
1. Connect a standard touch-tone analog telephone (or fax machine) to first FXS port.  
2. Connect another standard touch-tone analog telephone (or fax machine) to second FXS port.  
3. Insert the Ethernet cable into the WAN port of HT–502 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 HT–502.  
5. Insert the power adapter into the HT–502 and connect it to a wall outlet.  
FIGURE 2: CONFIGURING THE HT–502  
Internet  
ADSL/Cable  
Modem Ethernet  
Analog Phone  
Analog Phone  
WAN  
FXS  
FXS  
Cordless  
Phone  
Cordless  
Phone  
LAN  
Fax  
Fax  
z
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PRODUCT OVERVIEW  
The HT–502 is a full feature voice and fax-over IP device that offers a high-level of integration including  
dual 10M/100Mbps network ports with integrated router, NAT, DHCP server, dual port FXS telephone  
gateway, market-leading sound quality, rich functionalities, and a compact and lightweight design. The HT–  
502 fully compatible with SIP industry standard and can interoperate with many other SIP compliant  
devices and software on the market. Moreover, it supports comprehensive voice codecs including G.711  
(a/µ-law), G.723.1, G.726 (16/24/32/48 bit rates), G.729A/B/E and iLBC.  
KEY FEATURES  
Ethernet  
Ports  
FXS PSTN Pass – Voice Mail  
Remote  
Configuration  
DHCP  
Voice Codec  
Port  
through  
Indicator  
iLBC, G.711, G.723,  
G.726(16/24/32/48),  
T.38, G.729A/B/E  
2 RJ-45 (LAN)  
Server/Client  
2
No  
Yes  
TFTP/HTTP  
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TABLE3: HT–502 TECHNICAL SPECIFICATIONS  
HT502 Analog Telephone Adaptor  
Telephone Interfaces  
Network Interface  
LED Indicators  
2 FXS ports, 2 SIP accounts  
Two (2) 10M/100 Mbps, RJ-45  
Power, WAN, LAN, PHONE1 and PHONE2  
Factory Reset button.  
Reset Button  
Voice over Packet Voice Activity Detection (VAD) with CNG (comfort noise generation) and PLC  
Capabilities  
(packet loss concealment), Dynamic Jitter Buffer, Modem detection & auto-switch  
to G.711, Packetized Voice Protocol Unit (supports RTP/RTCP and AAL2  
protocol), G.168 compliant Echo Cancellation, LEC (line echo cancellation) with  
NLP, Asymmetric RTP stream  
Voice Compression  
G.711 + Annex I (PLC), Annex II (VAD/CNG format) encoder and decoder,  
G.723.1A, G.726(ADPCM), G.729A/B/E, iLBC, G.726 provides proprietary VAD,  
CNG, and signal power estimation, Voice Play Out unit (reordering, fixed and  
adaptive jitter buffer, clock synchronization), AGC (automatic gain control), Status  
output, Decoder controlling via voice packet header  
DHCP Server/Client  
Telnet Server  
Yes, NAT Router or Switched Mode  
Yes  
Fax over IP  
T.38 compliant Group 3 Fax Relay up to 14.4kpbs and auto-switch to G.711 for  
Fax Pass-through (pending), Fax Datapump V.17, V.19, V.27ter, V.29 for T.38  
fax relay  
QoS  
Diffserve, TOS, 802.1 P/Q VLAN tagging  
RTP/RTCP  
IP Transport  
DTMF Method  
Flexible DTMF transmission method, user interface of In-audio, RFC2833, and/or  
SIP Info  
IP Signaling  
Provisioning  
Control  
SIP (RFC 3261)  
TFTP, HTTP, HTTPS (pending)  
TLS/SIPS , SIP over TCP/TLS  
Management  
Syslog support, HTTPS and telnet (pending), remote management using Web  
browser, Support Layer 2 (802.1Q, VLAN, 802.1p) and Layer 3 QoS (Tos,  
DiffSery, MPLS), Auto/manual provisioning system  
Dial Plan  
UPnP  
Yes  
Yes  
Power  
Output: 12VDC / Input: 100–240 VAC/50-60 Hz  
Operational: 32o–104oF or 0o–40oC  
Storage: 10o–130o F / Humidity: 10–90% Non-condensing  
115mm (L) x 75mm (W) x 27mm (H)  
Environmental  
Dimensions  
(H x W x D)  
Short and long haul  
REN3: Up to150 ft on 24 AWG line  
Call  
Features  
Handling Caller ID display or block, Call waiting caller ID, Call waiting/flash, Call transfer,  
hold, forward, mute, 3-way conferencing, message waiting, Do-Not-Disturb  
(DND), call-return service  
Caller ID  
Bellcore Type 1 & 2, ETSI, BT, NTT, and DTMF-based CID  
Polarity Reversal  
Wink  
/
Yes  
EMC  
EN55022/EN55024 and FCC part15 Class B  
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TABLE 4: HT–502 HARDWARE SPECIFICATIONS  
LAN Interface  
LED  
2 x RJ45 10/100Mbps (integrated router)  
5 LEDs (GREEN)  
Universal  
Switching  
Input: 100-240V AC, 50/60Hz, 0.5A Max  
Output: 12V DC, 1.25A  
UL certified  
Power Adaptor  
Dimension  
Weight  
115mm (L) x 75mm (W) x 27mm (H)  
94 g (0.21lbs)  
Temperature  
Humidity  
32~104°F / 0~40°C  
10% - 90% (non-condensing)  
FCC, CE  
Compliance  
BASIC OPERATIONS  
BECOMING FAMILIAR WITH THE VOICE PROMPTS  
HT–502 stores a voice prompt menu (Interactive Voice Response or IVR) for quick browsing and simple  
configuration. The IVR menu and the LED button work with any of the FXS port. To enter the IVR menu,  
pick up the handset and dial “***”.  
TABLE 5: HT–502 IVR MENU DEFINITIONS  
Menu  
Voice Prompt  
Options  
Main Menu “Enter a Menu Option”  
Press “*” for the next menu option  
Press “#” to return to the main menu  
Enter 01-05, 07,10, 12-17,47 or 99 menu options  
01  
02  
“DHCP Mode”,  
“Static IP Mode”  
Press “9” to toggle the selection  
If using “Static IP Mode”, configure the IP address  
information using menus 02 to 05.  
If using “Dynamic IP Mode”, all IP address information  
comes from the DHCP server automatically after reboot.  
“IP Address “ + IP address  
The current WAN IP address is announced  
If using “Static IP Mode”, enter 12 digit new IP address.  
03  
04  
“Subnet “ + IP address  
“Gateway “ + IP address  
Same as menu 02  
Same as menu 02  
05  
07  
“DNS Server “ + IP address Same as menu 02  
Preferred Vocoder  
Press “9” to move to the next selection in the list:  
PCM U  
PCM A  
iLBC  
G-726  
G-723  
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G-729  
10  
12  
“MAC Address”  
Announces the Mac address of the unit.  
WAN Port Web Access  
Press “9” to toggle between enable / disable  
13  
14  
15  
Firmware  
Address  
Server  
IP  
Announces current Firmware Server IP address. Enter  
12 digit new IP address.  
Configuration Server IP  
Address  
Announces current Config Server Path IP address. Enter  
12 digit new IP address.  
Upgrade Protocol  
Upgrade protocol for firmware and configuration update.  
Press “9” to toggle between TFTP / HTTP  
16  
17  
Firmware Version  
Firmware Upgrade  
Firmware version information.  
Firmware upgrade mode. Press “9” to toggle among the  
following three options:  
- always check  
- check when pre/suffix changes  
- never upgrade  
47  
99  
“Direct IP Calling”  
“RESET”  
Enter the target IP address to make a direct IP call, after  
dial tone. (See “Make a Direct IP Call”.)  
Press “9” to reboot the device  
Enter MAC address to restore factory default setting  
(See Restore Factory Default Setting section)  
“Invalid Entry”  
Automatically returns to main menu  
Five Success Tips when using the Voice Prompt  
1. “*” shifts down to the next menu option  
2. “#” returns to the main menu  
3. “9” functions as the ENTER key in many cases to confirm an option  
4. All entered digit sequences have known lengths - 2 digits for menu option and 12 digits for IP  
address. For IP address, add 0 before the digits if the digits are less than 3 (i.e. - 192.168.0.26  
should be key in like 192168000026. No decimal is needed).  
5. Key entry can not be deleted but the phone may prompt error once it is detected  
PLACING A PHONE CALL  
PHONE OR EXTENSION NUMBERS  
1. Dial the number directly and wait for 4 seconds (Default “No Key Entry Timeout”); or  
2. Dial the number directly and press # (Use # as dial key” must be configured in web configuration).  
Examples:  
1. Dial an extension directly on the same proxy, (e.g. 1008), and then press the # or wait for 4 seconds.  
2. Dial an outside number (e.g. (626) 666-7890), first enter the prefix number (usually 1+ or international  
code) followed by the phone number. Press # or wait for 4 seconds. Check with your VoIP service  
provider for further details on prefix numbers.  
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DIRECT IP CALLS  
Direct IP calling allows two parties, that is, a FXS Port with an analog phone and another VoIP Device, to  
talk to each other in an ad hoc fashion without a SIP proxy.  
Elements necessary to completing a Direct IP Call:  
1. Both HT–502 and other VoIP Device, have public IP addresses, or  
2. Both HT–502 and other VoIP Device are on the same LAN using private IP addresses, or  
3. Both HT–502 and other VoIP Device can be connected through a router using public or private IP  
addresses (with necessary port forwarding or DMZ).  
HT-502 supports two ways to make Direct IP Calling:  
Using IVR  
1. Pick up the analog phone then access the voice menu prompt by dial “***”  
2. Dial 47” to access the direct IP call menu  
3. Enter the IP address after the dial tone and voice prompt “Direct IP Calling”  
Using Star Code  
1. Pick up the analog phone then dial “*47”  
2. Enter the target IP address.  
Note: NO dial tone will be played between step 1 and 2.  
Destination ports can be specified using “*” (encoding for “:”) followed by the port number.  
Examples:  
a) If the target IP address is 192.168.0.160, the dialing convention is  
*47 or Voice Prompt with option 47, then 192*168*0*160.  
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.  
b) If the target IP address/port is 192.168.1.20:5062, then the dialing convention would be:  
*47 or Voice Prompt with option 47, then 192*168*0*160*5062 followed by pressing the “#” key  
if it is configured as a send key or wait for 4 seconds.  
NOTE: When completing direct IP call, the “Use Random Port” should set to “NO”. You cannot make  
direct IP calls between FXS1 to FXS2 since they are using same IP.  
CALL HOLD  
Place a call on hold by pressing the “flash” button on the analog phone (if the phone has that button).  
Press the “flash” button again to release the previously held Caller and resume conversation. If no “flash”  
button is available, use “hook flash” (toggle on-off hook quickly). You may drop a call using hook flash.  
CALL WAITING  
Call waiting tone (3 short beeps) indicates an incoming call, if the call waiting feature is enabled. Toggle  
between incoming call and current call by pressing the “flash” button. First call is placed on hold. Press  
the “flash” button to toggle between two active calls.  
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CALL TRANSFER  
Blind Transfer  
Assume that call Caller A and B are in conversation. A wants to Blind Transfer B to C:  
3. Caller A presses FLASH on the analog phone to hear the dial tone.  
4. Caller A dials *87 then dials caller C’s number, and then # (or wait for 4 seconds)  
5. Caller A will hear the confirm tone. Then, A can hang up.  
NOTE: Enable Call Feature” must be set to “Yes” in web configuration page.  
Caller A can place a call on hold and wait for one of three situations:  
1. A quick confirmation tone (similar to call waiting tone) followed by a dial-tone. This indicates the  
transfer is successful (transferee has received a 200 OK from transfer target). At this point, Caller  
A can either hang up or make another call.  
2. 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.  
3. Continuous busy tone. The phone has timed out. Note: continuous busy tone does not indicate  
the transfer has been successful, nor does it indicate the transfer has failed. It often means there  
was a failure to receive second NOTIFY – check firmware for most recent release.  
Attended Transfer  
Assume that Caller A and B are in conversation. Caller A wants to Attend Transfer B to C:  
1. Caller A presses FLASH on the analog phone for dial tone.  
2. Caller A then dials Caller C’s number followed by # (or wait for 4 seconds).  
3. If Caller C answers the call, Caller A and Caller C are in conversation. Then A can hang up to  
complete transfer.  
4. If Caller C does not answer the call, Caller A can press “flash” to resume call with Caller B.  
NOTE: When Attended Transfer fails and A hangs up, the HT-502 will ring back user A to remind A that B  
is still on the call. A can pick up the phone to resume conversation with B.  
3-WAY CONFERENCING  
The HT-502 supports Bellcore style 3-way Conference.  
Instructions for 3-way conference:  
Assue that call party A and B are in conversation. Caller A(HT502) wants to bring third Caller C into  
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 presses 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. If A presses FLASH during conference, C will be dropped out  
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Call Features  
The HT–502 supports all the traditional and advanced telephony features.  
TABLE 6: HT–502 CALL FEATURE DEFINITIONS  
Key  
*30  
*31  
*47  
Call Features  
Block Caller ID (for all subsequent calls)  
Send Caller ID (for all subsequent calls)  
Direct IP Calling. Dial “*47” + “IP address”. No dial tone is played in the middle. Detail  
see Direct IP Calling section on page 12.  
*50  
*51  
*67  
*82  
Disable Call Waiting (for all subsequent calls)  
Enable Call Waiting (for all subsequent calls)  
Block Caller ID (per call). Dial “*67” + ” number ”. No dial tone is played in the middle.  
Send Caller ID (per call). Dial “*82” + ” number ”. No dial tone is played in the middle.  
*69  
Call Return Service: Dial *69 and the phone will dial the last incoming phone number  
received.  
*70  
*71  
*72  
*73  
Disable Call Waiting (per call). Dial “*70” + ” number ”. No dial tone is played in the  
middle.  
Enable Call Waiting (per call). Dial “*71” + ” number ”. No dial tone is played in the  
middle.  
Unconditional Call Forward: Dial “*72” and then the forwarding number followed by  
“#”. Wait for dial tone and hang up. (dial tone indicates successful forward)  
Cancel Unconditional Call Forward. To cancel “Unconditional Call Forward”, dial  
“*73”, wait for dial tone, then hang up.  
*78  
Enable Do Not Disturb (DND): When enabled all incoming calls are rejected.  
*79  
*87  
*90  
Disable Do Not Disturb (DND): When disabled, incoming calls are accepted.  
Blind Transfer  
Busy Call Forward: Dial “*90” and then the forwarding number followed by “#”. Wait  
for dial tone then hang up.  
*91  
*92  
*93  
Cancel Busy Call Forward. To cancel “Busy Call Forward”, dial “*91”, wait for dial tone,  
then hang up.  
Delayed Call Forward. Dial “*92” and then the forwarding number followed by “#”. Wait  
for dial tone then hang up.  
Cancel Delayed Call Forward. To cancel Delayed Call Forward, dial “*93”, wait for dial  
tone, then hang up.  
Flash/Hook Toggles between active call and incoming call (call waiting tone). If not in conversation,  
flash/hook will switch to a new channel for a new call.  
#
Pressing pound sign will serve as Re-Dial key.  
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SENDING AND RECEIVING FAX  
HT–502 supports fax in two modes: 1) T.38 (Fax over IP) and 2) 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 T.38 as fax mode (default). If the  
service provider does not support T.38, pass-through mode may be used. If you have problems with  
sending or receiving Fax, toggle the Fax Tone Detection Mode setting.  
CONFIGURATION GUIDE  
CONFIGURING THE HT–502 VIA VOICE PROMPTS  
DHCP MODE  
Select voice menu option 01 to enable HT–502 to use DHCP.  
STATIC IP MODE  
Select voice menu option 01 to enable HT–502 to use STATIC IP mode, then use option 02, 03, 04, 05 to  
set up IP address, Subnet Mask, Gateway and DNS server respectively.  
FIRMWARE SERVER IP ADDRESS  
Select voice menu option 13 to configure the IP address of the firmware server.  
CONFIGURATION SERVER IP ADDRESS  
Select voice menu option 14 to configure the IP address of the configuration server.  
UPGRADE PROTOCOL  
Select voice menu option 15 to choose firmware and configuration upgrade protocol. User can choose  
between TFTP and HTTP.  
FIRMWARE UPGRADE MODE  
Select voice menu option 17 to choose firmware upgrade mode among the following three options: 1)  
always check, 2) check when pre/suffix changes, and 3) never upgrade  
WAN PORT WEB ACCESS  
Select voice menu option 12 to enable WAN Port Wed Access of the device configuration pages.  
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CONFIGURING THE HT–502 USING THE WEB BROWSER  
HT–502 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 HT–502 through a web browser such as  
Microsoft’s IE and AOL’s Netscape.  
ACCESS THE WEB CONFIGURATION MENU  
The HT–502 HTML configuration menu can be accessed via LAN or WAN port:  
From the LAN port:  
1. Directly connect a computer to the LAN port.  
2. Open a command window on the computer  
3. Type in “ipconfig /release”, the IP address etc. becomes 0.  
4. Type in “ipconfig /renew”, the computer gets an IP address in 192.168.2.x segment by default  
5. Open a web browser, type in the default gateway IP address. http://192.168.2.1. You will see the  
login page of the device.  
From the WAN port:  
The WAN port HTML configuration option is disabled by default from factory. To access the HTML  
configuration menu from the WAN port:  
1. Enable the “WAN Port Web Access” option via IVR option 12.  
2. Find the WAN IP address of the HT–502 using voice prompt menu option 02.  
3. Access the HT–502 Web Configuration page by the following URI via WAN port: http://HandyTone-  
IP-Address (the HT IP-Address is the WAN IP address for the HT–502).  
NOTE: If using a web browser to enter the configuration page, strip the leading “0”s because the browser  
will parse in octet. (i.e. if the IP address is: 192.168.001.014, please type in: 192.168.1.14).  
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END USER CONFIGURATION  
Once the HTTP request is sent from a web browser, the user will see a log in screen. There are two  
default passwords for the login page:  
User Level:  
Password: Web pages allowed:  
End User Level  
Administrator Level  
123  
Only Status and Basic Settings  
Browse all pages  
admin  
Only an administrator can access the “ADVANCED SETTINGS” configuration page.  
FIGURE 3: SCREENSHOT OF CONFIGURATION LOG IN PAGE  
NOTE: If you cannot log into the configuration page by using default password, please check with the VoIP  
service provider. The service provider may have provisioned and configured the device for you. The Basic  
Configuration Page is the first web GUI the user will see.  
Important Settings  
The end-user must configure the following settings according to the local environment. NOTE: Most  
settings on the web configuration pages are set to the default values.  
NAT SETTINGS  
If you plan to keep the gateway within a private network behind a firewall, we recommend using STUN  
Server. The following three (3) settings are useful in the STUN Server scenario:  
1. STUN Server (under Advanced Settings webpage)  
Enter a STUN Server IP (or FQDN) that you may have, or look up a free public STUN Server on the  
internet and enter it on this field. If using Public IP, keep this field blank.  
2. Use Random Ports (under Advanced Settings webpage)  
This setting depends on your network settings. Generally if you have multiple IP devices under the  
same network, it should be set to Yes. If using a Public IP address, set this parameter to No.  
3. NAT Traversal (under the Profile web pages)  
Set this to Yes when gateway is behind firewall on a private network.  
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DTMF METHODS  
DTMF Settings are in Profile pages.  
DTMF in-audio  
DTMF via RTP (RFC2833)  
DTMF via SIP INFO  
Enable one or more DTMF methods based on your PBX system.  
PREFERRED VOCODER (CODEC)  
The HT-502 supports a broad range of voice codecs. Under Profile web pages, choose your preferred  
order of different codecs:  
PCMU/A (or G711µ/a)  
G729 A/B/E  
G723  
G726 (16/24/32/40)  
iLBC  
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TABLE 7: HT–502 BASIC CONFIGURATION SETTINGS DEFINITIONS  
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End User Password Password to access the Web Configuration Menu. This field is case sensitive  
with a maximum length of 25 characters.  
Web Port  
By default, HTTP uses port 80. This field is for customizable web port.  
Default is set to YES.  
Telnet Server  
IP Address  
There are two modes to operate the HT–502:  
DHCP mode: all the field values for the Static IP mode are not used (even  
though they are still saved in the Flash memory.) The HT–502 acquires its IP  
address from the first DHCP server it discovers from the LAN it is connected.  
Using the PPPoE feature: set the PPPoE account settings. The HT–502 will  
establish a PPPoE session if any of the PPPoE fields is set.  
Static IP mode: configure the IP address, Subnet Mask, Default Router IP  
address, DNS Server 1 (primary), DNS Server 2 (secondary) fields. These fields  
are set 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.  
DHCP vendor class Used by clients and servers to exchange vendor-specific information. Default is  
ID  
HT500.  
PPPoE account ID PPPoE username. Necessary if ISP requires you to use a PPPoE (Point to  
Point Protocol over Ethernet) connection.  
PPPoE password  
PPPoE account password.  
PPPoE  
Name  
Service This field is optional. If your ISP uses a service name for the PPPoE connection,  
enter the service name here. Default is blank.  
Time Zone  
Controls how the date/time is displayed according to the specified time zone.  
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Self Defined Time  
Zone  
The syntax is std offset dst [offset],start[/time],end[/time]  
Default is set to : MTZ+6MDT+5,M3.2.0,M11.1.0  
MTZ+6MDT+5,  
It indicates a time zone with 6 hours offset with 1 hour ahead which is the US  
central time.  
It is positive (+) if the local time zone is west of the Prime Meridian and negative  
(-) if it is east.  
Prime Meridian (a.k.a: International or Greenwich Meridian)  
M3.2.0,M11.1.0  
The 1st number indicates Month: 1,2,3,..,12 (for Jan, Feb, .., Dec)  
The 2nd number indicates the nth iteration of the weekday: (1st Sunday, 3rd  
Tuesday etc)  
The 3rd number indicates Weekday: 0,1, 2, ..,6(for Sun, Mon, Tue, .., Sat)  
Therefore, this example is the DST which starts from the second Sunday of  
March to the 1st Sunday of November.  
Language  
Languages supported with voice prompt  
Device Mode  
This parameter controls whether the device is working in NAT router mode or  
Bridge mode. Save the setting and reboot prior to configuring HT–502.  
Enable  
UPnP If set to “Yes”, the HT-502 would act as an UPnP gateway for your UPnP enable  
support  
applications.  
UPnP - Universal Plug n Play  
Reply to ICMP on If set to “Yes”, the HT–502 will respond to the PING command from other  
WAN port  
computers, but it also is vulnerable to the DOS attack. Default is No.  
WAN  
HTTP/Telnet  
Access  
side If set to “Yes”, user can access the configuration page through the WAN port,  
instead of through the “PC” port. Warning: this configuration is less secure than  
default option. Default is No.  
Cloned WAN MAC This allows you to change/set the MAC address on the WAN interface.  
Addr  
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 Value is set in units of hours. Default value is 120 hrs (5 Days.) The time IP  
Time  
address is assigned to the LAN clients.  
DMZ IP  
Forward all WAN IP traffic to a specific IP address if no matching port is used by  
HT–502 or defined in port forwarding.  
Port Forwarding  
Forwards 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.  
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TABLE 8: HT–502 DEVICE STATUS PAGE DEFINITIONS  
MAC Address  
The device ID in HEX format. This is needed for ISP troubleshooting. Note there  
are separate MAC addresses for the WAN side and the LAN side.  
WAN IP Address  
Product Model  
Shows WAN IP address of HT–502.  
Contains the product model info.  
Software Version  
Program: This is the main software release. Boot and Loader are seldom  
changed.  
System Up Time  
PPPoE Link Up  
Shows system up time since the last reboot.  
Indicates whether the PPPoE connection is up if the HT–502 is connected to DSL  
modem.  
NAT  
Indicates the type of NAT connection used by the HT–502 via its WAN port.  
Based on STUN protocol.  
Port Status  
Shows several information regarding the individual FXS ports.  
Port  
Hook  
Registration  
DND  
Forward  
Busy  
Delayed  
Forward Forward  
FXS1 On Hook  
FXS2 On Hook  
Registered  
Registered  
Yes  
No  
613  
614  
Both FXS port1 and FXS port2 are registered with this SIP Server.  
FXS Port 1 user has set Do Not Disturb.  
FXS Port 1 user has set his calls to be forwarded unconditionally to ext 613.  
FXS Port 2 user has set his calls to forward to 614 when his phone is busy.  
ADVANCED USER CONFIGURATION  
Log in to the advanced user configuration page the same way as for the basic configuration page. The  
password is case sensitive and the factory default password for Advanced User is “admin”.  
Advanced User configuration includes the end user configuration and the advanced configurations  
including: a) SIP configuration, b) Codec selection, c) NAT Traversal Setting and d) other miscellaneous  
configuration. HT-502 each FXS SIP account has its own configuration page. Their configurations are  
identical.  
TABLE 9: HT–502 ADVANCED CONFIGURATION PAGE DEFINITIONS  
Admin Password  
This contains the password to access the Advanced Web Configuration page.  
This field is case sensitive. Only the administrator can configure the “Advanced  
Settings” page. Password field is purposely left blank for security reasons after  
clicking update and saved. The maximum password length is 25 characters.  
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Layer 3 QoS  
This field defines the layer 3 QoS parameter which can be the value used for IP  
Precedence or Diff-Serv or MPLS. Default value is 48.  
Layer 2 QoS  
Value used for layer 2 VLAN tag. Default setting is blank.  
STUN Server  
IP address or Domain name of the STUN server.  
Keep-alive interval  
This parameter specifies how often the HT–502 sends a blank UDP packet to the  
SIP server in order to keep the “hole” on the NAT open. Default is 20 seconds.  
Minimum value is 20 seconds.  
Firmware Upgrade  
and Provisioning  
Enables HT–502 to download firmware or configuration file through either the  
TFTP or HTTP server.  
Via TFTP Server  
This is the IP address of the configured TFTP server. If selected and it is non-  
zero or not blank, the HT–502 retrieves the new configuration file or new code  
image from the specified TFTP server at boot time. After 5 attempts, the system  
will timeout and 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 is saved into the Flash memory.  
Note: Please do NOT interrupt the TFTP upgrade process (especially the power  
supply) as this will damage the device. Depending on the network environment  
this process can take up to 15 or 20 minutes.  
Via HTTP Server  
The URL for the HTTP server used for firmware upgrade and configuration via  
HTTP.  
For example, ttp://provisioning.mycompany.com:6688/Grandstream/1.0.0.34  
:6688” is the specific TCP port where the HTTP server is listening; it can be  
omitted if using default port 80.  
Note: If Auto Upgrade is set to No, HT–502 will only do HTTP download once at  
boot up.  
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, HT–502 will request 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  
Config File Postfix  
Default is blank. End user should keep it blank.  
Default is blank. End user should keep it blank.  
Automatic Upgrade Choose “Yes” to enable automatic upgrade and provisioning. In “Check for new  
firmware every” field, enter the number of days to enable HT-502 to check the  
server for firmware upgrade or configuration in the defined period of days. When  
set to No, HT-502 will only do upgrade once at boot up. “Always check for New  
Firmware.” Check New Firmware only when F/W pre/suffix changes”  
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Firmware Key  
Used for firmware encryption. Should be 32 digit in hexadecimal representation.  
End user should keep it blank.  
Authenticate Conf  
File  
If set to Yes, config file is authenticated before acceptance. This protects the  
configuration from an unauthorized change.  
Firmware Key  
Used for firmware encryption. Should be 32 digit in hexadecimal representation.  
End user should keep it blank.  
Lock Keypad  
Update  
If set to “Yes”, the configuration update via keypad is disabled.  
Disable Voice  
Prompt  
Default is No  
Disable Direct IP-IP Default is No.  
Calling  
NTP server  
URI or IP address of the NTP (Network Time Protocol) server. Used by the  
phone to synchronize the date and time.  
Syslog Server  
Syslog Level  
The IP address or URL of System log server. This feature is especially useful for  
the ITSP (Internet Telephone Service Provider)  
Select the HT–502 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:  
1.  
2.  
3.  
4.  
5.  
6.  
7.  
8.  
9.  
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)  
10. 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  
Example: May 19 02:40:38 192.168.1.14 GS_LOG: [00:0b:82:00:a1:be][000]  
Ethernet link is up  
TABLE 10: HT–502 INDIVIDUAL ACCOUNT SETTINGS DEFINITIONS  
Profile Active  
SIP Server  
When set to Yes the FXS port is activated.  
SIP Server’s IP address or Domain name provided by VoIP service provider.  
Outbound Proxy  
IP address or Domain name of Outbound Proxy, or Media Gateway, or  
Session Border Controller. Used by HT–502 for firewall or NAT penetration in  
different network environments. If symmetric NAT is detected, STUN will not  
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work and ONLY outbound proxy can correct the problem.  
User can select UDP or TCP or TLS.  
SIP transport  
NAT Traversal  
(STUN)  
This parameter defines whether or not the HT–502 NAT traversal mechanism  
is activated. If activated (by choosing “Yes”) and a STUN server is also  
specified, then the HT–502 performs according to the STUN client  
specification. Using this mode, the embedded STUN client will detect if and  
what type of firewall/NAT is being used. If the detected NAT is a Full Cone,  
Restricted Cone, or a Port-Restricted Cone, the HT–502 will use its mapped  
public IP address and port in all of its SIP and SDP messages.  
If the NAT Traversal field is set to “Yes” with no specified STUN server, the  
HT–502 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.  
SIP User ID  
User account information, provided by VoIP service provider (ITSP). Usually in  
the form of digit similar to phone number or actually a phone number.  
SIP service subscriber’s Authenticate ID used for authentication. Can be  
identical to or different from SIP User ID.  
Authenticate ID  
Authenticate  
Password  
SIP service subscriber’s account password.  
Name  
SIP service subscriber’s name for Caller ID display.  
Use DNS SRV  
Default is No. If set to “Yes” the client will use DNS SRV to look up server.  
User ID is Phone  
Number  
If the HT–502 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  
Controls whether the HT–502 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.  
Outgoing Call w/o  
Registration  
Default is No. If set to “Yes,” user can place outgoing calls even when not  
registered (if allowed by ITSP) but is unable to receive incoming calls.  
Register Expiration  
This parameter allows the user to specify the time frequency (in minutes) the  
HT–502 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  
Defines the local SIP port the HT–502 will listen and transmit. The default  
value for FXS port 1 is 5060. The default value for FXS port 2 is 5062.  
Defines the local RTP-RTCP port pair the HT–502 will listen and transmit. It is  
the base RTP port for channel 0. When configured,  
channel 0 uses this port _value for RTP and the port_value+1 for its RTCP;  
channel 1 uses 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  
5012.  
Use Random Port  
This parameter forces the random generation of both the local SIP and RTP  
ports when set to Yes. This is usually necessary when multiple HT–502 are  
behind the same NAT.  
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Refer to Use Target  
Contact  
Default is NO. If set to YES, then for Attended Transfer, the “Refer-To” header  
uses the transferred target’s Contact header information.  
DTMF Payload Type  
DTMF in-audio  
Sets the payload type for DTMF using RFC2833.  
Send DTMF as inband (in-audio).  
DTMF via RFC2833  
DTMF via SIP INFO  
Send DTMF via RTP (According to RFC 2833).  
Send DTMF via SIP INFO message.  
Send Flash Event  
Default is No. If set to yes, flash will be sent as DTMF event.  
Enable Call Features Default is Yes. (If Yes, call features using star codes will be supported locally)  
Offhook Auto-Dial  
This parameter allows users to configure a User ID or extension number to be  
automatically dialed upon off-hook. Only the user part of a SIP address needs  
is entered here. The HT–502 will automatically append the “@” and the host  
portion of the corresponding SIP address.  
Proxy-Require  
Use NAT IP  
SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.  
NAT IP address used in SIP/SDP message. Default is blank.  
Distinctive Ring Tone Custom Ring Tone 1 to 3 with associate Caller ID: when selected, if Caller ID  
is configured, then the device will ONLY uses this ring tone when the incoming  
call is from the Caller ID. System Ring Tone is used for all other calls. When  
selected but no Caller ID is configured, the selected ring tone will be used for  
all incoming calls.  
Disable Call Waiting  
Default is No.  
Disable Call Waiting  
Tone  
Default is No. This is to disable the stutter Call Waiting Tone when a Call  
Waiting call arrives. The CWCID will still be displayed.  
Ring Timeout  
Incoming call will stop ringing when not picked up given a specific period of  
time.  
No Key Entry  
Timeout  
Default is 4 seconds.  
Early Dial  
Default is No. Use only if proxy supports 484 response. This parameter  
controls whether the phone will 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 likely be rejected by the  
proxy (with a 404 Not Found error).  
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 Dial Key  
Allows users to configure the “#” key as the “Send” (or “Dial”) key. If set to  
“Yes”, “#” will send the number. In this case, this key is essentially equivalent  
to the “Dial” key. If set to “No”, this “#” key can be included as part of number.  
Dial Plan  
Dial Plan Rules:-  
1. Accept Digits: 1,2,3,4,5,6,7,8,9,0  
2. Grammar: x - any digit from 0-9;  
a. xx+ - at least 2 digit number;  
b. ^ - exclude;  
c. [3-5] - any digit of 3, 4, or 5;  
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d. [147] - any digit 1, 4, or 7;  
e. <2=011> - replace digit 2 with 011 when dialing  
Example 1: {[369]11 | 1617xxxxxxx} –  
Allow 311, 611, 911, and any 10 digit numbers of leading digits 1617  
Example 2: {^1900x+ | <=1617>xxxxxxx} –  
Block any number of leading digits 1900 and add prefix 1617 for any  
dialed 7 digit numbers  
Example 3: {1xxx[2-9]xxxxxx | <2=011>x+} –  
Allow any length of number with leading digit 2 and 10 digit-numbers of  
leading digit 1 and leading exchange number between 2 and 9; If leading  
digit is 2, replace leading digit 2 with 011 before dialing  
3. Default: Outgoing - {x+}  
Subscribe for MWI  
Send Anonymous  
Default is No. When set to “Yes” a SUBSCRIBE for Message Waiting  
Indication will be sent periodically.  
If this parameter is set to “Yes”, the “From” header along with Privacy and P_  
Asserted_Identity headers in outgoing INVITE message will be set to  
anonymous, blocking Caller ID.  
Anonymous Call  
Rejection  
Default is No. If set to Yes, incoming calls with anonymous Caller ID will be  
rejected with 486 Busy message.  
Special Feature  
Default is Standard. Choose the selection to meet some special requirements  
from Softswitch vendors.  
Preferred Vocoder  
The HT-502 supports up to 5 different Vocoder types including G.711 A-/U-  
law, G.726 (Supports bit rates 16, 24, 32 and 40), G.723.1, G.729A/B/E and  
iLBC. The user can configure Vocoders in a preference list that will be  
included with the same preference order in SDP message. The first Vocoder  
is entered by choosing the appropriate option in “Choice 1”. The last Vocoder  
is entered by choosing the appropriate option in “Choice 8”.  
G723 Rate  
Defines the encoding rate for G.723 vocoder. By default, 6.3kbps rate is  
chosen.  
iLBC Frame Size  
Sets the iLBC frame size in 20ms or 30ms  
iLBC Payload type  
Defines payload type for iLBC. Default value is 97. The valid range is between  
96 and 127.  
G726-16 Payload  
type  
Default value is 98. Range is from 96 to 127.  
Default value is 99. Range is from 96 to 127.  
Default value is 100. Range is from 96 to 127.  
Default value is 103. Range is from 96 to 127.  
Default value is 102. Range is from 96 to 127.  
G726 - 24 Payload  
type  
G726 - 32 Payload  
type  
G726 - 40 Payload  
type  
G729E payload type  
VAD  
Default is No. VAD allows detecting the absence of audio and conserve  
bandwidth by preventing the transmission of "silent packets" over the network.  
Symmetric RTP  
Fax Mode  
Default is No. When set to Yes the device will change the destination to send  
RTP packets to the source IP address and port of the inbound RTP packet last  
received by the device.  
T.38 (Auto Detect) FoIP by default, or Pass-Through (must use codec  
PCMU/PCMA)  
Fax Tone Detection  
Default is Callee. This decides whether Caller or Callee sends out the re-  
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Mode  
INVITE for T.38 or Fax Pass Through.  
Jitter Buffer Type  
Jitter Buffer Length  
SLIC Setting  
Select either Fixed or Adaptive based on network conditions.  
Select Low, Medium or High based on network conditions.  
Dependent on standard phone type (and location)  
Caller ID Scheme  
• Bellcore/Telcordia  
• ETSI-FSK  
• ETSI-DTMF  
• SIN 227 - BT  
• NTT Japan  
Polarity Reversal  
Default is No. If set to “Yes”, polarity will be reversed upon call establishment  
and termination.  
Loop Current  
Disconnect  
Set it to Yes if the traditional PBX you are using with HT502 uses this method  
for signaling call termination. Default is No.  
Hook Flash Timing  
Time period when the cradle is pressed (Hook Flash) to simulate FLASH. To  
prevent unwanted activation of the Flash/Hold and automatic phone ring-back,  
adjust this time value.  
Gain  
Handset volume adjustment.  
RX is for receiving volume,  
TX is for transmission volume.  
Default values are 0dB for both parameters. Loudest volume: +6dB Lowest  
volume: -6dB.  
Call Progress/ Ring  
Tones  
Configure ring or tone frequencies according to preference. By default tones  
are set to North American frequencies. Frequencies should be configured with  
known values to avoid uncomfortable high pitch sounds.  
SAVING THE CONFIGURATION CHANGES  
Click the “Update” button in the Configuration page to save the changes to the HT–502 configuration. The  
following screen confirms that the changes are saved. Reboot or power cycle the HT–502 to make the  
changes take effect.  
FIGURE 4: SCREENSHOT OF SAVE CONFIGURATION PAGE  
REBOOTING THE HT–502 FROM REMOTE  
Remotely reboot the HT–502 by clicking the “Reboot” button at the bottom of the configuration page.  
When finished, re-login to the HT–502 after waiting for about 30 seconds.  
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CONFIGURATION THROUGH A CENTRAL SERVER  
Grandstream HT-502 can be automatically configured from a central provisioning system.  
When HT-502 boot up, it will send TFTP or HTTP request to download configuration file,  
“cfg000b82xxxxxx”, where “000b82xxxxxx” is the LAN side MAC address of the HT-502  
Download the configuration files via TFTP or HTTP from the central server. A service provider or an  
enterprise with large deployment of HT-502 can easily manage the configuration and service provisioning  
of individual devices remotely from a central server.  
GAPS – GRANDSTREAM AUTOMATED PROVISIONING SYSTEM  
Grandstream provides a licensed provisioning system called GAPS (Grandstream Automated Provisioning  
System) that is used to support automated configuration of HT-502. GAPS uses enhanced (NAT friendly)  
TFTP or HTTP (thus no NAT issues) and other communication protocols to communicate with each  
individual HT-502 for firmware upgrade, remote reboot, etc.  
Grandstream provides GAPS service to VoIP service providers. Use GAPS for either simple redirection or  
with certain special provisioning settings. At boot-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  
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SOFTWARE UPGRADE  
Software upgrade can be done via either TFTP or HTTP. The corresponding configuration settings are in  
the ADVANCED SETTINGS configuration page.  
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/1.0.0.39  
e.g. 168.75.215.189  
NOTES:  
TFTP server in IP address format can be configured via IVR. Please refer to section  
CONFIGURATION GUIDE for instructions. If TFTP server is in FQDN format, it must be set via  
web configuration interface.  
End users recommended using our TFTP server. Its address can be found at  
http://www.grandstream.com/firmware.html . Currently, the TFTP server, your HT502 can be  
upgraded from has an IP address 168.75.215.189. For companies, we recommend to maintain  
their own TFTP/ HTTP server for upgrade and provisioning procedures.  
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 HT ATA will  
attempt to retrieve the new image files by downloading them into the HT-502’s SRAM. During this  
stage, the HT-502’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 HT-502 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  
http://support.solarwinds.net/updates/New-customerFree.cfm. Our latest official release can be  
Directions to Download TFTP Server:  
1. Unzip the file and put all of them under the root directory of the TFTP server.  
2. Put the PC running the TFTP server and the HT–502 device in the same LAN segment.  
3. Please go to File -> Configure -> Security to change the TFTP server's default setting from  
"Receive Only" to "Transmit Only" for the firmware upgrade.  
4. Start the TFTP server, in the phone’s web configuration page  
5. Configure the Firmware Server Path with the IP address of the PC  
6. Update the change and reboot the unit  
web server.  
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CONFIGURATION FILE DOWNLOAD  
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 LAN side MAC address of the device, i.e.,  
“cfg000b820102ab”. The configuration file name should be in lower cases.  
FIRMWARE AND CONFIGURATION FILE PREFIX AND POSTFIX  
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.  
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.  
AutomaticUpgrade:  
No  
Check every day  
Check every week  
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RESTORE FACTORY DEFAULT SETTING  
WARNING! Restoring the Factory Default Setting will DELETE all configuration information of the phone.  
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 VoIP service  
provider.  
There are two ways to reset the device.  
RESET VIA THE RESET BUTTON  
1. Locate a needle-sized hole on the back panel of the HT–502 unit next to the power connection.  
2. Insert a pin in this hole, and press for about 7 seconds. The back LEDs for LAN and WAN will  
be solid on to indicate the reset.  
3. Take out the pin. All unit settings are restored to factory settings.  
RESET VIA IVR  
1. Find the MAC address of the device. It is a 12 digits HEX number located on the bottom of  
the unit.  
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”.  
3. To perform factory reset:  
-
-
-
-
Pick up the headset and dial “***” for voice prompt.  
Enter “99” and get the voice prompt “Reset”.  
Enter the encoded MAC address of the device.  
Wait for 15 seconds. The device will reboot automatically and restore to factory  
default setting.  
NOTE:  
1. Factory Reset will be disabled if the “Lock keypad update” is set to “Yes”.  
2. Please be aware by default the HT-502 WAN side HTTP access is disabled. After a factory reset, the  
device’s web configuration page can be accessed only from its LAN port.  
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GLOSSARY OF TERMS  
ADSL 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.  
AGC 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.  
ARP Address Resolution Protocol is a protocol used by the Internet Protocol (IP) [RFC826], specifically  
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, generate 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 sub-sampling.  
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 multi-carrier time division multiple access, time division duplex (MC-  
TDMA-TDD) radio transmission technique using ten radio frequency channels from 1880 to 1930 MHz,  
each 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 multi-frame (16 frames). Voice signals are  
digitally encoded into a 32 Kbit/s signal using Adaptive Differential Pulse Code Modulation.  
DNS Short for Domain Name System (or Service or Server), an Internet service that translates domain  
names into IP addresses  
DID 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 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 #).  
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FQDN 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 and.com is the top level domain.  
FXS Foreign eXchange Office. An FXS 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 FXS interface will accept calls from FXS or PSTN interfaces. All countries and regions have  
their own standards.  
FXS is complimentary to FXS (and the PSTN).  
FXS Foreign eXchange Station. An FXS device has hardware to generate the ring signal to the FXS  
extension (usually an analog phone).  
An FXS device will allow any FXS 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 FXS devices (by an FXS 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 A suite of standards for multimedia conferences on traditional packet-switched networks.  
HTTP 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 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.  
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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 Network Address Translation  
NTP 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 Caller. The OBP/SBC acts  
as if it was the called VoIP phone and places a second call to the called Caller. The effect of this behavior  
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 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.  
PSTN Public Switched Telephone Network. The phone service we use for every ordinary phone call, or  
called POT (Plain Old Telephone), or circuit switched network.  
RTCP 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.  
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 Session Description Protocol is a format for describing streaming media initialization parameters. It  
has been published by the IETF as RFC 2327.  
SIP 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 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 well  
with non-symmetric NAT routers.  
TCP 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 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.  
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UDP 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 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 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.  
VoIP Voice over the Internet. VoIP encompasses many protocols. All the protocols do some form of  
signaling of call capabilities and transport of voice data from one point to another. e.g.: SIP, H.323, etc.  
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