Grandstream Networks HT-503 User manual

HT503 User Manual www.grandstream.com
Firmware Version 1.0.1.57 [e-mail]
Grandstream Networks, Inc.
HT503
FXS/FXO Port
Analog Telephone Adaptor

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TABLE OF CONTENTS
HT503 USER MANUAL
WELCOME.......................................................................................................................................................4
Safety Compliances ................................................................................................................ 4
Warranty ................................................................................................................................. 4
CONNECT YOUR HT503...................................................................................................................................5
Equipment Packaging ................................................................................................................. 5
Connecting the HT503................................................................................................................ 5
PRODUCT OVERVIEW.......................................................................................................................................7
Software Features Overview....................................................................................................... 7
Hardware Specification............................................................................................................... 8
BASIC OPERATIONS ........................................................................................................................................9
Understanding HT503 Voice Prompt ......................................................................................... 9
Placing a Phone Call................................................................................................................. 10
Phone or Extension Numbers................................................................................................ 10
Direct IP Calls....................................................................................................................... 10
Call Hold................................................................................................................................... 11
Call Waiting.............................................................................................................................. 11
Call Transfer ............................................................................................................................. 11
Blind Transfer....................................................................................................................... 11
Attended Transfer ................................................................................................................. 12
3-way Conferencing.................................................................................................................. 12
PSTN Pass Through.................................................................................................................. 12
VoIP-to-PSTN Calls ................................................................................................................. 13
PSTN-to-VoIP Calls ................................................................................................................. 13
Route Calls to PSTN................................................................................................................. 14
Forward Calls to PSTN............................................................................................................. 14
Forward Calls to VoIP.............................................................................................................. 15
One Stage Dialing..................................................................................................................... 15
Fax Support............................................................................................................................... 15
CALL FEATURES ...........................................................................................................................................16
CONFIGURATION GUIDE.................................................................................................................................17
Configuring HT503 through Voice Prompt.............................................................................. 17
Configuring HT503 with Web Browser ................................................................................... 18
Access the Web Configuration Menu................................................................................... 18
Saving the Configuration Changes ........................................................................................... 33
Rebooting from Remote............................................................................................................ 34
Configuration through a Central Server.................................................................................... 34
SOFTWARE UPGRADE ...................................................................................................................................35
Firmware Upgrade through TFTP/HTTP/HTTPS.................................................................... 35
Configuration File Download................................................................................................... 36
Firmware and Configuration File Prefix and Postfix................................................................ 36
Managing Firmware and Configuration File Download........................................................... 36
RESTORE FACTORY DEFAULT SETTING..........................................................................................................37

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TABLE OF FIGURES
HT503 USER MANUAL
FIGURE 1: CONNECTING THE HT503 ...............................................................................................................5
FIGURE 2: INTERCONNECTION DIAGRAM OF THE HT503 ...................................................................................6
TABLE OF TABLES
HT503 USER MANUAL
Table 1: Definitions of the HT503 Connectors................................................................................6
Table 2: HT503 LED Definitions .....................................................................................................6
Table 3: HT503 Technical Specifications........................................................................................7
Table 4: HT503 Hardware Specification.........................................................................................8
Table 5: HT503 IVR Menu Definitions ............................................................................................9
Table 6: HT503 Call Feature Definitions.......................................................................................16
Table 7: Status Page.....................................................................................................................19
Table 8: Basic Settings .................................................................................................................20
Table 9: Advanced Settings..........................................................................................................22
Table 10: FXS PORT Settings......................................................................................................24
Table 11: FXO PORT Settings......................................................................................................28
TABLE OF GUI INTERFACES
HT503 USER MANUAL
(http://www.grandstream.com/support/ht_series/ht503/documents/ht503_gui.zip)
1. SCREENSHOT OF CONFIGURATION LOGIN PAGE
2. SCREENSHOT OF STATUS PAGE
3. SCREENSHOT OF BASIC SETTINGS CONFIGURATION PAGE
4. SCREENSHOT OF ADVANCED SETTINGS CONFIGURATION PAGE
5. SCREENSHOT OF FXS ACCOUNT CONFIGURATION
6. SCREENSHOT OF FXO ACCOUNT CONFIGURATION
7. SCREENSHOT OF CALL PROGRESS TONES CONFIGURATION PAGE
8. SCREENSHOT OF SAVED CONFIGURATION CHANGES
9. SCREENSHOT OF REBOOT PAGE

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WELCOME
Thank you for purchasing Grandstream’s HT503, the affordable, feature rich, Analog Telephone
Adaptor/IAD. The HT503 combines a sleek design with the latest technology to offer more advanced
telephony features and significantly better integrated router performance than its predecessor – the
HT488. It is the second ATA/IAD in the HandyTone 50x series. The HT503 functions as a true 3-in-1
gateway for PSTN network, analog telephone FXS interface and IP network. It enables remote call
origination and termination from/to PSTN and supports the feature of “hop-on/hop-off” calling.
This manual will help you learn how to operate and manage your HT503 Analog Telephone Adaptor/IAD
and make the best use of its many upgraded features including simple and quick installation, 3-way
conferencing, and remote call origination and “hop-on/hop-off” calling using the programmable PSTN
FXO port. This HT503 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 remote user.
This document is subject to changes without notice. The latest electronic version of this user manual can
be downloaded from the following location:
http://www.grandstream.com/support/ht_series/ht503/documents/ht503_usermanual_english.pdf
Safety Compliances
The HT503 adaptor complies with FCC/CE and various safety standards. The HT503 power adaptor is
compliant with UL standard. Only use the universal power adapter provided with the HT503 package.
The manufacturer’s warranty does not cover damages to the phone caused by unsupported power
adaptors.
Warranty
If you purchased your HT503 from a reseller, please contact them for replacement, repair or refund. If
you purchased the product directly from Grandstream, contact your Grandstream Sales and Service
Representative for an 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 avoid your manufacturer
warranty.
•This document contains links to Grandstream GUI Interfaces. Please remember to download these
examples from http://www.grandstream.com/support/ht_series/ht503/documents/ht503_gui.zip for
your reference.
•This document is subject to change without notice. The latest electronic version of this user manual is
available for download from the following location:
http://www.grandstream.com/support/ht_series/ht503/documents/ht503_usermanual_english.pdf
•Reproduction or transmittal of the entire or any part, in any form or by any means, electronic or print,
for any purpose without the express written permission of Grandstream Networks, Inc. is not
permitted.

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CONNECT YOUR HT503
Equipment Packaging
The HT503 ATA package contains:
•One HT503 Main Case
•One Universal Power Adaptor
•One Ethernet Cable
•One HT503 Vertical Stand
Connecting the HT503
The HT503 is designed for easy configuration and easy installation. Configure the HT503 following the
directions in the Configuration section of this manual.
1. Connect a standard touch-tone analog telephone to the PHONE port.
2. Insert a standard RJ11 telephone cable into the LINE port and connect the other end of the
telephone cable to a wall jack.
3. Insert the Ethernet cable into the WAN port of HT503 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 HT503 if it is being used as a router.
5. Insert the power adapter into the HT503 and connect it to a wall outlet.
The HT503 Analog Telephone Adaptor is an all-in-one VoIP integrated device designed to be a total
solution for networks providing VoIP services. The HT503 VoIP features and functions are available
using a regular analog telephone.
FIGURE 1: CONNECTING THE HT503
The HT503 has one FXS port and one FXO port. The PHONE port next to the power supply is an FXS
port. The LINE port on the back right of the HT503 is an FXO port. Both the FXS port and the FXO port
can have a separate SIP account. This is a key feature of HT503 as it supports simultaneous calls on
both the FXS port and FXO port. Telephone calls can be originated from or terminated on the PSTN
network remotely via the FXO port.
HT503
Front View
HT503
Back View
Display LEDs
(Green)
RJ-45 Ports
10/100 Mbps Reset Power
Supply
(12V)
RJ11
FXS Port RJ11 FXO Port

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TABLE 1: DEFINITIONS OF THE HT503 CONNECTORS
12VDC, 0.5A Power adapter connection
LAN Port (RJ-45) Connect the LAN port with an Ethernet cable to your PC.
WAN Port (RJ-45) Connect the WAN port to the internal LAN network or router.
PHONE (RJ-11) FXS port to be connected to analog phones / fax machines.
LINE (RJ-11) FXO port should be connected to the PSTN line
TABLE 2: HT503 LED DEFINITIONS
LEDs
POWER LED Indicates Power. Remains ON when power is connected
WAN LED Indicates LAN (or WAN) port activity
LAN LED Indicates PC (or LAN) port activity
PHONE/ LINE LED
Indicates the status of the FXS and FXO ports on the back
panel.
Busy – ON (Solid Green) Available – OFF
Slow blinking FXS LEDs indicates voicemail for that port.
Note: Slow blinking of POWER, WAN, and LAN LEDs together indicate firmware upgrade/provisioning state.
FIGURE 2: INTERCONNECTION DIAGRAM OF THE HT503
Internet ADSL/Cable
Modem Ethernet
Fax
Cordless
Analog Phone
WAN
LAN
FXSFXO
PSTN
Clou
d

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PRODUCT OVERVIEW
The HT503 is an affordable, high-quality, integrated IP telephony solution for both the residential
customers and the ‘road-warriors’ who need advanced call features between traditional PSTN network
and IP network. The HT503 enables IP connectivity for any phone or fax using the FXS port and a web-
based GUI for easy configuration and installation. It functions as a true FXO gateway that enables remote
call origination and termination from/to PSTN and supports the feature of “hop-on/hop-off” using the
programmable FXO port.
Software Features Overview
The HT503 features 2 SIP account profiles and supports advanced telephony features including caller ID,
call waiting, call transfer, 3-way conferencing (with either IP or PSTN calls), and multi-language voice
prompts. From a technical standpoint, the HT503 offers a power-outage survivable life line and internet-
disconnect survivable fail-over-to-PSTN support, dual 10/100Mbps Ethernet ports with integrated high-
performance NAT router, a flexible dial plan and a broad range of popular voice codecs.
TABLE 3: HT503 TECHNICAL SPECIFICATIONS
Interfaces 1 FXS telephone port (RJ11), 1 FXO PSTN line port (RJ11) with lifeline support
Two (2) 10M/100 Mbps ports (RJ45) with integrated Nat router
Protocol Support TCP/UDP/IP, RTP/RTCP, HTTP/HTTPS, ARP/RARP, ICMP, DNS, DHCP, NTP, TFTP,
PPPoE, STUN & TELNET protocols
LED Indicators Power, WAN, LAN, PHONE, and LINE
RESET Button Factory Reset Button
Device Management Web interface or via secure (AES encrypted) central configuration file for mass
deployment
Support device configuration via built-in IVR, Web browser or central configuration file
through TFTP, HTTP or HTTPS
Support Layer 2 (802.1Q, VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS)
Auto/manual provisioning system
NAT-friendly remote software upgrade (via TFTP/HTTP/HTTPS) for deployed devices
including behind firewall/NAT
Syslog support
DHCP Server/Client Yes
Audio Features Advanced Digital Signal Processing (DSP)
Dynamic negotiation of codec and voice payload length
Support for G.723, G.729/E, G.711, G.726-40/32/24/16, iLBC, T.38 codecs
In-band and out-of-band DTMF (in audio, RFC2833, SIP INFO)
Silence Suppression, VAD (voice activity detection), CNG (comfort noise generation),
ANG (automatic gain control)
Adaptive jitter buffer control
Packet delay & loss concealment (PLC) & G.168 compliant Line Echo Cancellation
Support volume amplification
Support configurable Call Progress Tones
Call Handling Features Caller ID display or block, Call waiting caller ID, Call waiting/flash, Call transfer, hold,
call forward, do not disturb, 3-way conferencing
Network and
Provisioning Manual or dynamic host configuration protocol (DHCP) network setup; RTP and NAT
support traversal via STUN
Fax over IP T.38 compliant Group 3 Fax Relay up to 14.4kpbs and auto-switch to G.711 for Fax

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Pass-through (pending), Fax Data pump V.17, V.19, V.27ter, V.29 for T.38 fax relay
Security DIGEST authentication and encryption using MD5 and MD5-sess
Physical Design Stylish and compact design; small universal power supply, ideal for travel
Hardware Specification
The table below lists the hardware specification of HT503.
TABLE 4: HT503 HARDWARE SPECIFICATION
LAN interface 1xRJ45 10/100 Mbps Port
WAN interface 1xRJ45 10/100 Mbps Port
FXS telephone port 1 x FXS (RJ11)
FXO telephone port (PSTN Port) 1x PSTN pass-through and life line port
LED Power, WAN, LAN, PHONE, and LINE (Green)
Universal Switching
Power Adaptor Input: 100–240 VAC, 50-60 Hz
Output: 12VDC, 0.5A, UL certified
Dimension 25mm x 115mm x 75mm (when laying flat);
115mm x 25mm x 75mm (standing up)
Weight Approximately 0.6lbs (0.3kg)
Temperature Operational: 32° - 104°F or 5° – 45°C
Storage: 10°–130°F
Humidity 10% - 90%
(non-condensing)
Compliance

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BASIC OPERATIONS
Understanding HT503 Voice Prompt
HT503 has a built-in voice prompt menu for simple device configuration. The voice prompt menu is
designed for the FXS port only. To enter the voice prompt menu, press *** from the analog phone
connected to the FXS port.
TABLE 5: HT503 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 “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.
02 “IP Address “ + IP address The current WAN IP address is announced
If using “Static IP Mode”, enter 12 digit new IP address.
03 “Subnet “ + IP address Same as menu 02
04 “Gateway “ + IP address Same as menu 02
05 “DNS Server “ + IP address Same as menu 02
07 Preferred Vocoder Press “9” to move to the next selection in the list:
•PCM U / PCM A
•iLBC
•G.726
•G.723
•G.729
10 MAC Address Announces the MAC address.
12 WAN Port Web Access Press “9” to toggle between enable / disable
13 Firmware Server IP Address Announces current Firmware Server IP address. Enter 12 digit new
IP address.
14 Configuration Server IP
Address Announces current Config Server Path IP address. Enter 12 digit
new IP address.
15 Upgrade Protocol Upgrade protocol for firmware and configuration update. Press “9”
to toggle between TFTP / HTTP / HTTPS
16 Firmware Version Firmware version information.
17 Firmware Upgrade Firmware upgrade mode. Press “9” to toggle among the following
three options:
- always check
- check when pre/suffix changes
- never upgrade
47 “Direct IP Calling” Enter the IP address to make a direct IP call, after dial tone. (See
“Make a Direct IP Call”.)
86 Voice Mail Number of voice mails
99 “RESET” Press “9” to reboot the device; or
Enter encoded MAC address to restore factory default setting (See
“Restoring Factory Settings”)

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“Invalid Entry” Automatically returns to main menu
NOTE:
•“*” shifts down to the next menu option
•“#” returns to the main menu
•“9” functions as the ENTER key in many cases to confirm an option
•All entered digit sequences have known lengths - 2 digits for menu option. For IP address, the "*"
key represent the dot "." (Like 192.168.0.26 should be key in like 192*168*0*26). 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
Placing a Phone Call
Phone or Extension Numbers
There are currently two methods to make an extension number call:
a) Dial the numbers directly and wait for 4 (default) seconds.
b) Dial the numbers directly, and press # (assuming that “use # as dial key” is selected in the web
configuration).
EXAMPLES:
•To dial another extension on the same proxy, such as 1008, simply pick up the attached phone,
dial 1008 and then press the # or wait for 4 seconds.
•To dial a PSTN number such as 6266667890, you may need a prefix number followed by the
phone number. Please check with your VoIP service provider for this information. If your phone is
assigned a PSTN-like number such as 6265556789, you will most likely follow the rule 1 + (the
number) – 16266667890. Press # or wait for 4 seconds.
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:
•Both HT503 and other VoIP Device, have public IP addresses, or
•Both HT503 and other VoIP Device are on the same LAN using private IP addresses, or
•Both HT503 and other VoIP Device can be connected through a router using public or private IP
addresses (with necessary port forwarding or DMZ).
HT503 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 using format ex. 192*168*0*160 after the dial tone.
Using Star Code
1. Pick up the analog phone then dial “*47”

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2. Enter the target IP address using same format as above.
Note: NO dial tone will be played between step 1 and 2.
Destination ports can be specified by 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 can not make
direct IP calls between FXS1 to FXS2 since they are using same IP.
Call Hold
This function is applicable on the FXS port for VoIP calls only. While in conversation, pressing the “flash”
button on the connected phone (if the phone has that button) places the remote end on hold. Pressing the
“flash” button again releases the previously held party and the conversation can resume. If no “flash”
button is available, then on-off hook quickly (hook flash) will do the same thing. You may lose the call if
‘hook flash’ is not quick enough.
Call Waiting
This function is applicable on FXS port for VoIP calls only. If the call waiting feature is enabled, the user
will hear a special stutter tone if there is another call on the line. Press the flash button to place the
current party on hold and switch to the other call. Pressing the flash button toggles between two active
calls. The HT503 also provides CWCID (call waiting caller ID) information which includes caller ID
information in addition to the special stutter tone. The analog phone must support this feature for it to
work on the HT503. Both call waiting functions (call waiting and CWCID) are activated and deactivated
from the configuration pages menu.
Call Transfer
The HT503 supports both blind transfer and attended transfer.
Blind Transfer
This function is applicable using the FXS port for VoIP calls only. Assume that parties Aand Bare in
conversation. Party A wants to Blind Transfer Party B to C:
3. Apresses FLASH on the analog phone to hear the dial tone.
4. Then Adials *87, then dials C’s number, and then presses #
5. A can hang up.
NOTE: “Enable Call Feature” has to be set to “Yes” in web configuration page.

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Three situations can follow the transfer:
1. A quick confirmation tone (temporarily using the call waiting indication tone) followed by a
dialtone. This indicates the transfer was successful (transferee has received a 200 OK from
transfer target). 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 indicates the transfer has failed.
3. Busy tone keeps playing. This means we have failed to receive the second NOTIFY from the
transferee and the call has timed 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 situation occurs. In bad network scenarios, this could also happen,
although the transfer may have been completed successfully.
Attended Transfer
This function is applicable on the FXS port for VoIP calls only. Assume that parties Aand B are in
conversation. Party A wants to Attend Transfer Party B to C:
1. A presses FLASH on the analog phone to get a dial tone;
2. Athen dial C’s number followed by #.
3. If C answers the call, Aand C are in conversation. Then Acan hang up to complete transfer.
4. If C does not answer the call, Acan press “flash” back to talk to B.
NOTE: When Attended Transfer fails and A hangs up, the HT503 will ring user A back again to remind
A that party B is still on the call. Party A can pick up the phone to resume a conversation with party B.
3-way Conferencing
The HT503 supports Bellcore Style 3-way conferencing.
Assume that parties Aand Bare in conversation. Party A(using the HT503) wants to bring C into a 3-
way conference:
1. Apresses FLASH (on the analog phone, or Hook Flash for old model phones) to get a dial
tone.
2. Adials C’s number then # (or wait for 4 seconds).
3. If C answers the call, then Apresses FLASH to bring B, C in the conference.
4. If C does not answer the call, Acan press FLASH back to talk to B.
5. If Apresses FLASH during the conference, C will be dropped out.
6. If Ahangs up, the conference will be terminated for all three parties when configuration
“Transfer on Conference Hangup” is set to “No”. If the configuration is set to “Yes”, Awill
transfer B to C so that B and C can continue the conversation.
PSTN Pass Through
HT503 supports PSTN pass through using the FXS port. The user can place and receive PSTN calls
using analog phone connected to FXS port.

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•To receive PSTN calls, pick up the phone when it rings;
•To complete a PSTN call, press the PSTN access code (*00 is default, or any number configured
in the web configuration) to switch to the PSTN line, listen for a dial tone, then dial the number.
•If the 503 loses power or lost registration with SIP server, device will switch to mode when PSTN
line will be transparently connected directly to phone connected to FXS port. It will function as a
jack, enabling a direct connection to the PSTN Line.
VoIP-to-PSTN Calls
This function is available using the FXO port. The FXO port functions as a bridge between the Internet
and PSTN. The user can remotely use a PSTN line to initiate a call.
TO MAKE A VOIP-TO-PSTN CALL:
1. Dial the FXO SIP account phone number to establish the VoIP session. The caller will hear the
ring back tone once. Then the caller hears either a special continuous tone or a dial tone. The
special continuous tone is played if the pin code is configured, otherwise, the caller will hear a dial
tone.
2. Enter the PIN code (if configured under the BASIC configuration page). The caller will hear a dial
tone and be connected to the PSTN line if the PIN code is valid. If the PIN code is invalid, the
continuous tone is played to prompt caller to enter the PIN code again. The user may try up to 3
times to enter a correct PIN code. After three (3) tries, the HT503 hangs up.
3. After the caller hears a dial tone from PSTN line, the caller can place the next call.
4. The user can hit the # key to identify the end of the pin code or wait 4 seconds for a new dial tone
and then dialing the PSTN number.
Note:
•Users can choose whether or not to apply password protection for VoIP-to-PSTN calls. A PIN
(Pin for PSTN calls) consists of up to 8 numeric digits and can be configured using the BASIC
SETTINGS of the web configuration page. By default, there is no password protection. (I.e. there
is no authentication required for callers on the use of PSTN line through HT503).
•When a PIN is configured for VOIP-to-PSTN call flow, the VoIP device that calls into the HT503
FXO account needs to configure RFC2833 or SIP Info for DTMF digit transmission.
•The special continuous tone is the prompt to enter a valid PIN code. If a caller doesn’t enter a
valid PIN, the HT503 times out after 10 seconds. Users may press the “#” key to indicate the end
of an input or wait 4 seconds.
•On the web configuration page, if the “Forward to PSTN” is configured, the second stage dialing
format is eliminated, so after dialing into the FXO SIP account number, the PSTN number will be
called automatically
PSTN-to-VoIP Calls
This function is available using the FXO port. The FXO port functions as a bridge between the Internet
and PSTN and enables calls to be passed from the PSTN network to VoIP. The user can make VoIP calls
remotely by dialing into the FXO line port on HT503.
To Make a PSTN-to-VoIP Call:
1. Make an incoming call to the PSTN line on FXO port. The phone will ring for 4 times by default
(this setting is configurable on the FXO port configuration page).

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2. If no one answers the call after 4 rings (default configuration), then the caller hears either a
special continuous tone (prompting a PIN number) or a dial tone.
3. Enter a valid PIN (if configured under the BASIC configuration page). The caller will hear dial
tone and be bridged to VoIP. If an incorrect PIN is input, the continuous tone prompts caller to
enter a valid PIN. The caller may try 3 times to enter a valid PIN, if it is invalid the HT503 will
hang up.
4. The caller can dial a VoIP number followed by # (or wait for 4 seconds); the VoIP call will be
initiated from the SIP account configured on the FXO port.
NOTE:
•Users can choose whether or not to apply password protection for VoIP-to-PSTN calls. A PIN
(Pin for PSTN calls) consists of up to 8 numeric digits and can be configured using the BASIC
SETTINGS of the web configuration page. By default, there is no password protection. (I.e. there
is no authentication required for callers on the use of PSTN line through HT503).
•When a PIN is configured for VOIP-to-PSTN call flow, the VoIP device that calls into the HT503
FXO account needs to configure RFC2833 or SIP Info for DTMF digit transmission.
•The special continuous tone is the prompt to enter a valid PIN code. If a caller doesn’t enter a
valid PIN, the HT503 times out after 10 seconds. Users may press the “#” key to indicate the end
of an input or wait 4 seconds.
•On the web configuration page, if the “Forward to VoIP” is configured, the second stage dialing
format is eliminated, so after dialing into the FXO SIP account number, the PSTN number will be
called automatically
Route Calls to PSTN
The FXO port enables access to the PSTN network. By default, the HT503 is in VoIP mode at off-hook.
If “Route Call to PSTN” is configured, certain calls will be initiated from the FXO PSTN line port. This call
feature is especially useful for emergency calls or local telephone calls.
To use this feature, users need to specify a special rule using the dial plan parameter located under FXS
Port configuration page. If the dialed digits match the specified prefix, outbound calls will be initiated from
the PSTN line.
Note: The route to PSTN feature is only applicable to a phone connected to the FXS Port. The
configuration is done using the “dial plan” feature under the FXS tab. An example of the configuration is
{L: 911x+}. This shows that only calls that start with 911 are immediately forwarded to the PSTN line. All
other numbers will not be routed to the PSTN. An normal # would be: {L: 617x+|x+} or {x+| L: 617x+}
For example, if “Route Call to PSTN” is configured as {L: 626x+}, all outgoing calls starting with 626 will
be initiated from the PSTN line.
Forward Calls to PSTN
Any VOIP call may be forwarded to a specified PSTN number. FXO port should be registered with some
preconfigured number (for example 1111). Any VoIP extension can dial this FXO account number and will
be automatically forwarded to preconfigured PSTN extension.
For example, if the end-user has configured a cell phone number in the field “Forward to PSTN” under
BASIC SETTINGS configuration page, all calls will be forwarded to the cell phone number after 4 rings.

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Forward Calls to VoIP
By default, each incoming PSTN call is received over the FXS port. The end-user may forward such a
call to any preconfigured VoIP extension, in case the call is not answered in a certain number of rings.
The Default value of the parameter “Number of Rings” is 4. This parameter located under “FXO Port”
configuration page. If during 4 rings, the incoming from the PSTN call is not answered, the call will be
forwarded to another VoIP number previously configured in the field:”Forward to VoIP”. This parameter
can also be found under BASIC SETTINGS configuration page.
One Stage Dialing
This feature is applicable for VoIP to PSTN calls. Any VoIP extension may dial directly to a local PSTN
number if the one-stage dialing feature is activated. This feature is configured under the FXO
Configuration page and requires SIP Server configuration and support. The special dial plan feature must
be activated in the SIP Server. An outbound call will be sent directly to the assigned FXO port account,
where there the HT503 will initiate a call to the local CO. The RequestURI header in the INVITE
message contains the phone number used to initiate the call to the local CO.
Fax Support
HT503 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 Fax mode to be T.38 (default). If the service provider
does not support T.38, pass-through mode may be used. To send or receive faxes in fax pass through
mode, users must select all the Preferred Codecs to be PCMU/PCMA (G.711-µ/a).

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CALL FEATURES
TABLE 6: HT503 CALL FEATURE DEFINITIONS
Key Call Features
*30 Block Caller ID (for all subsequent calls)
*31 Send Caller ID (for all subsequent calls)
*47 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 Disable Call Waiting (for all subsequent calls)
*51 Enable Call Waiting (for all subsequent calls)
*67 Block Caller ID (per call). Dial “*67” + ” number ”. No dial tone is played in the middle.
*82 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 Disable Call Waiting (per call). Dial “*70” + ” number ”. No dial tone is played in the middle.
*71 Enable Call Waiting (per call). Dial “*71” + ” number ”. No dial tone is played in the middle.
*72 Unconditional Call Forward: Dial “*72” and then the forwarding number followed by “#”. Wait for
dial tone and hang up. (dial tone indicates successful forward)
*73 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 Disable Do Not Disturb (DND): When disabled, incoming calls are accepted.
*87 Blind Transfe
r
*90 Busy Call Forward: Dial “*90” and then the forwarding number followed by “#”. Wait for dial tone
then hang up.
*91 Cancel Busy Call Forward. To cancel “Busy Call Forward”, dial “*91”, wait for dial tone, then
hang up.
*92 Delayed Call Forward. Dial “*92” and then the forwarding number followed by “#”. Wait for dial
tone then hang up.
*93 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 server as Re-Dial key.
*02 Forcing a Codec per call. *027110 (PCMU), *027111 (PCMA), *02723 (G723), *02729 (G729),
*0272616 (G726-r16), *0272624 (G724-r24), *0272632 (G726-r32), *0272640 (G726-r40),
*027201 (iLBC)

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CONFIGURATION GUIDE
Configuring HT503 through Voice Prompt
DHCP MODE
Follow Table 4 with voice menu option 01 to enable HT503 to use DHCP.
STATIC IP MODE
Follow Table 4 with voice menu option 01 to enable HT503 to use STATIC IP mode, then use option 02,
03, 04 to set up HT503’s IP, Subnet Mask, Gateway 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, HTTP and HTTPS.
FIRMWARE UPGRADE MODE
Select voice menu option 17 to choose firmware upgrade mode. There are three options:
1) always check, 2) check only 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 HT503 with Web Browser
HT503 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 HT503 through a Web browser such as
Microsoft’s IE, AOL’s Netscape or Mozilla Firefox installed on Windows or Unix OS. (Macintosh OS is not
included).
Access the Web Configuration Menu
The HT503 HTML configuration page can be accessed via LAN or WAN ports.
•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 IP address of the LAN port. http://192.168.2.1. You
will see the log in page of the device.
•FROM THE WAN PORT:
1. Follow table 4 to find the WAN side IP address.
2. Open a web browser, type in the WAN side IP address – for example:
http://HT503-WAN-IP-Address
Note:
•WAN side HTTP access is disabled by default for security reason. You can enable HTTP access
on the configuration page by setting “WAN side HTTP access” to be YES.
•Initial access to the configuration pages is always from the LAN port. The instructions are listed
above.
•The IVR announces 12 digits IP address, you need to strip out the leading “0” in the IP address.
For ex. IP address: 192.168.001.014, you need to type in http://192.168.1.14 in the web browser.
Once the HTTP request is entered and 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 123 Only Status and Basic Settings
Administrator Level admin Browse all pages
The password is case sensitive with maximum length of 25 characters. The factory default password for
End User and administrator is “123” and “admin” respectively. Only an administrator can access the
“ADVANCED SETTING”, “FXS PORT” and “FXO PORT” configuration pages.
NOTE: If you can not log into the configuration page by using the default password, please check with
the VoIP service provider. It is most likely the VoIP service provider has provisioned the device and
configured for you therefore the password has already been changed.

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Only an administrator can access the “ADVANCED SETTING”, “FXS PORT” and “FXO PORT”
configuration pages. Please reference the GUI pages using the following link:
http://www.grandstream.com/support/ht_series/ht503/documents/ht503_gui.zip.
DEFINITIONS
This section will describe the options in the Web configuration user interface. As mentioned, a user can
log in as an administrator or end-user.
Functions available for the end-user are:
•STATUS: Displays the network status, account status, software version and MAC-address of the
phone
•BASIC SETTINGS: Basic preferences such as date and time settings, multi-purpose keys and
LCD settings can be set here.
Additional functions available to administrators are:
•ADVANCED SETTINGS: To set advanced network settings, codec settings and XML
configuration settings.
•FXS PORT: To configure the FXS port.
•FXO PORT: To configure the FXO port.
TABLE 7: STATUS PAGE
MAC Address The device ID, in HEX format. This is very important ID for ISP troubleshooting. Both
LAN and WAN MAC addresses are located here. The LAN MAC address is used for
provisioning and is written on the label in the original box as well as on the label located
on the back panel of the device.
WAN IP Address This field shows IP address of the HT503.
Product Model This field contains the product model info, such as HT503.
Software Version Program: This is the main software release. This number is always used for firmware
upgrade. Current release is 1.0.0.15
Bootloader: current version is 1.0.0.7
Core: current version 1.0.0.23
Base: current version is 1.0.0.66
System Uptime This shows system up time since last reboot.
PPPoE Link Up This shows whether the PPPoE is up if connected to DSL modem
NAT This shows what kind of NAT the HT503 is connected to. It is based on STUN
protocol. If the detected NAT is symmetric NAT, STUN will not work and Outbound
Proxy needed to make HT503 functioning correctly.
Port Status Displays information regarding the individual FXS ports.
Port Hook Registration DND Forward Busy
Forward Delayed Forward
FXS On Hook Registered Yes 613
FXO Idle Registered No 614
•Both FXS port and FXO port are registered with this SIP Server.
•FXS Port user has set Do Not Disturb.
•FXS Port user has set his calls to be forwarded unconditionally to ext 613.
FXO Port user has set his calls to forward to 614 when his phone is busy.

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TABLE 8: BASIC SETTINGS
End User Password This contains the password for end user to access the Web Configuration Menu. User
can put new password here. This field is case sensitive with maximum of 25 characters
Web Port This is the device’s internal HTTP server port. Default is 80.
Telnet Server Default is set to YES. Telnet access is allowed to the device in this case. Used only for
special purposes such as debugging and troubleshooting. List of available commands
will be shown by pressing >help command from telnet console.
IP Address •If DHCP mode is enabled, then all the field values for the Static IP mode are not
used (even though they are still saved in the Flash memory.) The HT503 will acquire
its IP address from DHCP in the network.
•PPPoE settings are usually for DSL/ADSL modem users. The HT503 will attempt to
establish a PPPoE session if PPPoE account is set.
•If Static IP mode is selected, the IP address, Subnet Mask, Default Router IP
address, DNS Server 1 (mandatory), DNS Server 2 (optional) fields need to be
configured.
DHCP hostname This option specifies the name of the client. This field is optional but may be required
by some Internet Service Providers. Default is blank.
DHCP domain This option specifies the domain name that client should use when resolving
hostnames via the Domain Name System. Default is blank.
DHCP vendor class ID This option is used by clients and servers to exchange vendor-specific information.
Default is blank.
PPPoE account ID PPPoE username. Necessary if your ISP requires you to use a PPPoE (Point to Point
Protocol over Ethernet) connection
PPPoE password PPPoE account password
PPPoE Service name This field is optional. If your ISP uses a service name for the PPPoE connection, enter
the service name here. Default is blank.
Preferred DNS The address of your preferred DNS server.
Time Zone This parameter controls how the displayed date/time will be adjusted according to the
specified time zone.
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,
This indicates a time zone with 6 hours offset with 1 hour ahead which is U.S central
time. If 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…)
The 3rd number indicates weekday: 0,1,2,..,6( for Sun, Mon, Tues,..,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 the 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 the HT503.
NAT Maximum Ports The number of ports that can be managed while in NAT router mode.
Range: 0 – 4096, default is 1024. Typically one port per connection.
NAT TCP Timeout NAT TCP idle timeout in seconds. Connection will be closed after preconfigured,
timeout if not refreshed.
Range: 0 - 3600