Grandstream Networks, Inc.
HT503
FXS/FXO Port
Analog Telephone Adaptor
HT503 User Manual
Firmware Version 1.0.0.6
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TABLE OF FIGURES
HT503 USER MANUAL
FIGURE 1: CONNECTING THE HT503 ............................................................................................................... 5
FIGURE 2: INTERCONNECTION DIAGRAM OF THE HT503 ................................................................................... 6
FIGURE 3: SCREENSHOT OF CONFIGURATION LOG-IN PAGE............................................................................ 19
FIGURE 4: SCREENSHOT OF REBOOTING SCREEN........................................................................................... 31
TABLE OF TABLES
HT503 USER MANUAL
TABLE 1: DEFINITIONS OF THE HT503 CONNECTORS ....................................................................................... 5
TABLE 2: HT503 TECHNICAL SPECIFICATIONS ................................................................................................. 7
TABLE 3: HT503 HARDWARE SPECIFICATION................................................................................................... 8
TABLE 4: HT503 IVR MENU DEFINITIONS ........................................................................................................ 9
TABLE 5: HT503 CALL FEATURE DEFINITIONS ............................................................................................... 16
TABLE 6: HT503 LED DEFINITIONS............................................................................................................... 16
TABLE 7: HT503 DEVICE STATUS PAGE DEFINITIONS..................................................................................... 19
TABLE 8: HT503 BASIC SETTINGS PAGE DEFINITIONS.................................................................................... 19
TABLE 9: HT503 ADVANCED SETTINGS PAGE DEFINITIONS ............................................................................ 22
TABLE 10: HT503 FXS PORT SETTINGS PAGES DEFINITIONS ....................................................................... 23
TABLE 11: HT503 FXO PORT SETTINGS PAGES DEFINITIONS....................................................................... 26
TABLE 12: HT503 CALL PROGRESS TONES SETTINGS PAGE DEFINITIONS ...................................................... 30
TABLE OF GUI INTERFACES
HT503 USER MANUAL
1. SCREENSHOT OF CONFIGURATION LOGIN PAGE
2. STATUS CONFIGURATION PAGE DEFINITIONS
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/resources.html
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 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.
•
•
This document is contains links to Grandstream GUI Interfaces. Please remember to download these
This document is subject to change without notice. The latest electronic version of this user manual is
available for download from the following location:
•
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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HT503 User Manual
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INSTALLATION
EQUIPMENT PACKAGING
The HT503 ATA package contains:
•
•
•
•
One HT503 Main Case
One Universal Power Adaptor
One Ethernet Cable
One HT503 Vertical Stand
CONNECTING YOUR ATA
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
HT503
HT503
Back View
Front View
RJ-45 Ports
10/100 Mbps
Display LEDs
(Green)
Power
Supply
(12V)
Reset
RJ11
RJ11
FXS Port FXO Port
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.
TABLE 1: DEFINITIONS OF THE HT503 CONNECTORS
12VDC, 0.5A
Power adapter connection
LAN Port (RJ-45)
WAN Port (RJ-45)
PHONE (RJ-11)
LINE (RJ-11)
Connect the LAN port with an Ethernet cable to your PC.
Connect to the internal LAN network or router.
FXS port to be connected to analog phones / fax machines.
FXO port should be connected to the PSTN line
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FIVE EASY STEPS TO INSTALL 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.
FIGURE 2: INTERCONNECTION DIAGRAM OF THE HT503
Internet ADSL/Cable
Modem Ethernet
Analog Phone
WAN
FXO
FXS
PSTN
Cloud
Cordless
LAN
Fax
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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.
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 2: 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
Factory Reset Button
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 or HTTP
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) for deployed devices including
behind firewall/NAT
Syslog support
Yes
DHCP Server/Client
Audio Features
Advanced Digital Signal Processing (DSP)
Dynamic negotiation of codec and voice payload length
Support for G.723.1A, G.729A/B/E, G.711, G.726-40/24/16, 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
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
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HARDWARE SPECIFICATION
The table below lists the hardware specification of HT503.
TABLE 3: HT503 HARDWARE SPECIFICATION
LAN interface
1xRJ45 10/100 Mbps Port
1xRJ45 10/100 Mbps Port
1 x FXS (RJ11)
WAN interface
FXS telephone port
FXO telephone port (PSTN Port) 1x PSTN pass-through and life line port
LED
Power, WAN, LAN, PHONE, and LINE (Green)
Input: 100–240 VAC, 50-60 Hz
Output: 12VDC, 0.5A, UL certified
Universal Switching
Power Adaptor
Dimension
25mm x 115mm x 75mm (when laying flat);
115mm x 25mm x 75mm (standing up)
Weight
Approximately 0.6lbs (0.3kg)
Operational: 32° - 104°F or 5° – 45°C
Storage: 10°–130°F
Temperature
Humidity
10% - 90%
(non-condensing)
Compliance
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BASIC OPERATIONS
GET FAMILIAR WITH VOICE PROMPT
HT503 has a stored voice prompt menu for quick browsing and simple configuration. Currently, the voice
prompt menu is designed for the FXS port only. Dial “***” from the analog phone to enter the voice
prompt.
TABLE 4: 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, 07,12-17,47 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
04
“Subnet “ + IP address
“Gateway “ + IP address
Same as menu 02
Same as menu 02
05
07
“DNS Server “ + IP address
Preferred Vocoder
Same as menu 02
Press “9” to move to the next selection in the list:
•
•
•
•
•
PCM U / PCM A
G.723
G.729
G.726
iLBC
12
13
WAN Port Web Access
Press “9” to toggle between enable / disable
Firmware Server IP Address
Announces current Firmware Server IP address. Enter 12 digit new
IP address.
14
15
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 a 12 digit IP address to make a direct IP call, after dial tone.
(See “Make a Direct IP Call”.)
Press “9” to reboot the device; or
Enter encoded MAC address to restore factory default setting (See
“Restoring Factory Settings”)
“Invalid Entry”
Automatically returns to main menu
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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 and 12 digits for IP
address. For IP address, add 0 before the digits if the digits are less than 3 (like 192.168.0.26
should be key in like 192168000026, no dot needed while input). 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 PHONE CALLS
CALLING 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 HT with an analog phone and another VoIP Device, to talk to
each other in an ad hoc fashion without a SIP proxy. This kind of VoIP calls can be made between two
parties if:
•
Both HT503 and other VoIP Device (i.e. another Handytone ATA or Budgetone SIP phone or
other VoIP unit) 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).
TO PLACE A DIRECT IP CALL:
1. Pick up the analog phone (or use the speakerphone),
2. Access the voice menu prompt by dial “***”
3. Dial “47” to access the direct IP call menu
4. At voice prompt “Direct IP Calling” and dial tone, enter a 12-digit target IP address to
make a call.
Destination ports can be specified by using “*4” (encoding for “:”) followed by the port number.
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EXAMPLES:
1. If the target IP address is 192.168.0.10, the dialing convention is
Voice Prompt with option 47, then 192 168 000 010
followed by pressing the “#” key if it is configured as a send key or wait for more than 5 seconds.
2. If the target IP address/port is 192.168.1.20:5062, then the dialing convention would be:
Voice Prompt with option 47, then 192168001020*45062
followed by pressing the “#” key if it is configured as a send key or wait for 4 seconds.
NOTE: When placing a direct IP call, the “Use Random Port” should be set to “NO”.
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 A and B are in
conversation. Party A wants to Blind Transfer Party B to C:
1. A presses FLASH on the analog phone to hear the dial tone.
2. Then A dials *87, then dials C’s number, and then presses #
3. 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 A and 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. A then dial C’s number followed by #.
3. If C answers the call, A and C are in conversation. Then A can hang up to complete transfer.
4. If C does not answer the call, A can press “flash” back to talk to B.
NOTE: When Attended Transfer 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 both Star Code Style and Bellcore Style 3-way conferencing.
Star Code Style 3-way Conference
This function is applicable on the FXS port for VoIP calls only. Assume that parties A and B are in
conversation. Party A wants to bring C into a 3-way conference:
1. A presses FLASH (on the analog phone, or Hook Flash for old model phones) to get a dial tone.
2. A dials *23 then C’s number then # (or wait for 4 seconds).
3. If C answers the call, then A 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.
NOTE: “Enable Call Feature” has to be set to YES in FXS PORT in the web configuration page.
Bellcore Style 3-way Conference
To use the Bellcore Style conference, the “Use Bell-style 3-way Conference” field in FXS PORT web
configuration must be enabled.
Assume that parties A and B are in conversation. Party A (using the HT503) wants to bring C into a 3-
way conference:
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1. A presses FLASH (on the analog phone, or Hook Flash for old model phones) to get a dial
tone.
2. A 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 the conference, C will be dropped out.
Note: Party A is the call initiator for both calls with party B and party C.
PSTN PASS THROUGH
HT503 supports PSTN pass through using the FXS port. The user can place and receive PSTN calls
when the FXS port is in use.
•
•
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.
•
It the HT503 loses power, 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 (configured on the 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
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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 configuration page).
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. 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 prefix or a telephone number in the “Route call to PSTN” in
the BASIC SETTINGS web configuration page. If the dialed digits match the specified prefix, outbound
calls will be initiated from 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 626, all outgoing calls starting with 626 will be
initiated from the PSTN line.
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FORWARD CALLS TO PSTN
Any VOIP call may be forwarded to a specified PSTN number if the call is not answered after a pre
configured numbers of rings. By default “Number of Rings” parameter has value 4.
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.
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. 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-u/a).
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CALL FEATURES
TABLE 5: HT503 CALL FEATURE DEFINITIONS
Key
Call Features
*30
Block Caller ID (for all subsequent calls)
*31
*47
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
*70
*71
*72
Call Return Service: Dial *69 and the phone will dial the last incoming phone number received.
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)
*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
*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
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 serve as Re-Dial key.
LED Light Pattern Indication
TABLE 6: HT503 LED DEFINITIONS
LEDs
POWER LED
WAN LED
Indicates Power. Remains ON when power is connected
Indicates LAN (or WAN) port activity
LAN LED
Indicates PC (or LAN) port activity
PHONE/ LINE LED
Indicates the status of the FXS port 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.
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CONFIGURATION GUIDE
CONFIGURING HT503 THROUGH VOICE PROMPT
DHCP MODE
Follow Table 3 with voice menu option 01 to enable HT503 to use DHCP.
STATIC IP MODE
Follow Table 3 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.
TFTP SERVER ADDRESS
Follow Table 3 with voice menu option 06 to configure the IP address of the TFTP server.
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. There are three options:
1) always check, 2) check when pre/suffix changes, and 3) never upgrade
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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
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:
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.
END USER CONFIGURATION
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
Administrator Level
123
Only Status and Basic Settings
Browse all pages
admin
Only an administrator can access the “ADVANCED SETTING” configuration page. Please reference the
Once this HTTP request is entered and sent from a Web browser, the HT503 will respond with the
following login screen:
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FIGURE 3: SCREENSHOT OF CONFIGURATION LOG-IN PAGE
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” configuration page.
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.
After a correct password is entered in the login screen, the embedded web server will respond with the
Configuration pages which are explained in details below.
TABLE 7: HT503 DEVICE STATUS PAGE DEFINITIONS
MAC Address
The device ID, in HEX format. This is very important ID for ISP troubleshooting.
This field shows IP address of the HT503.
WAN IP Address
Product Model
Software Version
This field contains the product model info, such as HT503.
Program: This is the main software release. This number is always used for firmware
upgrade. Current release is 1.0.0.6
Bootloader: current version is 1.0.0.7.
Core: current version 1.0.0.14
Base: current version is 1.0.0.5
System Uptime
PPPoE Link Up
NAT
This shows system up time since last reboot.
This shows whether the PPPoE is up if connected to DSL modem
This shows what kind 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
FXO
On Hook
On Hook
Registered
Registered
Yes
No
613
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.
TABLE 8: HT503 BASIC SETTINGS PAGE DEFINITIONS
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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.
Default is set to YES
Telnet Server
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
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 ID
PPPoE account ID
This option is used by clients and servers to exchange vendor-specific information.
Default is blank.
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
Time Zone
The address of your preferred DNS server.
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
NAT TCP Timeout
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 idle timeout in seconds. Connection will be closed after preconfigured,
timeout if not refreshed.
Range: 0 - 3600
NAT UDP Timeout
NAT TCP idle timeout in seconds. Connection will be closed after preconfigured,
timeout if not refreshed.
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Range: 0 – 3600, default is 300
Uplink Bandwidth
Downlink Bandwidth
Enable UPnP
The maximum uplink bandwidth permitted by the device. This function is disabled by
default. The total bandwidth can be set as: 128K, 256K, 512K, 1M, 4M or 10M.
For example if 64 is configured, there will be at least 64kbps reserved for VoIP. The
primary function of this setting is to reserve bandwidth for VoIP.
The maximum downlink bandwidth permitted by the device. This function is disabled
by default. The total bandwidth can be set as: 128K, 256K, 512K, 1M, 4M or 10M.
For example if 128 is configured, there will be at least 128kbps reserved for VoIP. The
primary function of this setting is to reserve bandwidth for VoIP.
When set to “Yes”, the HT503 acts as an UPnP gateway for your UPnP enabled
applications. UPnP = “Universal Plug and Play”
Reply to ICMP on WAN
Port
When set to “Yes”, the HT503 responds to the PING command from other computers,
but is also made vulnerable to DOS attacks. Default is No.
WAN Side HTTP/Telnet
Access
When set to “Yes”, the user can access the web configuration pages through the WAN
port, instead of through the PC port. Warning: this configuration is less secure than the
default option. Default is No.
Cloned WAN MAC
Address:
This allows the user to change/set a specific MAC address on the WAN interface.
Note: Set in Hex format
LAN Subnet Mask
LAN DHCP Base IP:
DHCP IP Lease Time
DMZ IP:
Sets the LAN subnet mask. Default value is 255.255.255.0
Base IP for the LAN port, which functions as default gateway for its LAN. Default value
is 192.168.2.1
The length of time the IP address is assigned to the LAN clients. Value is set in units of
hours. Default value is 120 hrs (5 days).
This function forwards all WAN IP traffic to a specific IP address if no matching port is
used by HT503 or in the defined port forwarding.
Port Forwarding:
Allows users to forward a matching (TCP/UDP) port to a specific LAN IP address with a
specific (TCP/UDP) port.
PSTN access code
PIN for PSTN calls
PIN for VoIP calls
Route Call to PSTN
The code to access the PSTN line (Maximum 5 digits). Default is “*00”.
PIN code to bridge from VoIP to PSTN (Maximum 8 digits, No Default)
PIN code to bridge from PSTN to VoIP (Maximum 8 digits, No Default)
If the dialed digits match one of the specified prefix here, outbound calls will be initiated
from PSTN line. This field is especially useful for emergency calls.
Unconditional Call
Forward to PSTN
Calls are unconditionally forwarded to the specified PSTN phone number for all
incoming VoIP calls on FXO port.
Unconditional Call
Forward to VoIP
Calls are unconditionally forwarded to the specified VoIP phone number for all
incoming PSTN calls.
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ADVANCED CONFIGURATION AND FXS/FXO PORTS PARAMETERS
To login to the Advanced Setting and FXS port configuration pages, administrator password is required.
The default administrator password is “admin”. User can change the administrator password here. The
password is case sensitive and the maximum length is 25 characters.
TABLE 9: HT503 ADVANCED SETTINGS PAGE DEFINITIONS
Admin Password
Administrator password. Only the administrator can configure the “Advanced Settings”
page. Password field is purposely blanked for security reason after clicking update and
saved. The maximum password length is 25 characters.
Layer 3 QoS
Layer 2 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 settings. Default setting is blank. VLAN supported equipment is required
when configuring these settings.
STUN Server
IP address or Domain name of the STUN server.
Keep-alive interval
This parameter specifies how often the HT503 sends a blank UDP packet to the SIP
server in order to keep the NAT “pin hole” open. Default is 20 seconds.
Firmware Upgrade and
Provisioning
Enables the HT503 to download firmware or configuration files through either TFTP or
HTTP servers. The default method is HTTP.
Via TFTP Server
This is the IP address of the configured TFTP server. If this is configured, the HT503
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: Firmware upgrades may take up to 10 minutes depending on your network
environment. On a LAN it usually takes about 2 minutes. 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.
“:6688” is the specific TCP port where the HTTP server is listening; Omit if using
default port 80. Note: If Auto Upgrade is set to No, F/W will download at boot time.
Firmware Server Path
Config Server Path
Firmware File Prefix
IP address or domain name of firmware server.
IP address or domain name of configuration server.
Default is blank. If configured, HT503 will request the firmware file with the prefix. This
setting is useful for ITSPs. End user should keep it blank.
Firmware File Postfix
Config File Prefix
Default is blank. End users should keep it blank.
Default is blank. End users should keep it blank.
Default is blank. End users should keep it blank.
Config File Postfix
Automatic Upgrade
Choose “Yes” to enable automatic upgrades and provisioning. In “check for new
firmware every” field, enter the number of days to set the frequency in which the HT503
will check the server for firmware or configuration upgrades. When set to “No” the
HT503 will only upgrade at boot time. You can also have the HT503 check for updates
only when the firmware file prefix/suffix changes. The Default is “Yes”.
Authenticate Conf File
Firmware Key
This protects the configuration from an unauthorized change. If set to “Yes, the
configuration file is authenticated before acceptance.
Key for firmware encryption. (32 digits in hexadecimal format. End users should keep it
blank.
Call Progress Tones
Using these settings, users can configure tone frequencies according to their
preference. By default they are set to North American frequencies.
These should be configured with known values to avoid uncomfortable high pitch
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sounds. ON is the period of ringing (“On time” in ‘ms’) while OFF is the period of
silence. In order to set a continuous tone, OFF should be zero. Otherwise it will ring ON
ms and a pause of OFF ms and then repeat the pattern.
Here is an example for the configuration of the dial tone for North America:
f1=350@-13,f2=440@-13,c=0/0;
Syntax: f1=freq@vol, f2=freq@vol, c=on1/off1-on2/off2-on3/off3; [...]
(Note: freq: 0 - 4000Hz; vol: -30 - 0dBm)
Lock Keypad Update
Disable Voice Prompt
Disable Direct IP Calling
NTP server
If set to “Yes”, the configuration update via keypad is disabled.
Disables the voice prompt configuration. Default is “No.”
Disables the Direct IP Call function. Default is “No.”
URL or IP address of the NTP server, Used to synchronize the date/time.
The IP address or URL of syslog server, especially useful for ITSP
Syslog Server
Syslog Level
Select the ATA to report the log level. Default is NONE. The level is either one of
DEBUG, INFO, WARNING or ERROR. Syslog messages are sent based on the
following events:
•
•
•
•
•
•
•
•
•
•
product model/version on boot up (INFO level)
NAT related info (INFO level)
sent or received SIP message (DEBUG level)
SIP message summary (INFO level)
inbound and outbound calls (INFO level)
registration status change (INFO level)
negotiated codec (INFO level)
Ethernet link up (INFO level)
SLIC chip exception (WARNING and ERROR levels)
memory exception (ERROR level)
The Syslog uses USER facility. In addition to standard Syslog payload, it contains the
following components: GS_LOG: [device MAC address][error code] error message
Ex. May 19 02:40:38 192.168.1.14 GS_LOG: [00:0b:82:00:a1:be][000] Ethernet link is up
Download Device
Configuration
This is a special feature that enables the user to create a text file backup of your
existing configuration.
TABLE 10: HT503 FXS PORT SETTINGS PAGES DEFINITIONS
Account Active
SIP Server
When set to yes the FXS port is activated.
This field contains the URL string or the IP address (and port, if different from 5060) of
the SIP proxy server. e.g., the following are some valid examples: sip.my-voip-
provider.com, or sip:my-company-sip-server.com, or 192.168.1.200:5066
IP address or Domain name of Outbound Proxy, or Media Gateway, or Session Border
Controller. Used by ATA for firewall or NAT penetration in different network
environment. If symmetric NAT is detected, STUN will not work and ONLY Outbound
Proxy will work.
Outbound Proxy
SIP Transport
User can select UDP or TCP or TLS.
NAT Traversal (STUN)
This setting decides whether the NAT traversal mechanism is activated. It should be
set to “Yes” if the device is behind a NAT router. If no outbound proxy is configured, a
STUN server needs to be set to activate STUN detection mechanism. Usually ITSP will
provide these settings. If this field is set to “Yes”, then the device will periodically send
a dummy UDP packet to the SIP server to pinhole the NAT.
SIP User ID
User account information, provided by VoIP service provider (ITSP), usually has the
form of digit similar to phone number or actually a phone number. This field contains
the user part of the SIP address for this phone. e.g., if the SIP address is
sip:my_user_id@my_provider.com, then the SIP User ID is: my_user_id.
Do NOT include the preceding “sip:” scheme or the host portion of the SIP address in
this field.
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Authenticate ID
ID used for authentication, usually same as SIP user ID, but could be different and
decided by ITSP.
Authentication Password Password for ATA to register to (SIP) servers of ITSP. Purposely left blank once saved
for security. Maximum length is 25.
Name
SIP service subscriber’s name which will be used for Caller ID display
Use DNS SRV:
User ID is Phone Number
Default is No. If set to Yes the client will use DNS SRV to lookup for the SIP server.
If “Yes” is set, a “user=phone” parameter will be attached to the “From” header in SIP
request
SIP Registration
This parameter controls whether the HT503 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 device will first send registration request to remove all
previous bindings. Use only if proxy supports this remove bindings request.
Outgoing Call w/o
Registration
This parameter lets users place outgoing calls even when not registered (if allowed by
ITSP) but it’s unable to receive incoming calls.
Register Expiration
This parameter allows the user to specify the time frequency (in minutes) the
HandyTone ATA refreshes its registration with the specified registrar. The default
interval is 60 minutes (or 1 hour). The maximum interval is 65535 minutes (about 45
days).
Local SIP port
Local RTP port
This parameter defines the local SIP port the HT503 will listen and transmit. The default
value for FXS port is 5060.
This parameter defines the local RTP-RTCP port pair the HandyTone ATA will listen
and transmit. It is the base RTP port for channel 0. When configured, channel 0 will use
this port _value for RTP and the port_value+1 for its RTCP; channel 1 will use
port_value+2 for RTP and port_value+3 for its RTCP. The default value for FXS port is
5004.
Use Random Port
Refer to Use Target
Default No. If set to Yes, the device will pick randomly-generated SIP and RTP ports.
This is usually necessary when multiple HandyTone ATAs are behind the same NAT.
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
This parameter sets the payload type for DTMF using RFC2833
Send DTMF as inband (in-audio).
DTMF Via RFC2833
DTMF Via SIP INFO
Send Flash Event
Enable Call Features
Send DTMF via RTP (According to RFC 2833)
Send DTMF via SIP INFO message.
Default is No. If set to yes, flash will be sent as DTMF event.
Default is Yes. Toggles support for advanced call features and star code functions.
Offhook
Auto-Dial
This parameter allows users to configure a User ID or extension number to be
automatically dialed upon offhook. Please note that only the user part of a SIP address
needs to be entered here. The HT503 will automatically append the “@” and the host
portion of the corresponding SIP address.
Note: Please write down the IP address of the ATA if you use this feature as it
will prevent access to the IVR and the only way to access the device configuration will
be via the web configuration page.
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
Customizes Ring Tones 1 to 3 with an associated Caller ID. When selected, the device
will ONLY use this ring tone when the incoming call is set up for Caller ID. The System
Ring Tone is used for all other calls. When selected with no Caller ID 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 during a CWC. The
CWCID will still be displayed.
Ring Timeout
Sets the time in which an incoming call will stop ringing when not picked up.
Default is 4 seconds.
No Key Entry Timeout
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
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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. 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).
Note: 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
This 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”, the “#” key can be included as part of a 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;
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.
When set to “Yes”, the “From” header along with Privacy and P_Asserted_Identity
headers in outgoing INVITE messages 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 a 486 busy message.
Special Features
Default is “Standard.” Choose the selection to meet some special requirements from
Softswitch vendors.
Preferred Vocoder
The HT503 supports 5 different Vocoder types including
1. G.711 A/µ law, (Displayed as PCMA/PCMU)
2. G.723.1,
3. G.726 (Supports bit rates 16, 24, 32, and 40)
4. G.729A/B/E,
5. iLBC
Users can configure Vocoders in a preference list that will be included with the same
preference order in SDP message.
G723 Rate:
This defines the encoding rate for G723 vocoder. Default setting is 6.3kbps.
This sets the iLBC size in 20ms or 30ms
iLBC Frame Size:
iLBC Payload Type:
This defines payload type for iLBC. Default value is 97. The valid range is between 96
and 127.
G726-16 Payload Type
G726-24 Payload Type
G726-40 Payload Type
G729E Payload Type
VAD
Defines payload type for G726-16. Default value is 98. Range is from 96 to 127.
Defines payload type for G726-24. Default value is 99. Range is from 96 to 127.
Defines payload type for G726-40. Default value is 103. Range is from 96 to 127.
Defines payload type for G729E. Default value is 102. Range is from 96 to 127
Default is “No.” VAD allows detecting the absence of audio and conserves bandwidth
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by preventing the transmission of “silent packets” over the network.
Symmetric RTP
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.
Fax Mode
T.38 (Auto Detect) FoIP by default, or fax Pass-Through (must use PCMU/PCMA)
Fax Detection Mode
Default is Callee. This decides whether Caller or Callee sends out the re-invite for T.38
or Fax Pass-Through.
Jitter Buffer Type
Jitter Buffer Length
SRTP Mode
Select either Fixed or Adaptive based on network conditions.
Select Low, Medium, or High based on network conditions.
Secure RTP protocol used for media transmission over VoIP. Disabled by default.
Other modes are: enabled but not forced & enabled and forced.
SLIC Setting
Dependent on standard phone type (and location).
Called ID Scheme
Caller ID TX Level (dB)
Bellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 – BT, & NTT Japan
A value for Caller ID information sent by a phone connected to the FXS port.
(-96 – 0dB. Default -14dB)
Polarity Reversal
If set to “Yes”, polarity will be reversed upon call establishment and termination.
Default is No.
Loop Current Disconnect Set it to “Yes” of the traditional PBX you are using with HT503 uses this method fir
signaling call termination. Default is No.
Loop Current Disconnect A configurable period of time in which the FXS port will drop off voltage on the line to
Duration
indicate to the local party that the call is disconnected from the remote side.
(100-10000 ms. Default 200 ms)
Hook Flash Timing
The time period when the cradle is pressed (Hook Flash) to simulate a FLASH. Adjust
this time value to prevent unwanted activation of the Flash/Hold and automatic phone
ring-back.
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
This function lets you 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 high pitch sounds.
TABLE 11: HT503 FXO PORT SETTINGS PAGES DEFINITIONS
Account Active
SIP Server
When set to “Yes” the FXO 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 HT503 for firewall or NAT penetration in different network
environments. If symmetric NAT is detected, STUN will not work and ONLY way to
correct the problem is to use the outbound proxy.
SIP Transport
User can select UDP, TCP or TLS
NAT Traversal (STUN)
This parameter defines whether or not the HT503 NAT traversal mechanism is
activated. If set to “Yes” with a STUN server also specified, the HT503 will perform
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 HT503 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 HT503 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.
Authenticate ID
The SIP service subscriber’s ID used for authentication. Can be identical to or different
from SIP User ID.
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Authenticate Password
Name
SIP service subscriber’s account password.
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 the server.
User ID is Phone Number
If the HT503 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 HT503 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 Without
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 HT503
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 HT503 will listen and transmit. The default value for FXS
port is 5062.
Defines the local RTP-RTCP port pair the HT503 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 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 HT503 units are behind the same
NAT.
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
Sends DTMF using RFC2833
Sends DTMF as inband (in-audio).
DTMF via RFC2833
DTMF via SIP INFO
Send Flash Event
Enable Call Features
Proxy Require
Sends DTMF via RTP (according the RFC2833).
Send DTMF as a SIP INFO message.
Default is No. If set to “Yes”, flash will be sent as DTMF event
Default is Yes. Toggles support for advanced call features and star code functions.
SIP Extension to notify SIP server that the unit is behind a NAT/Firewall.
NAT IP address used in SIP/SDP message. Default is blank.
Use NAT IP
Distinctive Ring Tone
Customizes Ring Tones 1 to 3 with an associated Caller ID. When selected, the device
will ONLY use this ring tone when the incoming call is set up for Caller ID. The System
Ring Tone is used for all other calls. When selected with no Caller ID configured, the
selected ring tone will be used for all incoming calls.
Disable Call Waiting
Turns off the Call Waiting feature. Default is No.
Disable Call Waiting
Tone
Default is No. This is to disable the stutter Call Waiting Tone during a CWC. The
CWCID will still be displayed.
Ring Timeout
Early Dial
Sets the time in which an incoming call will stop ringing when not picked up.
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. 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).
Note: This feature is NOT designed to work with and should NOT be enabled for
direct IP-to-IP calling.
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Dial Plan Prefix
Sets the prefix added to each dialed number.
Use # as Dial Key
This 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”, the “#” key can be included as part of a number.
Dian 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;
f. xx+ - at least 2 digit number;
g. ^ - exclude;
h. [3-5] - any digit of 3, 4, or 5;
i.
[147] - any digit 1, 4, or 7;
j. <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.
When set to “Yes”, the “From” header along with Privacy and P_Asserted_Identity
headers in outgoing INVITE messages 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 a 486 busy message.
Special Features
Default is “Standard.” Choose the selection to meet some special requirements from
Softswitch vendors.
Preferred Vocoder
The HT503 supports 5 different Vocoder types including
1. G.711 A/µ law, (Displayed as PCMA/PCMU)
2. G.723.1,
3. G.726 (Supports bit rates 16, 24, 32, and 40)
4. G.729A/B/E,
5. iLBC
Users can configure Vocoders in a preference list that will be included with the same
preference order in SDP message.
G723 Rate
This defines the encoding rate for G723 vocoder. Default setting is 6.3kbps.
This sets the iLBC size in 20ms or 30ms
iLBC Frame Size
iLBC Payload Type
This defines the payload type for iLBC. Default value is 97. The valid range is between
96 and 127.
G726-16 Payload
G726-24 Payload Type
G726-40 Payload Type
G729E Payload Type
VAD
Defines payload type for G726-16. Default value is 100. Range is from 96 to 127.
Defines payload type for G726-24. Default value is 99. Range is from 96 to 127.
Defines payload type for G726-40. Default value is 103. Range is from 96 to 127.
Defines payload type for G729E. Default value is 102. Range is from 96 to 127
Default is “No.” VAD allows detecting the absence of audio and conserves bandwidth
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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 fax Pass-Through (must use PCMU/PCMA)
Fax Tone Detection
Mode
Default is Callee. This decides whether Caller or Callee sends out the re-invite for T.38
or Fax Pass-Through.
Jitter Buffer Type
Jitter Buffer Length
SRTP Mode
Select either Fixed or Adaptive based on network conditions.
Select Low, Medium, or High based on network conditions.
Secure RTP protocol used for media transmission over VoIP. Disabled by default.
Other modes are: enabled but not forced & enabled and forced.
Caller ID Scheme
Bellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 – BT, & NTT Japan
Caller ID RX Level (dB)
An adjustable value for the Caller ID signal to help our device to recognize Caller ID
from different networks. (-96 -0dB. Default -15dB)
Hook Flash Timing
Gain
The time period when the cradle is pressed (Hook Flash) to simulate a FLASH. Adjust
this time value to prevent unwanted activation of the Flash/Hold and automatic phone
ring-back.
Handset volume adjustment.
•
•
RX is for receiving volume,
TX is for transmission volume.
Default = 0dB for both parameters. Loudest volume: +6dB Lowest volume: -6dB.
Enable Current
Disconnect
The Default value is Yes. This value should be used in case the PSTN provider uses
line power drop to indicate call completion to the end point. In this case the HT503 will
search for a power drop for a preconfigured time frame to disconnect such calls from a
VoIP extension.
Current Disconnect
Threshold (ms)
This is a preconfigured value of duration for a line power drop used by specific service
providers. For example, for a configured value of 500ms the device will ignore any
random voltage drops on the line less than 500ms and the call will be considered as
terminated. This is useful to prevent call drops in some low quality PSTN lines.
Enable PSTN Disconnect If set to Yes, a Tone is used as the disconnect signal.
Tone Detection
PSTN Disconnect Tone
In certain countries, the central office will send a special busy tone to indicate when a
call is disconnected from the remote side. To make the HT503 recognizable, this
parameter can pre-configure this tone. The user should know the frequency values
and cadences of these tones.
Here is an example for the syntax for a busy tone in the U.S.A:
(Syntax: f1=freq@vol, f2=freq@vol, c=on1/off1-on2/off2-on3/off3; [...])
(Note: freq: 0 - 4000Hz; vol: -30 - 0dBm)
(Default: Busy Tone - f1=480@-24,f2=620@-24,c=500/500;)
AC Termination
Number of Rings
AC Termination is a configurable value of impedance of the line provided by different
service providers in different countries. 14 are selectable in this version of the F/W.
Default is 4. This setting specifies number of phone rings (on the phone connected to
the FXS port) before a PSTN incoming call is bridged to VoIP
Note: The number of rings feature serves as a PSTN answer delay, and should be set
to a larger value to allow enough time for the HT503 to decode the Caller ID signal set
by the central office.
PSTN Ring Thru FXS
If Yes, the phone connected to the FXS port will ring a configured amount of times (see
above). If not, the phone connected to the FXS port will not ring.
PSTN Ring Thru Delay
(sec)
If the PSTN Ring Thru Delay is set to Yes, all incoming PSTN calls through FXO will
ring the phone connected to the FXS port, after this delay or after caller id is detected
(whichever comes first).
DTMF Digit Length (ms)
Digit length and Dial Pause are port digit dialing configurations; FXO needs to dial out
digits for VOIP to PSTN 1 stage calls, and unconditional call forward to PSTN, and
route to PSTN. Digit Length is the play time for each digit.
Note: In order to receive the caller ID information, the delay should be set to a value
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larger than the delay required to complete the PSTN caller ID delivery.
DTMF Dial Pause (ms)
First Digit Timeout (sec)
Dial pause is the time between 2 digits for the same scenario as explained above.
Used for PSTN to VoIP calls. PSTN users need to enter the FIRST digit within the first
digit timeout period. Otherwise the call will be dropped.
Inter Digit Timeout
Wait for Dial Tone
When dialing from the PSTN to VoIP, subsequent digits have to be input within the
period of inter-digit timeout. Otherwise the dial plan thinks it is the end of the digit input.
Wait for Dial tone is used for one stage VoIP to PSTN calls. If set to Yes, the device
will first obtain a PSTN line and a dial tone from a central office. After obtaining the dial
tone, the digits dialed will be sent to the central office.
Stage Method (1/2)
This configuration is applicable for VoIP to PSTN calls and indicates one or two stage
dialing methods.
Note: General settings have the same meaning as explained in the FXS page definitions. Here they are
described only as parameters related to the FXO port.
TABLE 12: HT503 CALL PROGRESS TONES SETTINGS PAGE DEFINITIONS
Call Progress Tones
Using these settings, user can configure tone frequencies according to their preference.
By default they are set to North American frequencies.
Frequencies should be configured with known values to avoid uncomfortable high pitch
sounds. ON is the period of ringing (“On time” in ‘ms’) while OFF is the period of
silence. In order to set a continuous tone, OFF should be zero. Otherwise it will ring ON
ms and a pause of OFF ms and then repeat the pattern.
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SAVING THE CONFIGURATION CHANGES
Once a change is made, users should click on the “Update” button in the Configuration page. The HT503
will display a confirmation screen to confirm that the changes have been saved. Click ‘Reboot’ to save
all changes. Please reference the GUI pages using the following link:
Remote Reboot of the HT503
The administrator can remotely reboot the HT503 by clicking on the “Reboot” button at the bottom of the
configuration page. Once done, the following screen will be displayed to indicate that rebooting is
underway. You can login again after about 30 seconds.
FIGURE 4: SCREENSHOT OF REBOOTING SCREEN
Grandstream Device Configuration
The device is rebooting now...
You may relogin by clicking on the link below in 30 seconds.
All Rights Reserved Grandstream Networks, Inc. 2004
NOTE: Interrupting the ‘booting up’ process could permanently damage the device.
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CONFIGURATION THROUGH A CENTRAL SERVER
The Grandstream HT503 can be automatically configured from a central provisioning system.
When the HT503 boots up, it will send TFTP or HTTP request to download configuration file,
“cfg000b82xxxxxx”, where “000b82xxxxxx” is the LAN side MAC address of the HT503
The configuration files can be downloaded via TFTP or HTTP from the central server. A service provider
or an enterprise with large deployment of HT503 can easily manage the configuration and service
provisioning of individual devices remotely from a central server.
Grandstream provides a licensed provisioning system called GAPS that can be used to support
automated configuration of HT503. GAPS (Grandstream Automated Provisioning System) uses enhanced
(NAT friendly) TFTP or HTTP (thus no NAT issues) and other communication protocols to communicate
with each individual HT503 for firmware upgrade, remote reboot, etc.
Grandstream provide GAPS (Grandstream Automated Provisioning System) service to VoIP service
providers. It could be either simple redirection or with certain special provisioning settings. Initially upon
booting up, Grandstream devices by default point to Grandstream provisioning server GAPS, based on
the unique MAC address of each device, GAPS provision the devices with redirection settings so that
they will be redirected to customer’s TFTP or HTTP server for further provisioning. Grandstream also
provide GAPSLITE software package which contains our NAT friendly TFTP server and a configuration
tool to facilitate the task of generating device configuration files.
The GAPSLITE configuration tool is now free to end users. The tool and configuration template are
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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/1.0.0.6
e.g. 168.75.215.190
NOTES:
1. The TFTP server in IP address format can be configured via IVR. Please refer to the
CONFIGURATION GUIDE section for instructions. If the TFTP server is in FQDN format, it must
be set via the web configuration interface.
2. End users recommended using our TFTP server. Its address can be found at
http://www.grandstream.com/firmware.html. Currently, the TFTP server, your HT503 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.
3. Once a “Firmware Server Path” is set, the 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 HT503
will attempt to retrieve the new image files by downloading them into the SRAM. During this
stage, the HT503 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 HT503 will stop the TFTP/HTTP process
and simply boot using the existing code image in the flash.
4. Firmware upgrades usually take around 2 minutes when 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.
5. 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
Directions to download a free 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 GXW400X 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
Microsoft IIS 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 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”, the 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, however they are defined. This allows the device to 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, the Server Provider can spread the Firmware or Configuration File download in minutes
to reduce the Firmware or Provisioning Server load at any given time.
Grandstream Networks, Inc.
HT503 User Manual
Page 34 of 35
Firmware 1.0.0.6
Last Updated: 6/2007
Download from Www.Somanuals.com. All Manuals Search And Download.
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.
FACTORY RESET
IVR Command
Reset default factory settings using the IVR Prompt (Table 5):
1. Dial “***” for voice prompt.
2. Enter “99” and wait for “reset” voice prompt.
3. Enter the encoded MAC address (Look below on how to encode MAC address).
4. Wait 15 seconds and device will automatically reboot and restore factory settings.
Encoding the MAC Address
1. Locate the MAC address of the device. It is the 12 digit HEX number on the bottom of the
unit.
2. Key in the MAC address. Use the following mapping:
0-9: 0-9
a. A: 22 (press the “2” key twice, “A” will show on the LCD)
b. B: 222
c. C: 2222
d. D: 33 (press the “3” key twice, “D” will show on the LCD)
e. E: 333
f. F: 3333
For example: if the MAC address is 000b8200e395, it should be keyed in as “0002228200333395”.
RESET Button
Initiate the Factory Reset procedure by pressing the RESET button located in back panel of the device for
approximately 8 seconds. All port LEDs will turn on and device will restart itself.
NOTE:
1. Factory Reset will be disabled if the “Lock keypad update” is set to “Yes”.
2. Please be aware by default the HT503 WAN side HTTP access is disabled. After a factory reset, the
device’s web configuration page can be accessed only from its LAN port.
3. If the HT503 was previously locked by your local service provider, pressing the RESET button will
only restart the unit. The device will not return to factory default settings.
Grandstream Networks, Inc.
HT503 User Manual
Page 35 of 35
Firmware 1.0.0.6
Last Updated: 6/2007
Download from Www.Somanuals.com. All Manuals Search And Download.
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