Grandstream networks Ht-503 User manual

HT503 USER MANUAL

Grandstream Networks, Inc.

HT503

FXS/FXO Port

Analog Telephone Adaptor

FIRMWARE VERSION 1.0.6.8 HT503 USER MANUAL Page 2 of 48

HT503 User Manual

Index

GNU GPL INFORMATION...........................................................................4

CHANGE LOG.............................................................................................5

CHANGES FROM 1.0.5.10 USER MANUAL...........................................................................................5

WELCOME ..................................................................................................6

SAFETY COMPLIANCS...........................................................................................................................6

WARRANTY .............................................................................................................................................6

CONNECT YOUR HT503.............................................................................8

EQUIPMENT PACKAGING......................................................................................................................8

CONNECTING THE HT503......................................................................................................................8

PRODUCT OVERVIEW.............................................................................11

SOFTWARE FEATURES OVERVIEW...................................................................................................11

HARDWARE SPECIFICATION..............................................................................................................12

BASIC OPERATIONS ...............................................................................13

UNDERSTANDING HT503 VOICE PROMPT........................................................................................13

PLACING A PHONE CALL.....................................................................................................................14

PHONE OR EXTENSION NUMBERS ...............................................................................................14

DIRECT IP CALLS .............................................................................................................................14

CALL HOLD............................................................................................................................................15

CALL WAITING.......................................................................................................................................15

CALL TRANSFER ..................................................................................................................................16

3-WAY CONFERENCING......................................................................................................................16

PSTN PASS THROUGH ........................................................................................................................17

VOIP-TO-PSTN CALLS..........................................................................................................................17

PSTN-TO-VOIP CALLS..........................................................................................................................18

ROUTE CALLS TO PSTN......................................................................................................................19

FORWARD CALLS TO PSTN................................................................................................................19

FORWARD CALLS TO VOIP.................................................................................................................19

ONE STAGE DIALING ...........................................................................................................................19

FAX SUPPORT.......................................................................................................................................20

CALL FEATURES .....................................................................................21

CONFIGURATION GUIDE.........................................................................22

CONFIGURING HT503 THROUGH VOICE PROMPT..........................................................................22

CONFIGURING HT503 WITH WEB BROWSER...................................................................................23

DEFINITIONS.........................................................................................................................................24

SAVING THE CONFIGURATION CHANGES........................................................................................43

REBOOTING FROM REMOTE..............................................................................................................43

CONFIGURATION THROUGH A CENTRAL SERVER.........................................................................43

SOFTWARE UPGRADE............................................................................44

FIRMWARE UPGRADE THROUGH TFTP/HTTP/HTTPS.....................................................................44

CONFIGURATION FILE DOWNLOAD...................................................................................................45

FIRMWARE AND CONFIGURATION FILE PREFIX AND POSTFIX ....................................................45

MANAGING FIRMWARE AND CONFIGURATION FILE DOWNLOAD.................................................45

RESTORE FACTORY DEFAULT SETTING..............................................47

FIRMWARE VERSION 1.0.6.8 HT503 USER MANUAL Page 3 of 48

TABLE OF FIGURES

HT503 USER MANUAL

Figure 1: CONNECTING THE HT503..............................................................................................9

Figure 2: INTERCONNECTION DIAGRAM OF THE HT503.........................................................10

Figure 3: UPLINK/DOWNLINK BANDWIDTH LIMITATION..........................................................27

TABLE OF TABLES

HT503 USER MANUAL

Table 1: DEFINITIONS OF THE HT503 CONNECTORS ..............................................................9

Table 2: HT503 LED DEFINITIONS ...............................................................................................9

Table 3: HT503 TECHNICAL SPECIFICATIONS.........................................................................11

Table 4: HT503 HARDWARE SPECIFICATION...........................................................................12

Table 5: HT503 IVR MENU DEFINITIONS...................................................................................13

Table 6: HT503 CALL FEATURE DEFINITIONS .........................................................................21

Table 7: STATUS PAGE...............................................................................................................24

Table 8: BASIC SETTINGS..........................................................................................................25

Table 9: ADVANCED SETTINGS.................................................................................................28

Table 10: FXS PORT SETTINGS.................................................................................................31

Table 11: FXO PORT Settings......................................................................................................37

TABLE OF GUI INTERFACES

HT503 USER MANUAL

(http://www.grandstream.com/products/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

FIRMWARE VERSION 1.0.6.8 HT503 USER MANUAL Page 4 of 48

GNU GPL INFORMATION

HT503 firmware contains third-party software licensed under the GNU General Public License (GPL).

Grandstream uses software under the specific terms of the GPL. Please see the GNU General Public

License (GPL) for the exact terms and conditions of the license.

Grandstream GNU GPL related source code can be downloaded from Grandstream web site from:

http://www.grandstream.com/support/faq/gnu_gpl .

FIRMWARE VERSION 1.0.6.8 HT503 USER MANUAL Page 5 of 48

CHANGE LOG

This section documents significant changes from previous versions of HT503 user manuals. Only major

new features or major document updates are listed here. Minor updates for corrections or editing are not

documented here.

CHANGES FROM 1.0.5.10 USER MANUAL

Add the option to change the Voice Frames per TX. [Voice Frames per TX]

Add a configuration parameter to override User-Agent header [Use SIP User-Agent Header]

Added new Prompt Tone and Prompt Tone Access Code [Prompt Tone Access Code]

Added Send SIP log in Syslog [Send SIP Log]

Add NTP update interval option [NTP Update Interval]

Added support for WebUI for Update and Apply Changes

Changed the device design to accept parameters without requiring reboot

FIRMWARE VERSION 1.0.6.8 HT503 USER MANUAL Page 6 of 48

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/products/ht_series/ht503/documents/ht503_usermanual_english.pdf

SAFETY COMPLIANCS

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.

FIRMWARE VERSION 1.0.6.8 HT503 USER MANUAL Page 7 of 48

This document contains links to Grandstream GUI Interfaces. Please remember to download these

examples from http://www.grandstream.com/products/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/products/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.

FIRMWARE VERSION 1.0.6.8 HT503 USER MANUAL Page 8 of 48

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.

FIRMWARE VERSION 1.0.6.8 HT503 USER MANUAL Page 9 of 48

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

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.

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.

FIRMWARE VERSION 1.0.6.8 HT503 USER MANUAL Page 10 of 48

Figure 2: INTERCONNECTION DIAGRAM OF THE HT503

Internet ADSL/Cable

Modem Ethernet

Fax

Cordless

Phone

Analog Phone

WAN

LAN

FXS

FXO

PSTN

Cloud

FIRMWARE VERSION 1.0.6.8 HT503 USER MANUAL Page 11 of 48

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

FIRMWARE VERSION 1.0.6.8 HT503 USER MANUAL Page 12 of 48

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, 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

FIRMWARE VERSION 1.0.6.8 HT503 USER MANUAL Page 13 of 48

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

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

“DHCP Mode”,

“Static IP Mode”

Press “9” to toggle the selection

If using “Static IP Mode”, configure the IP address information using

menus 02 to 05.

If using “Dynamic IP Mode”, all IP address information comes from

the DHCP server automatically after reboot.

“IP Address “ + IP address

The current WAN IP address is announced

If using “Static IP Mode”, enter 12 digit new IP address. You need

to reset the HT to take affect the new IP address.

“Subnet “ + IP address

Same as menu 02

“Gateway “ + IP address

Same as menu 02

“DNS Server “ + IP address

Same as menu 02

Preferred Vocoder

Press “9” to move to the next selection in the list:

PCM U / PCM A

iLBC

G.726

G.723

G.729

MAC Address

Announces the MAC address.

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.

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 / HTTPS

Firmware Version

Firmware version information.

Firmware Upgrade

Firmware upgrade mode. Press “9” to toggle among the following

three options:

- always check

- check when pre/suffix changes

- never upgrade

FIRMWARE VERSION 1.0.6.8 HT503 USER MANUAL Page 14 of 48

“Direct IP Calling”

Enter the IP address to make a direct IP call, after dial tone. (See

“Make a Direct IP Call”.)

Voice Mail

Number of voice mails

“RESET”

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

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 cannot 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).

FIRMWARE VERSION 1.0.6.8 HT503 USER MANUAL Page 15 of 48

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”

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.

FIRMWARE VERSION 1.0.6.8 HT503 USER MANUAL Page 16 of 48

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.

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.

FIRMWARE VERSION 1.0.6.8 HT503 USER MANUAL Page 17 of 48

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.

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.

FIRMWARE VERSION 1.0.6.8 HT503 USER MANUAL Page 18 of 48

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).

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.