Grandstream Networks 286 User Manual

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User Manual

HandyTone-286 Rev 3.0

Analog Telephone Adaptor

For Firmware Version 1.0.8.32

Grandstream Networks, Inc.

www.grandstream.com

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HandyTone -286 User Manual Grandstream Networks, Inc.

Table of Contents

1 WELCOME...................................................................................................................................... 4

2 INSTALLATION............................................................................................................................. 5

3 WHAT IS INCLUDED IN THE PACKAGE........ ........................................................................6

3.1 S

3.2 W

AFETY COMPLIANCES.................................................................................................................6

ARRANTY.................................................................................................................................. 6

4 PRODUCT OVERVIEW ................................................................................................................7

4.1 K

4.2 H

EY FEATURES............................................................................................................................. 7

ARDWARE SPECIFICATION .........................................................................................................8

5 BASIC OPERATIONS....................................................................................................................9

5.1 G

5.2 M

ET FAMILIAR WITH KEY PAD AND VOICE PROMPT ....................................................................9

AKE PHONE CALLS.................................................................................................................. 10

5.2.1 Calling Phone or Extension Numbers ................................................................................10

5.2.2 Direct IP Calls....................................................................................................................11

5.2.3 Call Hold ......... ... .......................................................................................... ... ...................11

5.2.4 Call Waiting.......................... ..............................................................................................11

5.2.5 Call Transfer....................................................................................................................... 12

5.2.5.1 Blind Transfer............................................................... .........................................................................................................12

5.2.5.2 Attended Transfer.......... ........................... ...................................................... ....................................... ........................... .....12

5.2.6 3-way Conferencing.................... ... ..................................................................................... 13

5.2.6.1 Bell style conference ....................... ........................... ....................................................... ....................................... .............13

5.2.6.2 Non-bell style conference .............................................. ....................................................... ................................................13

5.3 C

5.4 F

5.5 LED

ALL FEATURES.........................................................................................................................13

AX ............................................................................................................................................ 14

LIGHT PATTERN INDICATION............................................................................................. 14

6 CONFIGURATION GUIDE.........................................................................................................16

6.1 C

ONFIGURING HANDYTONE-286 IP THROUGH VOICE PROMPT..................................................16

6.1.1 DHCP Mode .......................................................................................................................16

6.1.2 STATIC IP Mode..................................... ... ............................................................ ... ..........16

6.1.3 TFTP Server Address.......................................................................................................... 16

6.2 C

ONFIGURING HANDYTONE-286 WITH WEB BROWSER .............................................................16

6.2.1 Access the Web Configuration Menu........................... ....................................................... 16

6.2.2 End User Configuration ................................................................................................... 177

6.2.3 Advanced User Configuration............................ ...... ...... ... ...... ...... ... ...... ...... ... ...... ...... ...... . 22

6.2.4 Saving the Configuration Changes..................................................................................... 33

6.2.5 Rebooting the HandyTone-286 from Remote .....................................................................33

6.3 C

ONFIGURATION THROUGH A CENTRAL SERVER........................................................................34

7 SOFTWARE UPGRADE WITH TFTP ......................................................................................35

7.1 F

7.2 C

7.3 F

IRMWARE UPGRADE THROUGH TFTP/HTTP............................................................................35

ONFIGURATION FILE DOWNLOAD ............................................................................................ 36

IRMW ARE AND CONFIGURATION FILE PREFIX AND POSTFIX.....................................................36

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7.4 M

8 RESTORE FACTORY DEFAULT SETTING...........................................................................37

9 GLOSSARY OF TERMS............. ............................................................ ... ................................ 388

ANAGING FIR MWARE AND CONFIGURATION FILE DOWNLOAD ............................................... 36

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HandyTone -286 User Manual Grandstream Networks, Inc.

1 Welcome

Congratulations on becoming an owner of HandyTone-286. You made an excellent choice and we hope you will enjoy all its capabilities.

Grandstream's award-wining HandyTone-286 is innovative Analog Telephone Adaptor that offers a rich set of functionality and superb sound quality at ultra-affordable pr ice. They are fully compatible with SIP industry standard and can interopera te w ith m a ny other SIP compliant devices and software on the market.

This document is subject to changes without notice. The latest electronic versio n of this user manual can be downloaded from the following location:

http://www.grandstream.com/user_manuals/HandyTone.pdf

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HandyTone -286 User Manual Grandstream Networks, Inc.

2 Installation

HandyTone-286 is a VoIP Analog Telephone Adaptor designed to work with an ordinary analog telephone. The following photo illustrates the appearance of a HandyTone-286.

RJ45

10M Ethernet

Interconnection Diagram of the HandyTone-286:

Internet ADSL/Cable

Modem Ethernet

LAN

BUTTON/

RED LED/

GREEN LED

+5V/1200mA

RJ11

Telephone

Analog Phone

PHONE

Cordless Phone

Fax

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3 What is Included i n the Package

The HandyTone-286 package co ntains:

1) One HandyTone-286

2) One universal power adaptor

3) One Ethernet cable

3.1 Safety Compliances

The HandyTone-286 is com plia nt with various safety standards including FCC/CE and C-Tic k. Its power adaptor is compliant with UL standard. The HandyTone ATA should only operate with the universal power adaptor provided in the package.

3.2 Warranty

Grandstream has a reseller agreement with our reseller customer. End users should contact the company from whom you purchased the product 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.

Grandstream reserves the right to remedy warra nty policy without prior notification.

Warning: Please do not attempt to use a different power adaptor. Using other power adaptor may damage the HandyTone-286.

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 Manu al, could void your manufacturer warranty.

Information in this document is subject to change without notice. No part of this document may be reproduced or transmitted in any form or by any means, electronic or mechanical, for any purpose without the express written permission of Grandstream Networks, Inc..

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4 Product Overview

4.1 Key Features

• Supports SIP 2.0(RFC 3261), TCP/UDP/IP, RTP/RTCP, HTTP, ICMP, ARP/RARP, DNS,

DHCP (both client and server), NTP, PPPoE, STUN, TFTP, etc.

• Powerful digital signal processing (DSP) to ensure superb audio quality; advanced adaptive

jitter control and packet loss concealment technology

• Support various codecs including G.711 (PCM a-law and u-law), G.723.1 (5.3K/6.3K), G.726

(32K), G.729A and iLBC.

• Support Caller ID/name display or block, Call waiting caller ID, Hold, Call Waiting/Flash, Call

Transfer, 3-way conference (on Rev. 2.0), Call Forward, in-band and out-of-band DTMF, etc.

• Support fax pass throu gh (fo r PCMU and PCMA) an d T.3 8 FoIP (Fax over IP)

• Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise

Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control)

• Support standard encryption and authentication (DIGEST using MD5 and MD5-sess)

• Support for Lay er 2 (802.1Q VLAN, 802.1 p) and Layer 3 QoS (ToS, DiffServ, MPLS)

• Support automated NAT traversal without manual manipulation of firewall/NAT

• Support device configur atio n via built-in IVR, Web browser or encrypted configuration fi les

through TFTP or HTTP server

• Support firmware upgrade via TFTP or HTTP

• Support SIP Session Timer

• Support Syslog (on Rev.2.0)

• Ultra compact (wallet size) and lightweight design, great companion for travelers

• Compact, lightweight Universal Power adaptor

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4.2 Hardware Specification

The table below lists the hardware specification of HandyTone-286.

Model

LAN interface 1xRJ45 10Base-T Button 1 LED GREEN & RED color Universal

Power Adaptor

Dimension 65mm (W)

Weight 0.57 lbs (0.26kg) Operating

Temperature Humidity 10% - 95%

Compliance FCC/CE/C-Tick

HandyTone-286

Input: 100-240VAC Output: +5VDC, 1200mA UL certified

93mm (D) 27mm (H)

32 - 104oF 0 - 40oC

(non-condensing)

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5 Basic Operations

5.1 Get Familiar with Key Pad and Voice Prompt

HandyTone-286 stores a voice prompt menu for quick browsing and sim ple configuration. To enter this voice prompt menu, simply press the button on the HandyTone-286 or pick up the phone and dial “***”. The following table show s how to use the voice prompt menu to configure the device.

Menu Voice Prompt User’s Options

Main Menu “Enter a Menu Option” Enter “*” for the next menu option

Enter “#” to return to the main menu Enter 01 – 06, 47, 86 or 99 Menu option

01 “DHCP Mode”, or

“Static IP Mode”

02 “IP Address “ + IP address The current WAN IP address is announced

03 “Subnet “ + IP address Same as Menu op tion 02 04 “Gateway “ + IP address Same as Menu option 02 05 “DNS Server “ + IP address Same as Menu option 02 07 Preferred Vocoder Enter “9” to go to the next selection in the list:

12 WAN Port Web Access Enter “9” to toggle between:

13 Firmware Server IP Address The current Firmware Server IP address is

14 Configuration Server IP

Address

15 Upgrade Protocol Upgrade protocol for firmware and

Enter ‘9’ to toggle the selection

If user selects “Static IP Mode”, user need configure all the IP address information through menu 02 to 05. If user selects “Dynamic IP Mode”, the device will retrieve all IP address information from DHCP server automatically when user reboots the de v i ce.

Enter 12-digit new IP address if in Static IP Mode.

- PCM U

- PCM A

- G-723

- G-729

- iLBC

- G-726

- enable

- disable

announced. Enter 12 digit new IP address. The current Config Server Path IP address is announced. Enter 12 digit new IP address.

configuration update. Enter “9” to toggle between:

- TFTP

- HTTP

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16 Firmware Ver s i on Firmware version informat ion. 17 Firmware Upgrade Firmware upgrade mode. Enter “9” to rotate

among the following three options:

- always check

- check when pre/suffix changes

- never upgrade

47 “Direct IP Calling” When entered, user will be prompted a dial

tone, dial a 12-digit IP address to make a direct IP call. (For details, see “4.2.2 Make a Direct IP Call”.)

99 “RESET” Enter “9” to reboot the device; or

Enter MAC address to restore factory default setting (For details, see section 8.)

“Invalid Entry” Automaticall y returns to Main Menu

NOTES:

• Once the LED button is pressed, it enters voice prompt main menu. If the button is pressed

again while it is already in the voice prompt menu state, it jumps to “Direct IP Calling” option and dial tone pla ys in th i s state

• “*” 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. Once all digits are accumulated, the device will automatically process them

• For IP address input, omit the dot and enter the digits directly, add 0 for those octets with less

than three digits. e.g.: IP: 192.168. 1.10, key in: 192168001010

• Key entry cannot be deleted but the phone may prompt error once it is detected

5.2 Make Phone Calls

5.2.1 Calling Phone or Extension Numbers

To make a phone or extension number call:

a) Dial the number directly and wait for 4 seconds (Default “No Key Entry Timeout”). Or b) Dial the number directly, a nd press # (assuming that “Use # as dial key” is selected in 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 might need to enter in some prefix number followed by the phone number. Please check with your VoIP service provider to get the information. If

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you phone is assigned with a PSTN-like number such as 6265556789, most likely you just follow the rule to dial 16266667890 as if you were calling from a regular analog phone, followed by pressing # or wait for 4 seconds.

5.2.2 Direct IP Call s

Direct IP calling allows two parties, that is, a HandyTone with an analog phone and another VoIP Device, to talk to each other in an ad hoc fashi on withou t a SIP prox y. This kind of VoIP calls can be made between two parties if:

• Both HandyTone ATA and other VoIP De vice(i.e., another HandyTo ne AT A or Budgetone SIP

phone or other VoIP unit) ha ve public IP addresses, or

• Both HandyTone ATA and other VoIP Device are on the same LAN using private IP addresses,

• Both HandyTone ATA and other VoIP Device can be connected through a router using public

or private IP addresses (with necessary port forwarding or DMZ).

To make a direct IP call, first pick up the analog phone or turn on the speakerphone on the analog phone, then access the voice me nu prompt by dial “***” or press the button on the HandyTone-286, and dials “47” to access the direct IP call menu. User will hear a voice prompt “Direct IP Calling” and a dial tone. Enter a 12-digit target IP address to make a call. Destination ports can be specified b y using “*4” (encoding for “:”) followed by the port number.

Examples:

If the target IP address is 192.168.0.160, the dialing convention is Voice Prompt with option 47, then 192168000160 followed by pressing the “#” key if it is configured as a send key or wait 4 seconds. In this case, the default destination port 5060 is used if no port is specified.

If the target IP address/port is 192.168.1.20:5062, then the dialing convention would be: Voice Prompt with option 47, then 192168001020 *45062 followed by pressing the “#” key if it is configured as a send key or wait for 4 seconds.

5.2.3 Call Hold

While in conversation, pressing the “flash” button on the attached phone will put the remote end on hold. Pressing the “flash” button again will release the previously held party and the bi-directional media will resume.

5.2.4 Call Waiting

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If call waiting feature is enabled, while the user is in a conversation, he will hear a speci al stutter tone if there is another incoming call. User can press the flash button to put the current call party on hold and switch to the other call. Pressing flash button toggles between two active calls.

5.2.5 Call Transfer

5.2.5.1 Blind Transfer

Assuming that call party A and B are in conversation. A wants to Blind Transfer B to C:

1. A presses FLASH (on the analog phone, or Hook Flash for old model phones) to get a dial tone.

2. Then A dials *87 then dials C’s number, and then #(or wait for 4 seconds)

3. A can hang up.

NOTE: “Enable Call Feature” has to be set to “Yes” in web configuration page.

A can hold on to the phone and wait for one of the three following behaviors:

• A quick confirmation tone (temporarily using the call waiting indication tone) followed by a

dial tone. This indicates the transfer is successful (transferee has received a 200 OK from transfer target). At this point, A can either hang up or make another call.

• A quick busy tone followed by a restored call (on sup p ort ed platforms only). This means the

transferee has rece iv ed a 4xx response for the INVI TE a nd we will try to recover the call. The busy tone is just to indicate to the transferor that the transfer has failed.

• Busy tone keeps playing. This means we have failed to receive the second NOTIFY from the

transferee and decided to time out. Note: this does not indicate the transfer has been successful, nor does it indicate the transfer has failed. When transferee is a client that does not support the second NOTIFY (such as our own earlier firmware) , this will be the case. In bad network scenarios, this could also happen, although the transfer may have been c ompleted successfully.

5.2.5.2 Attended Transfer

Assuming that call party A and B are in conversation. A wants to Attend Transfer B to C:

1. A presses FLASH (on the analog phone , or Hook Flash for old model phones) to get a dia l tone

2. A then dial C’s number then # (or wait for 4 sec onds).

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:

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• When Attended Transfer failed, if A hangs up, the HandTone-286 will ring user A again to

remind A that B is still on the call. A can pick up the phone to restore conversation with B.

5.2.6 3-way Conferencing

HandyTone-286 supports 3-way conference in two modes, which is configurable on the web configuration.

5.2.6.1 Bell style conference

Assuming that call party A and B are in conversation. A wants to bring C in a conference:

1. A presses FLASH (on the analog phone, or Hook Flash for old model phones) to get a dial tone.

2. A dials C’s number then # (or wait for 4 seconds).

3. If C answers the call, then A presses “flash” to bring B, C in the conference.

4. If C does not answer the call, A can press “flash” back to talk to B.

5.2.6.2 Non-bell style conference

Use *23 as conference code to initiate conference. Assuming that call party A and B are in conversation. A wants to bring C in a conference:

5. A presses FLASH (on the analog phone, or Hook Flash for old model phones) to get a dial tone.

6. A dials *23 followed by C’s number then # (or wait for 4 seconds).

7. If C answers the call, then A presses “flash” to bring B, C in the conference.

8. If C does not answer the call, A can press “flash” back to talk to B.

5.3 Call Features

Following table shows the ca l l fea t ures of HandyTone-286.

Key Call Features

*23 3-way conference code. *87 Bland transfer *30 Block Caller ID (for all subsequent calls) *31 Send Caller ID (for all subsequent c a lls) *67 Block Caller ID (per call) *82 Send Caller ID (per call) *50 Disable Call Waiting (for all subsequent calls) *51 Enable Call Waiting (for all subsequent ca lls)

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*70 Disable Call Waiting. (Per Call) *71 Enable Call Waiting (Per Call) *72 Unconditional Call Forward.

To use this feature, dial “*72” and get the dial tone. Then dial the forward number and “#” for a dial tone, then h ang up.

*73 Cancel Unconditional Call Forward

To cancel “Unconditional Call Forward”, dial “*73” and get the dial tone, then hang up.

*90 Busy Call Forward

To use this feature, dial “*90” and get the dial tone. Then dial the forward number and “#” for a dial tone, then h ang up.

*91 Cancel Busy Call Forward

To cancel “Busy Call Forward”, dial “*91” and get the dial tone, then hang up

*92 Delayed Call Forward

To use this feature, dial “*92” and get the dial tone. Then dial the forward number and “#” for a dial tone, then h ang up.

*93 Cancel Delayed Call Forward

To cancel this Forward, dial “*93” and get the dial tone, then hang up

Flash/Hook When in conversation, this action will switch to the new

incoming call if there is a call waiting indication. When in conversation without an incoming call, this action will switch to a new channel for a new call.

5.4 Fax

HandyTone-286 supports FAX in two modes: T.38 (Fax over IP) and 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. 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 will need to select all the Preferred Codecs to be PCMU/PCMA.

5.5 LED Light Pattern Indication

Following are the LED light pattern indications.

RED LED always indicates not abnormal status

DHCP Failed or WAN No Cable Button flashes every 2 seconds (if DHCP is

configured)

HandyTone-286 fails to register Button flashes every 2 seconds (if SIP server is

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configured) Firmware Upgrading Button flashes every 2 seconds Device Malfunctions Red light steady on

GREEN LED mostly indicates normal working status

Message Waiting Indication Button flashes every 2 seco nds RINGING Button flashes at 1/10 second RINGING INTERVAL Button flashes every second In Conversation Green light steady on

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6 Configuration Guide

6.1 Configuring HandyTone-286 IP through Voice Prompt

6.1.1 DHCP Mode

Follow section 5.1 with voice menu option 01 to enable H a ndyTone-286 to use DHCP.

6.1.2 STATIC IP Mode

Follow section 5.1 with voice menu option 01 to enable HandyTone-286 to use STATIC IP mode, then use option 02, 03, 04, 05 to set up IP address, Subnet Mask, Gateway and DNS server respectively.

6.1.3 TFTP Server Address

Follow section 5.1 with voice menu option 06 to configure the IP address of the TFTP server.

6.2 Configuring HandyTone-286 with Web Browser

HandyTone-286 has an embedded Web server that will respond to HTTP GET/POST requests. It also has embedded HTML pages that allows a user to configure the HandyTone-286 through a Web browser such as Microsoft’s IE and AOL’s Netscape.

6.2.1 Access the Web Configuration Menu

First, get the IP address of the HandyTone-286 through section 5.1 with menu option 02. Then access the HandyTone-286’s Web Configuration Menu using the following URI:

http:// HandyTone-IP-Address

where the HandyTone-IP-Address is the IP address of the HandyTone-286.

NOTE:

• To type IP address into browser to get into the configuration page, please strip out the leading

“0” as the browser will parse in octet. e.g.: if the IP address is: 192.168.001.014, please type in:

192.168.1.14.

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6.2.2 End User Configuration

Once this request is entered and sent from a Web browser, the IP phone will respond with the following login screen:

Grandstream Device Configuration

Password

The password is case sensi tive with maximum length of 25 characters. Th e factory default password for End User and admini strator is “123” and “admin” resp ectively. Only administrator can get access to “ADVANCED SETTING” configuration page.

NOTE:

If you cannot log into the configuration page by using default password, please check with the VoIP service provider. Most likel y, the service provider has provis ioned the device and configured fo r you and changed the default password.

• Status Page.

Grandstream Device Configuration

MAC Address:

IP Address:

Product Model:

Software Version:

System Up Time:

Registered:

PPPoE Link Up:

NAT:

NAT Mapped IP:

STATUS BASIC SETTINGS ADVANCED SETTING S

00.0B.82.08.74.D4

10.10.13.195 HT286 REV 3.0 Program-- 1.0.8.19 Bootloader-- 1.0.8.11 HTML-- 1.0.8.19 VOC--

1.0.0.10 0 day(s) 0 hour(s) 0 minute(s) Yes disabled detected NAT type is full cone

123.45.6.789 17

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NAT Mapped Port:

Total Inbound Calls:

Total Outbound Calls:

Total Missed Calls:

60946 0 0 0

Total Call Time (in

minutes):

Total SIP Message

Sent:

Total SIP Message

Received:

Total RTP Packet Sent:

Total RTP Packet

Received:

Total RTP Packet Loss:

MAC Address

IP Address Product Mod el

2 0 0 0

The device ID, in HEX format. This is very important ID for ISP troubleshooting.

IP address assigned on the device. This field contains the product mode l info.

Software Version

System Uptime

Registered

PPPoE Link Up

NAT

NAT Mapped IP

Program: This should be consistent with the firmware version. Bootloader: version number for bootloader, it could be lower than program

version.

HTML: This should be consistent with the firmware version. VOC: This is the codec program, normally not changed.

This shows system up time since last reboot.

This shows whether the unit is registe red to voip service provider’s server.

This shows whether the PPPoE is up if conne cted to DSL modem

This shows what kind of NAT the HandyTone-286 is behind.

WAN side mapped IP if HandyTone-286 is connected to a NAT router.

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NAT Mapp ed Port

Statistical Status

• Basic Settings.

End User Password: Web Port:

IP Address:

WAN side mapped port if HandyTone-286 is connected to a NAT router.

Self-explainable. Please refer to the page displayed.

Grandstream Device Configuration

STATUS BASIC SETTINGS ADVANCED SETTI NG S

(purposely not displayed for security protection)

(default for HTTP is 80)

dynamically assigned via DHCP (default) or PPPoE

(will attempt PPPoE if DHCP fails and following is non-blank)

DHCP hostname: DHCP domain: DHCP vendor class ID: PPPoE account ID: PPPoE password:

Preferred DNS server:

Time Zone:

Daylight Savings

Time:

statically configured as:

I P Address: Subnet Mask: Default Router: DNS Server 1: DNS Server 2:

GMT-5:00 (US Eastern Tim e, New York)

192.168 .0

160

No Yes (if set to Yes, display time will be 1 hour ahead of normal time)

Optional Rule:

4,1,7,2,0;10,-1,7,2,0;60

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Update

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End User Password

This contains the password to access the Web Configuration Menu. This field is case sensitive with max. 25 characters

Web Port

HandyTone -286 User Manual Grandstream Networks, Inc.

IP Address

This is the device’s internal HTTP server port. Default is 80. There are 2 modes under which the IP phone can operate:

- If DHCP mode is enabled, then all the field for the Static IP mode are not used (even though they are still saved in the Flash memory) and the IP phone will acquire its IP address from the DHCP server in the network.

To use PPPoE feature please set the PPPoE account settings if the HT-286 is connected directly to a DSL modem. The HT-286 will attempt to establish a PPPoE session if any of the PPPoE fields is set.

- If Static IP mode is selected, then the IP address, Subnet Mask, Default Router IP address, DNS Server 1 (mandatory), DNS Serve r 2 (optional) fields will need to be configured .

DHCP hostname

This option specifies the name of the client. This field is optional but may be required by some Internet Service Providers. Default is blank.

DHCP domain

This option specifies the domain name that client should use when resolving hostnames via the Domain Name System. Default is blank.

DHCP vendor class ID

This option is used by clients and servers to exchange vendor-specific information. Default is blank.

PPPoE account ID

PPPoE password Time Zone

Daylight Savings Time

PPPoE username. Fill this field if your ISP requires you to use a PPPoE (Point to Point Protocol over Ethernet) connection.

PPPoE account password. This parameter controls how date/time is displayed according to the specified

time zone. This parameter controls whether the displayed time will be daylight savings

time or not. If set to “Yes” and the Optional R ule is empt y, then the displayed time will be 1 hour ahead of normal time. The “Automatic Daylight Saving Time Rule” shall have the following syntax: start-time;end-time;saving Both start-time and end-time have the same syntax: month,day,weekday,hour,minute month: 1,2,3,..,12 (for Jan, Feb, .., Dec) day: [+|-]1,2,3,..,3 1 weekday: 1, 2, 3, .., 7 (for Mon, Tue, .., Sun), or 0 which means the daylight saving rule is not based on week days but based on the day of the month. hour: hour (0-23), minute: minute (0-59) If “weekday” is 0, it means the date to start or end daylight savin g is at exactly the given date. In that cas e, the “day” value must not be negative. If “weekday” is not zero and “day” is positive, then the daylight saving starts on the first “day” the iteration of the weekday (1st Sunday, 3rd Tuesday etc). If “weekday” is not zero and “day” is negative, then the daylight saving starts on the last “day”th iteration of th e weekday (last Sunday , 3rd last Tuesday etc). The saving is in the unit of minutes. The savin g time may also be preceded by a

21 negative (-) sign if subtrac tio n is desired instead of addition. The default value for “Automatic Daylight Saving Time Rule” shall be set to “04,01,7,02, 00;10,-1,7,02,00;60” which is the rule for US.

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6.2.3 Adv a nc e d Us e r Co nf i g u ra t io n

To login to the Advanced User Configuration page, follow the instruction in section 6.2.1 to get to the following login page. T he password is case sensit ive with a maximum len gth of 25 characters and the factory default pa ssw ord for A dvanced User is “admin”.

Grandstream Device Configuration

Admin Password:

SIP Server:

Outbound Proxy:

SIP User ID:

Authenticate ID:

Authenticate Password:

Name:

Advanced Options:

referred Vocoder:

(in listed order)

STATUS BASIC SETTINGS ADVANCED SETTING S

(purposel y not dis p l ayed for security protection)

sip.mycompany.com

(e.g., sip.mycompan y.com, or IP address)

(e.g., proxy.myprovider.com, or IP address, if any)

123456789

(the user part of an SIP address) (can be identical to or different from SIP User

ID) (purposel y not dis p l ayed for security

protection) (optional, e.g., John Doe)

choice 1: choice 2: choice 3: choice 4: choice 5: choice 6: choice 7:

current setting is " PCMU"

current setting is " PCMA"

current setting is " G723"

current setting is " G729"

current settin g i s " G 726-32"

current setting is " iLBC"

current setting is " PCMU"

G723 rate:

6.3kbps encoding rate 5.3kbps encoding rate

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iLBC frame size:

iLBC payload type:

Silence Suppression:

Voice Frames per TX:

ax Mode:

ayer 3 QoS: ayer 2 QoS:

Allow incoming SIP

messages

from SIP proxy only:

Use DNS SRV:

User ID is phone

number:

SIP Registration:

20ms 30ms

(between 96 and 127, default is 97)

No Yes

(up to 10/20/32/64 for G711/G726/G723/other codecs

respectively) T.38 (Auto De tec t ) Pass-Through

802.1Q/VLAN Tag

(Diff-Serv or Precedence value)

802.1p priority value

No Yes

Yes No

(0-7)

Unregister On Reboot:

egister Expiration:

arly Dial:

Allow outgoing call

without Registration:

ial Plan Prefix:

No Key Entry Timeout:

Use # as Dial Key:

local SIP port:

local RTP port:

Use random port:

NAT Tra versal:

keep-alive interval:

Yes No

3600

(in seconds. defau lt 1 hou r, max 45 days)

No Yes (use "Yes" only if proxy supports 484 response)

No Yes

(this prefix string is added to each dialed number)

(in seconds, default is 4 seconds)

No Yes (if set to Yes, "#" will function as the Dial key)

5060

(default 5060)

5004

(1024-65535, default 5004) No Yes No

Yes, STUN server is: (URI or IP:port)

(in seconds, defau lt 20 seconds)

Use NAT IP

(used in SIP/SDP message if specified)

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Proxy-Require:

SUBSCRIBE for MWI:

Offhook Auto-Dial:

Enable Call Features:

Use Bell-style

3-way Conference:

Disable Call-Waiting:

Send DTMF:

TMF Payload Type :

Send Flash Event:

Onhook Threshold:

XS Impedance:

Caller ID Scheme:

No, do not send SUBSCRIBE for Message Waiting Indication Yes, send periodical SUBSCRIBE for Message Waiting Indication

(User ID/extension to dial automaticall y when offhook)

No Yes (if Yes, Call Forwarding & Call-Waiting-Disable are supported locally)

No Yes (if Yes, *23 will be disable d)

No Yes in-audio via RTP (RFC 2833) via SIP INFO

101

No Yes (Flash will be sent as a DTMF event if set to Yes)

800 ms

600 Ohm (N o rth Am erica)

Bellcore (North America)

Onhook Voltage:

Polarity Reversal:

NTP Server:

Send Anonymous:

Anonymous Method:

Time to ring:

Special Feature:

Syslog Server:

Syslog Level:

Session Expiration:

Min-SE:

Caller Request Timer:

36V

No Yes (reverse polarity upon call establishment and termination)

time.nist.gov

(URI or IP address) No Yes (caller ID will be blocked if set to Yes) Use From Header Use Privacy Header

60 seconds

Standard

NONE

180

(in seconds. default 180 s e conds) (in seconds. default and minimum 90 seconds)

Yes No (Request for timer when making outbound calls)

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Callee Request Timer:

orce Timer:

UAC Specify Refresher:

UAS Specify Refresher:

Force INVITE:

irmware Upgrade and

rovisioning:

Yes No (When caller supports timer but did not request one) Yes No (Use timer even when remote party does not support) UAC UAS Omit (Recommended) UAC UAS (W hen UAC did not specify refreshe r tag) Yes No (Always refresh with INVITE instead of UPDATE) Upgrade Via TFTP HTTP

Firmware Server Path: Config Server Path:

fm.grandstream.com/gs

Firmware File Prefix: Firmware File Postfix: Config File Prefix: Config File Postfix:

Automatic Upgrade: No Yes, check for upgrade every

1008

minutes (default 7

days)

irmware Key:

Authenticate Conf File:

ock keypad update:

Allow conf SIP Account

in Basic Settings:

Override MTU Size:

Volume Amplification:

Call Progress Tones:

Always Check for New Firmware Check New Firmware only when F/W pre/suffi x changes Always Skip the Firmware Check

(in Hexadecimal Representation) No Yes (cfg file woul d be authenticated before acc eptance

if set to Yes) No Yes ( configuration up date via keypad is dis abled if set

to Yes)

No Yes

0dB default

Dial Tone Recall Dial Tone

Frequency 1 (Hz)

350 440 0

350 440 10

Frequency 2 (Hz)

ON (x10ms) (C1;C2;C3)

OFF (x1 0ms) (C1;C2;C3)

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Admin Password

SIP Server

Message Waiting Confirmation Audible Rin gin g Busy Tone Reorder Tone

Receiver Offhook Tone

350 440 10

440 480 200

480 620 50

480 620 25

1400 2600 10

Update

400

Administrator password. Only administrator can configure the “Advanced Settings” page. Password field is purposely left blank for security reason after clicking update and save d. The maximum password length is 25 characters.

This field contains the URI string or the IP address. e.g. sip.my-voipprovider.com; 192.168.1.200:5066

Outbound Proxy

SIP Use r ID

Authenticate ID

Authenticate

Password

Name

Home NPA

Preferred Vocoder

This field contains the URI string or the IP of the outbound proxy. If there is no outbound proxy, this field SHOULD be left blank. If it is not blank, all outgoing requests will be sent to this outbound proxy.

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. Please do NOT include the preceding “sip:” scheme or the host portion of the SIP address in this field. I t is g i ven by VoIP service provider.

SIP service subscriber’s Authenticate ID used for authentication. It can be identical to or different from SIP User ID and given by VoIP service provider.

SIP service subscriber’s account password. It is given by VoIP service provider.

SIP service subscriber’s name which will be used for Caller ID display. Local area code for North American Dial Plan. HandyTone-286 supports up to 7 different vocoder types including G711-ulaw

(PCMU), G711-alaw (PCMA), G723, G729A, G726-32, and iLBC. Depending on the product model, some of these vocoders may not be provided in standard release. A user can configure vo coders in a preferen ce order that will be offered in SIP INVITE message.

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G723 Rate:

iLBC frame size

iLBC payload type

Silen ce Su pp r ess i on

Voice Frames per

This defines the encoding rate for G723 vocoder. By default, 6.3kbps rate is chosen.

This defines the size of the iLBC codec frame. The default setting is 20ms. This defines the iLBC payl oad type. The default setting is 97. The valid range

is between 96 and 127. This controls the silence su ppression/VAD feature o f G723 and G729. If set to

“Yes”, when a silence is detected, small quantit y of VAD packets (instead of audio packets) will be sent during the period of no talking. If set to “No”, this feature is disabled.

This field contains the number of voice frames to be transmitted in a single packet. When setting this value, the user should be aware of the requested packet time (used in SDP message) as a result of configuring this parameter. This parameter is associated with the first vocoder in the above vocoder Preference List or the actual used payload type negotiated between the 2 conversation parties at run time. e.g., if the first vocoder is configured as G723 and the “Voice Frames per TX” is set to be 2, then the “ptime” value i n the SDP message of an INVITE r equest will be 60ms because each G723 voice frame contains 30ms of audio. Similarly, if this field is set to be 2 and if the first vocoder chosen is G729 or G711 or G726, then the “ptime” value in the SDP message of an INVITE request will be 20ms. If the configured voice f r am es p er TX ex ceeds th e maximum allowed value, the phone will use and save the maximum allowed value for the corresponding first vocoder choice. The maximum value for PCM is 10(x10ms) frames; for G726, it is 20 (x10ms) frames; for G723, it is 32 (x30ms) frames; for G729/G728, 64 (x10ms) and 64 (x2. 5ms ) frames respectively.

Fax Mode

Layer 3 QoS

Layer 2 QoS

Allow incoming SIP

messages from SIP

proxy only

Use DNS SRV

T.38 (Auto Detect) FoIP by default, or Pass-Through (must use codec PCMU/PCMA)

This field defines the layer 3 QoS parameter which can be the value used for IP Precedence or Diff-Serv. Default value is 48

This setting includes two fields. The 802.1Q/VLAN Tag contains the value used for layer 2 VLAN tag. Default setting is blank. And 802.1p priority value contains the value of the priority value.

If set to “Yes”, the device will ignore any SIP message that does not come from the IP address (Source IP in the IP header) that it is registered to. Default is No.

This parameter controls whether the IP phone supports the DNS SRV route function.

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User ID is phone

number

SIP Registration

Unregister On

Reboot

Registration

Expiration

Early Dial

If the HandyTone-286 has an assi gned PSTN tel ephone nu mber, then t his field will 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.

This parameter controls whether the IP phone needs to send REGISTER messages to the proxy server. The default setting is “Yes”.

Default is No. If set to yes, the device will first send registration request to remove previous bindings.

This parameter allows the user to specify the time frequency (in minutes) the phone will refresh its registration with the specified registrar. The default interval is 3600 seconds (or 1 ho ur). The maximum interval is 45 day s.

This parameter controls whether the phone will attempt to send an early INVITE each time a key is pressed when a user dials a numbe r. If set to “Yes”, an INVITE is sent using the dial-number collected thus far; Otherwise, no INVITE is sent until the “(Re-)Dial” button is pressed or after about 5 seconds have elapsed if the user forgets to press the “(Re-)Dial” button. The “Yes” option shoul d be us ed ON LY if th ere is a SIP proxy configured and the proxy server supports 484 Incomplete Address response. Otherwise, the call will most likely be rejected by the proxy (with a 404 Not Found error). Please note that this feat ure is NOT desi gned to work with and sho uld NOT be enabled for direct IP-to-IP calling.

Dial Plan Prefix

No Key Entry

Timeout

Use # as

Send Key

Local SIP port

Local RTP port

Use Random Port

This value contains the dial plan prefix string (typically an ASCII numeric string). If it is not blank, then this string will added to the dialed number.

Default is 4 seconds.

This parameter allows the user to configure the “#” key to be used as the “Send”(or “Dial”) ke y. Once set to “Yes”, pressing this key will immediately trigger the sending of dialed string collected so far. In this case, this key is essentially equivalent to the “(Re)Dial” key. If set to “No”, this # key will then be included as part of the dial string to be sent out.

This parameter defines the local SIP port the IP phone will listen and transmit on. The default value is 5060.

This parameter defines the local RTP-RTCP port pair the IP phone will listen and transmit on. 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 po rt_value+3 for its RTCP. The default value is 5004.

This parameter, when set to Yes, will force random generation of both the local SIP and RTP ports. This is usually necessary when multiple IP phones are behind the same NAT.

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NAT Traversal

keep-alive interval

Use NAT IP

Proxy-Require

SUBSCRIBE for

MWI

Offhook

Auto-Dial

This parameter defines whether the phone NAT traversal mechanism will be activated or not. If activated (b y choosing “Yes”) and a STUN server is also specified, then the phone will behave according to the STUN client specification. Under this mode, the embedded STUN client inside the phone will attempt to detect if and what type of firewall/NAT it is behind by sen din g appropriate request to the specified STUN server. If this field is set to “Yes” with no specified STUN server, then the phone will only periodically (every 20 seconds by default) send a blank UDP packet (with no payload data) to the SIP ser ver to keep the mapped port open on the NAT.

The HandyTone-286 sends a UDP package to the SIP server periodically in order to keep the port open on the router. This parameter defines the interval time that HT286 send the UDP package. The def ault se tting is 20 seco nd.

NAT IP address is used in SIP/SDP message. Default is blank. SIP Extension to notify SIP server that the unit is behind the NAT/F irewall. Default is “No”. When set to “Yes” a SUBSCRIBE for Message Waiting

Indication will be sent periodically This parameter allows the user to configure a User ID or extension n umber to

e automatically dialed upon offho ok. Please note that only the user part of a SIP address needs to be entered here. The phone will automatically append the “@” and the host portion of the corresponding SIP addre ss.

Enable Call Feature

Use Bell-style 3-way

conference

Disable Call

Waiting

Send DTMF

DTMF Payload

Type

Send Flash Event

FXS Impedance

Default is YES. If set to “Yes”, call featu res ar e su pported locall y, such as callwaiting, transfer, 3-way conference, etc.

Conference mode, default option is No. If set to yes, the feature code for coference *23 would be disabled.

Default is No.

This parameter specifies the mechanism to transmit DTMF digit. There are 3 modes supported: in audio which means DTMF is combined in audio signal (not very reliable with low-bit-rate codec), via RTP (RFC2833), or via SIP INFO.

This parameter sets the pa ylo ad type for DTMF using RFC2833

Default is NO. If set to yes, flash will be sent as DTMF event. Selects the impedance of the analog telephone connected to the Phone port.

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Caller ID Scheme

Onhoo k Volt ag e

Polarity Reversal

NTP server

Send Anonymous

Anonymous Method

Select the Caller ID Scheme to suit the standard of different area.

• Bellcore (North America)

• CID (Canada)

• DTMF (Brazil)

• DTMF (Denmark)

• DTMF (Sweden)

• ETSI-FSK (France, Germany, Norway, Taiwan, UK-CCA)

• ETSI-DTMF (Finland, Sweden) Select the onhook voltage to suit diffe rent area or PBX.

Select Polarity Reversal to adapt some call charge/billing system. Default is No.

This parameter defin es the URI or IP address of t he NTP server which the IP phone will use to display the current date/time.

If this parameter is set to “Yes”, the device is employing the mechanism to block its ID.

If it is set to “Use from header”. Callers’ SIP user ID will be sent as anonymous, essentially block the Caller ID from displaying. If it is set to “User privacy header”, the SIP INVIT E message con tains a “pr ivacy” hea der, and the s erver bloc ks the calle r ID from the ca lled party.

Time to ring

Special Features

Syslog Server

Allow user to adjust the ring time of the phone. Default is 60 seconds. Default is Standard. Choose the selection to meet some special requirements

from Soft Switch vendors like Nortel, Broadsoft, etc. The IP address or URL of System log server. Thi s feature is especially useful

for ITSP (Internet Telepho ne Serv ice Prov id er)

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Syslog Level

Select the ATA to repor t the log level. Default is NONE. Th e level is one of DEBUG, INFO, WARNING or ERROR. Syslog messages are sent based on the following events:

• produc t model/version on boot up (INFO level)

• NAT related info (INFO level)

• sent or rece i ve d SIP message (DEBUG level)

• SIP message summary (INF O level)

• inbound an d 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 (ERR OR level)

The Syslog uses USER facility. In addition to standard Syslog pay load, it contains the following components:

GS_LOG: [device MAC address][err or code] error message Here is an example: May 19 02:40:38 192.168.1.14 GS_LOG: [00:0b:82:00:a1:be][ 000] Ethernet link is up

Session Expiration

Min-SE

Caller Request

Timer

Callee Request

Timer

Grandstream implemented SIP Session Timer. The session timer extension enables SIP sessions to be periodically “refreshed” via a re-INVITE request. Once the session interval expires, if there is no refresh via a re-INVITE message, the session will be terminated.

Session Expiration is the time (in seconds) at which the session is considered timed out, if no successful session refresh transaction occurs beforehand. The default value is 180 seconds.

The minimum session expiration (in seconds). The default value is 90 seconds. If selecting “Yes” the phone will use session timer when it makes outbound

calls if remote party supports session timer. If selecting “Yes” the phone will use session timer when it receives inbound

calls with session timer request.

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Force Timer

UAC Specify

Refresher

UAS Specify

Refresher

Force INVITE

Firmware Upgrade

and Provisioning

Firmware Server

Path

Config Server Path

Firmware File

Prefix

If selecting “Yes” the phone will use session timer even if the remote party does not support this feature. Selecting “No” will allow the phone to enable session timer only when the remote party support this feature. To turn off Session Timer, select “No” for Caller Request Timer, Callee Request Timer, and Force Timer.

As a Caller, select UAC to use th e phone as the refresher, or UAS to use the Callee or proxy server as the refresher.

As a Callee, select UAC to use caller or proxy server as th e refresher, or UAS to use the phone as the refresher.

Session Timer can be refreshed using INVITE method or UPDATE method. Select “Yes” to use INVITE method to refresh the session timer.

Default HTTP. Firmware upgrading may take up to 10 minutes depends on network environment. Do not interrupt the firmware upgrading process.

IP address or domain name of fir mwa re se rv er.

IP address or domain n a me o f con figuration server Default blank. If it is configured, HT286 rev. 3.0 will request the firmware file

with the prefix. Useful f or ITSPs. End user should keep it blank.

Firmware File

Postfix

Config File Prefix

Config File Postfix

Automatic Upgrade

Firmware K ey

Authenticate Conf

File

Lock keypad

update

Allow conf SIP

Account

in Basic Settings

Override MTU size

Default is blank. End user should keep it blank.

Default is blank. End user should keep it blank. Default is blank. End user should keep it blank. Default is YES. 32 digit in Hexadecimal Representation. Useful for ITSP to encrypt firmware.

End user should keep it blank. Default NO. End user should use default setting.

If this parameter is set to “Yes”, except for IVR MENU items 1 to 5, the configuration update via keypad is disabled.

If set to Yes, these four fields: SIP User ID, Authenticate ID, Authenticate Password and Name will be included in Basic Settings configuration page.

Override the MTU size.

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Volume

Amplification

Handset volume adjustment. RX is for receiving volume, TX is for transmission volume. Default values are 0dB for both parameters. +6dB generates the highest volume and -6dB generates the lowest volume.

Call Progress Tones

Using these settings, users can configure various call progress tone frequencies and cadences according to their country standard. By default they are set to North American standard. 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 ring, OFF should be zero. Otherwise it will ring ON ms and a pause of OFF ms and then repeat the pattern. Up to three cadences are supported.

6.2.4 Saving the Configuration Changes

Once a change is made, the user should press the “Update” button in the Configura tion Menu. The IP phone will then display the fo llow ing screen to c onfir m that the changes have been saved.

Grandstream Device Configuration

STATUS BASIC SETTINGS ADVANCED SETTINGS

Your configuration changes have been saved.

They will take effect on next re boot.

Users are recommended to power cycle the HandyTone-286 after seeing the above message.

6.2.5 Rebooting the HandyTone-286 from Remote

The administrator of the phone can remotely reboot the phone by pressing the “Reboot” button at the bottom of the configuration menu. Once done, the following screen will be displayed to indicate that rebooting is under way.

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Grandstream Device Configuration

The device is rebooting now...

You may relogin by clicking on the link below in 30 seconds.

Click to relogin

At this point, the user can relogin to the phone after waiting for about 30 sec onds.

6.3 Configuration through a Central Server

Grandstream HandyTone ATA can be automatically configured from a central provisioning system.

When HandyTone ATA boot up, it will send TFTP or HTTP request to download configuration file “cfg000b82xxxxxx”, where “000b82xxxxxx” is the MAC address of the HandyTone ATA. The configuration file can be loaded into devices via TFTP or HTTP from the central provisioning server, so the service provider or an enterprise with large deployment of HandyTone ATAs can easily manage the configuration and service provision to individual devices remotely .

Grandstream provides a lic e nse d provisioning system called GAPS that can be used to support automated configurat io n of HandyTone ATA. GAPS (Grandstr eam Automated Provisioning System) uses enhanced (NAT friendly) TFTP or HTTP (thus no NAT issues) an d other communication protocols to communicate with each individual HandyTone ATA 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 ea ch device, GAPS provision the devices with redire c tion 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 templates can be downloaded fro m http://www.grandstream.com/DOWNLOAD/Configuration_Tool/.

For details on how GAPS works, please refer to the docum e ntation of GAPS product.

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7 Software Upgrade with TFTP

Software upgrade can be done via either TFTP or HTTP . The corresponding configur a tion settings are in the ADVANCED SETTINGS configuration page.

7.1 Firmware Upgrade through TFTP/HTTP

To upgrade via TFTP or HTT P, the “Firmware Upgrade and Provisioning up grade 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, s erver name can be in eith er FQDN or IP address format. He re are examp les of some valid URL. e.g. firmware.mycompany.com:6688/Grandstream/1. 0.8.19

e.g. 168.75.215.189

NOTES:

• TFTP server in IP address format c an be configured vi a IVR. Please refer to section 6.1 .3 for

instructions. If TFTP serve r is in FQDN format, it must be set v ia web configuration interface.

• Once a “Firmware Server Path” is set, user needs to update the settings and reboot the device.

If the configured firmware server is found and a new code image is available, the HandyTone ATA will attempt to retrieve the new image files by downloading them into the HandyTone ATA’s SRAM. During this stage, the HandyTone ATA’s LEDs will blink until the checking/downloading process is completed. Upon verification of checksum, the new code image will then be sav ed into the Flash. If TFTP/HTTP fail s for any r eason (e.g., TFTP/HTTP server is not responding, there are no code image files available for upgrade, or checksum test fails, etc), the HandyTone ATA will st op the TFTP/HTTP process and simply boot using the existing code image in the flash.

• Firmware upgrade may take as long as 1 to 20 minutes over Internet, or just 20+ seconds if it is

performed on a LAN. It is recommended to conduct firmware upgrade in a controlled LAN environment if possible. For users who do not have a local firmware upgrade server, Grandstream provides a NAT-friendly TFTP server on the public Internet for firmware upgrade. Please check the Services section of Grandstream’s Web site to obtain our public TFTP server’s IP address.

• Alternatively, user can download a free TFTP or HTTP server and conduct local firmware

upgrade. A free windows version TFTP server is available for download from http://support.solarwinds.net/updates/New-customerFree.cfm downloaded from http://www.grandstream.com/y-firmware.htm. Unzip the file and put all of them under the root directory of the TFTP server. Put the PC running the TFTP server and the HandyTone ATA in the same LAN segment. Please go to File -> Configure -> Security to change the TFTP server's default setting from "Receive Only" to "Transmit Only" for the firmware upgrade. Start the TFTP server, in the HandyTone ATA’s web configuration page, configure the Firmware Server Path with the IP address of the PC, update the change and reboot the unit. Please be advised that our client will pull out firmware from the WAN side, if

. Our latest official release can be

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the TFTP server is connected to the device’s LAN port, the firmware upgrade will not work by design.

7.2 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 “Confi g Server Path” can be same or different from the “Firmware Server Path”.

A configuration parameter i s associated with each particular field in the web configuration pa ge. 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 ADVANC ED SETTINGS pa ge. For a detailed parameter list, plea s e refer to the corresponding fi rmware 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.

7.3 Firmware and Configuration File Prefix and Postfix

Starting from firmware version 1.0.7.11 for HandyTone-286 Rev 3.0, adding prefix and postfix for both firmware and configuration file is supported.

Firmware Prefix and Postfi x allows device to download the fi rmware name with the matchin g 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 allow s de vice 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 Firmwar e only when F/W pre/suffix ch anges” is set to “Yes”, the device will only issue firmware upgrade request if there are changes in the firmware Prefix or Postfix.

7.4 Managing Firmware and Configuration File Download

When “Automatic Upgr ade” is set to “Yes”, Serv ice Provider can use P19 3 (Auto Check Interval, in minutes, default and minimum is 60 minutes) to have the devices periodically check with either Firmware Server or Config Server, whenever they are defin ed. This allows the device periodically check if there are any new changes need to be taken on a scheduled time. By defining different intervals in P193 for different d evices, Server Provider c an spread th e Firmware or Configuratio n File download in minutes to reduce the Firmware or Provisioning Server load at any given time.

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8 Restore Factory Default Settin g

Warning !!!

Restore the Factory Default Setting will DELETE all configuration information of the device. 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 service provider.

Please disconnect network cable and power cycle the unit before trying to reset the unit to factory default. The steps are as follows:

Step 1:

Find the MAC address of the device. It is a 12 digits HEX number located on the bottom of the unit.

Step 2:

Encode the MAC address. Please use the following mapping:

0-9: 0-9 A: 22 B: 222 C: 2222 D: 33 E: 333 F: 3333

For example, if the MAC addre ss is 000b8200e395, it should be encoded as “0002228200333395”.

Step 3:

To perform factory reset:

a. Press “***” or the LED button for voice prompt. b. Enter “99” and get the voice prompt “Reset ”. c. Enter the encoded MAC address of the device. d. Wait for 15 seconds.

The device will reboot automatically and restore to factory default setting.

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9 Glossary of Terms

ADSL

Asymmetric Digital Subscriber Line: Modems attached to twisted pair copper wiring that transmit from 1.5 Mbps to 9 Mbp s downstream (to the subscriber) and from 16 kbps to 800 kbps upstream, depending on line distanc e.

AGC

Automatic Gain Control, is an electronic system found in many type s of devices. Its purpose is to control the gain of a system in order to maintain so me measure of performance over a changing range of real world conditio n s .

ARP

Address Resolution Protocol is a protocol used by the Internet Protocol (IP) pecifically IPv4, to map IP network addresses protocol. The protoc ol oper a t es below the network layer as a par t of the interface between the OSI network and OSI link layer. It is used when IPv4 is used over Ethernet

ATA

Analogue Telephone Adapter. Covert analogue telephone to be used in data netwo rk for VoIP, like Grandstream HT se ries products.

CODEC

Abbreviation for Coder- Deco der. It's an analog-to-digital (A/D) and digital-to-analog (D/A) converter for translating the signals from the outside world to digital, and back again.

CNG

to the hardware addresses used by a data link

[RFC826],

Comfort Noise Generator, geneate artificial background noise used in radio and wireless communications to fill the silent time in a transmission resulting from voice activity detection.

DATAGRAM

A data packet carrying its own address information so it can be inde pendently routed from its source to the destination computer

DECIMATE

To discard portions of a sig nal in order to reduce the amount of information to be encoded or compressed. Lossy compression algorithms ordinarily decimate while subsampling.

DECT

Digital Enhanced Cordless Tele c ommunications: A standard developed by the European Telecommunication Standard Institute from 1988, governing pan-European digital mobile telephony. DECT covers wireless PBXs, telepoint, residential cordless telephones, wireless access to the public switched telephone network, Closed User Groups (CUGs), Local Area

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Networks, and wireless local loop. The DECT Common Interface radio standard is a multicarrier time division multiple access, time division duplex (MC-TDMA-TDD) radio transmission technique usin g ten radio frequency channels from 1880 to 1930 MHz, each divided into 24 time slots of 10ms, and twe l ve full-duplex accesses per carrier, for a total of 120 possible combinations. A DECT base station (an RFP, Ra dio Fixed Part) can transmit all 12 possible accesses (time slots) simultaneously by using different frequencies or using only one frequency. All signaling information is transmitted from the RFP within a multiframe (16 frames). Voice signals are digitally encoded into a 32 kbit/s signal using Adaptive Differential Pulse Code Modula tion.

DNS

service that translates

DID

Short for Domain Name System (or Service or Server), an Internet

domain names

into IP addresses

Direct Inward Dialing Direct Inward Dialing. The ability for an outside caller to dial to a PBX extension without

going through an attendant or auto-attendant.

DSP

Digital Signal Processing. Using computers to process signals such as sound, video, and other analog signals which have be en converted to digital form.

Digital Signal Processor. A specialized CPU used for digital signal processing.

Grandstream products all have DS P chips bu ilt inside.

DTMF

Dual Tone Multi Frequency The standard tone-pairs used on telephone terminals for dialing using in-band signaling. The

standards define 16 tone-pai rs (0-9, #, * and A-F) although most terminals support only 12 of them (0-9, * and #).

FQDN Fully Qualified Domain Name

A FQDN consists of a host and domain name, includ ing top-level domain. For example,

www.grandstream.com

is a fully qualified domain name. www is the host, grandstream is the

second-level domain, and.com is the top level domain.

FXO

Foreign eXchange Office

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An FXO device can be an analog phone, answering machine, fax, or anything that handles a call from the telephone company like AT&T. They should also operate the same way when connected to an FXS interface.

An FXO interface will accept calls from FXS or PSTN interfaces. All countries and regions have their own standards.

FXO is complimentary to FXS (and the PSTN).

FXS

Foreign eXchange Station An FXS device has hardwar e to generate the ring signal to the FXO extension (usually an

analog phone). An FXS device will allow any FXO device to operate as if it were connected to the phone

company. This makes your PBX the POTS+PSTN for the phone. The FXS Interface connects to FXO devices (by an FXO interface, of cour se).

DHCP

The Dynamic Host Configuration Protocol (DHCP) is an Internet protocol for automat in g the configuration of computers that use TCP/IP. DHCP can be used to automatically assign IP addresses, to deliver TCP/IP stack configuration parameters such as the subnet mask and default router, and to provide othe r configuration information such a s the addresses for printer, time and news servers.

ECHO CANCELLATION

H.323

HTTP

Echo Cancellation is used in telephony to describe the process of removing echo from a voice communication in order to im prove voice quality on a telephone call. In addition to improving quality, this process im proves bandwidth savings achieved through silence suppression by preventing echo from traveling across a network.

There are two types of echo of relevance in telephony: acoustic echo and hybrid echo. Speech

compression techniques and digital processing delay often contr ibute to echo generation in telephone networks.

A suite of standards for multimedia conferences on traditional packet-switched networks.

Hyper Text Transfe r Protocol; the World Wide Web protoco l that performs the request and retrieve functions of a server

Internet Protocol. A packet -based protocol for delivering data across networks.

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IP-PBX

IP-based Private Branch Exchan ge

IP Telephony

(Internet Protocol telephony, also known as Voice over IP Telephony) A general term for the technologies that use the Internet Protocol's packet-switched connections to exchange voice, fax, and other forms of information that have traditionally been carried over the dedicated circuit-switched connections of the public switched telephone network (PSTN). The basic steps involved in originating an IP Te lephony call are conversion of the analog voice signal to digital format and compression/translation of the signal into Internet protocol (IP) packets for transmission over the Internet or other packet-switched networks; the process is reversed at the receiving end. The terms IP Telephony and Internet Te lephony are often used to me an the same; however, they are not 100 per cent intercha ngeable, since Internet is only a subcase of packet-switched networks. For users who have free or fixed-price Internet access, IP Telephony software essentially provides free telephone calls any where in the world. However, the challenge of IP Telephony is maintaining the quality of service expected by subscribers. Session border controllers resolve this issue by providing quality assurance comparable to legacy telephone systems.

IVR

IVR is a software application that accepts a combination of voice telephone input and touchtone keypad selection and provides appropriate responses in the form of voice, fax, callback, email and perhaps other media.

MTU

A Maximum Transmission Unit (MTU) is the largest size packet (eight-bit bytes), that can be sent in a packet- or frame-based network such as the Internet. The maximum for Ethernet is 1500 byte.

NAT Network Address Translation

NTP

Network Time Pro tocol, a protocol to exchange and synchronize tim e over networks

The port used is UDP 123 Grandstream products using NTP to get time from Internet

OBP/SBC

Outbound Proxy or another name Session Border Controller. A de vice used in VoIP OBP/SBCs are put into the signaling and media path between calling and called party. The OBP/SBC acts as if it was the called VoIP phone and places a second call to the called party. The effect of this behaviour is that not only the signaling traffic, but also the media traffic (voice, video etc) crosses the OBP/SBC. Without an OBP/SBC, the media traffic travels directly between the VoIP phones. Private OBP/SBCs are used along with firewalls VoIP calls to and from a protected enterprise network. Public VoIP service providers use

or frame, specified in octets

networks.

to enable

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connections

PPPoE

PSTN

RTCP

RTP

SDP

OBP/SBCs to allow the use of VoIP protocols from private networks with internet using NAT

Point-to-Point Protocol over Ethernet, is a network protocol for encapsulating PPP frames in Ethernet frames. It is used mainly with cable modem and DSL services.

Public Switched Telephone Ne twork i.e. the phone service we use f or every ordinary phone call, or called POT (Plain Old

Telephone), or circ uit sw itc hed network.

Real-time Transport Control Protocol, define d in RFC 3550

, a sister protocol of the Real-time

Transport Protocol (RTP), It partners RTP in the delivery and packaging of multimedia data,

but does not transport any data itself. It is used periodically to transmit control packets to participants in a streaming multimedia session. The primary function of RTCP is to provide feedback on the quality of service being provided by RTP.

Real-time Transport Protocol defines a standardized packet format for delivering audio and video over the Internet. It was developed by the Audio-Video Transport Working Group of the

IETF

and first published in 1996 as RFC 1889

SIP

STUN

TCP

Session Description Protocol, is a form at for describing streaming media initialization parameters. It has been published by the IETF as RFC 2327.

Session Initiation Protocol, An IP telephony signaling protocol developed by the IETF (RFC3261). SIP is a text-based protocol suitabl e fo r inte grated vo ic e-data applications. SIP is designed for voice transmission an d uses fewer resources and is considerably less complex than H.323. All Grandstream products are SIP based

Simple Traversal of UDP over NATs, is a network protoc ol

allowing clients behind NAT (or multiple NATs) to find out its public address, the type of NAT it is behind and the internet side port associated by the NAT with a particular local port. This information is used to set up UDP communication between t wo hosts that are both behind NAT routers. The protocol is defined in

RFC 3489

. STUN will usually work good with non-symmetric NAT routers.

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. Using

to one

TFTP

UDP

VAD

VLAN

Transmission Control Protocol, is one of the core protocols of the Internet protocol suite TCP, applications on networked hosts can create connections to one another, over which they can exchange data or packets

. The protocol guaran tees reli able and in-order delivery of sender

to receiver data.

Trivial File Transfer Protocol, is a very simple file very basic form of FTP

; It uses UDP (port 69 ) as its transpo rt protocol.

transfer protocol, with the functionality of a

User Datagram Protoc ol (UDP ) is one of the core protocols of the Internet protocol suite UDP, programs on net worked computers can send s ho r t messages known as datagrams another. UDP does not provide the reliability and ordering guarantees that TCP

does; datagrams may arrive out of order or go missing without notice. However, as a result, UDP is faster and more efficient for many lightweight or time-sensitive purposes.

Voice Activity Detection or Voice Activity Detector is an algorithm used in speech processing wherein, the presence or absence of human speech is detected from the audio samples.

VoIP

A virtual LAN, known as a VLAN, is a logically-independent network. Several VLANs can coexist on a single physical switch. It is usually refer to the IEEE 802.1Q tagging protocol.

Voice over IP VoIP encompasses many prot oc ols. All the protocols do some form of signalling of call

capabilities and transport of voice data from one point to another. e.g: SIP, H.323, etc.