Grandstream Networks Handy Tone 386 User manual

User Manual
HandyTone-386
Analog Telephone Adaptor
For SW Release Version 1.0.2.5
Grandstream Networks, Inc.
www.grandstream.com

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Table of Contents
1WELCOME..................................................................................................................................3
2INSTALLATION.........................................................................................................................4
3WHAT IS INCLUDED IN THE PACKAGE............................................................................6
3.1 SAFETY COMPLIANCES ...........................................................................................................6
3.2 WARRANTY.............................................................................................................................6
4PRODUCT OVERVIEW............................................................................................................7
4.1 KEY FEATURES .......................................................................................................................7
4.2 HARDWARE SPECIFICATION ...................................................................................................8
5BASIC OPERATIONS................................................................................................................9
5.1 GET FAMILIAR WITH VOICE PROMPT .....................................................................................9
5.2 MAKE PHONE CALLS ............................................................................................................10
5.2.1 Calling phone or extension numbers.............................................................................10
5.2.2 Direct IP calls ...............................................................................................................10
5.2.3 Blind Transfer................................................................................................................11
5.2.4 Attended Transfer..........................................................................................................11
5.2.5 3-way Conferencing ......................................................................................................12
5.2.6 Send and Receive PSTN Calls Through PSTN Line Port..............................................12
5.3 CALL FEATURES ...................................................................................................................12
5.3.1 Call Features Table.......................................................................................................12
5.3.2 PSTN Pass Through / Life Line.....................................................................................13
5.4 FAX SUPPORT.......................................................................................................................13
5.5 LED LIGHT PATTERN INDICATION .......................................................................................13
6CONFIGURATION GUIDE.....................................................................................................14
6.1 CONFIGURING HANDYTONE-386 WAN IP THROUGH VOICE PROMPT.................................14
6.1.1 DHCP Mode..................................................................................................................14
6.1.2 STATIC IP Mode...........................................................................................................14
6.1.3 TFTP Server Address ....................................................................................................14
6.2 CONFIGURING HANDYTONE-386 WITH WEB BROWSER.......................................................14
6.2.1 Access the Web Configuration Menu ............................................................................14
6.2.2 End User Configuration................................................................................................14
6.2.3 Advanced User Configuration.......................................................................................17
6.2.4 Saving the Configuration Changes................................................................................25
6.2.5 Rebooting the HandyTone-386 from Remote................................................................26
6.3 CONFIGURATION THROUGH A CENTRAL SERVER.................................................................26
7SOFTWARE UPGRADE..........................................................................................................28
7.1 UPGRADE THROUGH HTTP (PENDING).................................................................................28
7.2 UPGRADE THROUGH TFTP ...................................................................................................28
8RESTORE FACTORY DEFAULT SETTING.......................................................................30

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1Welcome
Congratulations on becoming an owner of HandyTone-386. You made an excellent choice and we
hope you enjoy all of its capabilities.
Grandstream's HandyTone-386 is an all-in-one VoIP integrated access device that features superb
audio quality, rich functionalities, high level of integration, compactness and ultra-affordability. The
HandyTone-386 is fully compatible with SIP industry standard and can interoperate with many other
SIP compliant devices and software on the market.
Grandstream HandyTone-386 is a new addition to the popular HandyTone product family. The new
HandyTone-386 features two FXS ports each with independent SIP accounts.

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2Installation
HandyTone-386 Analog Telephone Adaptor is an all-in-one VoIP integrated device designed to be a
total solution for networks providing VoIP services.
The HandyTone-386 VoIP functionalities are available via regular analog telephones.
The following photo illustrates the appearance of a HandyTone-386.
Top View Side Views
Interconnection Diagram of the HandyTone-386:
BUTTON
RED LED
GREEN LED
RJ45
10M Ethernet
WAN
RJ11
FXS Port
(Phone)
+5V/1200mA
RJ11
Line Port
(Line )
Fax
Cordless Phone
Analog Phone
Internet ADSL/Cable
Modem Ethernet
Phone Line
LINE
WAN
FXS 1 FXS 2
Analog Phone
Cordless Phone
Fax
RJ11
FXS Port
(Fax)

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HandyTone-386 has two FXS ports. The RJ11 jack next to the WAN is called FXS port 2 and the
RJ11 jack on the corner is called FXS port 1. The RJ11 jack on the side on of the HandyTone-386 is a
LINE port. Each FXS port can have a separate SIP account. This is a key feature of HandyTone-386.
Both ports can make calls concurrently.
Following are the steps to install a HandyTone-386:
1. Connect a standard touch-tone analog telephone (or fax machine) to FXS port 1.
2. Connect another standard touch-tone analog telephone (or fax machine) to FXS port 2.
3. Insert a standard telephone cable into the LINE port of HandyTone-386 and connect the other
end of the telephone cable to a wall jack.
4. Insert the Ethernet cable into the WAN port of HandyTone-386 and connect the other end of
the Ethernet cable to an uplink port (a router or a modem, etc.)
5. Insert the power adapter into the HandyTone-386 and connect it to a wall outlet.
Please follow the instructions in section 6.2.1 to configure the HandyTone-386.

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3What is Included in the Package
The HandyTone-386 package contains:
1) One HandyTone-386
2) One universal power adaptor
3) One Ethernet cable
3.1 Safety Compliances
The HandyTone-386 is compliant with various safety standards including FCC/CE and C-tick. Its
power adaptor is compliant with UL standard. The HandyTone-386 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 warranty policy without prior notification.
Warning: Please do not attempt to use a different power adaptor. Using other power adaptor may
damage the HandyTone-386 and will void the manufacturer warranty.
Caution: Changes or modifications to this product not expressly approved by Grandstream, or
operation of this product in any way other than as detailed by this User Manual, could void your
manufacturer warranty.
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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4Product 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.
•Supports dual SIP accounts via dual FXS ports
•Powerful digital signal processing (DSP) to ensure superb audio quality; advanced adaptive
jitter control and packet loss concealment technology
•Support various vocoders including G.711 (a-law and u-law), G.723.1 (5.3K/6.3K), G.726
(40K/32K/24K/16K), as well as G.728, G.729A/B, and iLBC.
•Support Caller ID/Name display or block, Hold, Call Waiting/Flash, Call Transfer, Call
Forward, Call Waiting Caller ID, in-band and out-of-band DTMF, Dial Plans, etc.
•Support fax pass through and T.38
•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 Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS)
•Support automated NAT traversal without manual manipulation of firewall/NAT
•Support device configuration via built-in IVR, Web browser or central configuration file
through TFTP or HTTP
•Support firmware upgrade via TFTP or HTTP (pending) with encrypted configuration files.
•Ultra compact (wallet size) and lightweight design, great companion for travelers
•Compact, lightweight Universal Power adapter.

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4.2 Hardware Specification
The table below lists the hardware specification of HandyTone-386.
Model HandyTone-386
WAN interface 1xRJ45 10Base-T
FXS telephone port 2xFXS
PSTN Port 1x PSTN pass-through or life line port
Button 1
LED Green and red color
Universal Switching
Power Adaptor Input: 100-240VAC 50-60 Hz Output: +5VDC, 1200mA,
UL certified
Dimension 70mm (W)
130mm (D)
27mm (H)
Weight 0.6lbs (0.3kg)
Temperature 40 - 130oF
5 – 45oC
Humidity 10% - 90%
(non-condensing)
Compliance

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5Basic Operations
5.1 Get Familiar with Voice Prompt
HandyTone-386 has stored a voice prompt menu for quick browsing and simple configuration.
Currently, the voice prompt menu and the LED button are designed for FXS port 1 only.
To enter this voice prompt menu, simply press the button or “***” from the analog phone.
Menu Voice Prompt 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, 99 menu option
01 “DHCP Mode”,
“Static IP Mode” Enter “9” to toggle the selection
02 “IP Address “ + IP address The current WAN IP address is announced
Enter 12 digit new IP address if in Static IP
Mode
03 “Subnet “ + IP address Same as menu 02
04 “Gateway “ + IP address Same as menu 02
05 “DNS Server “ + IP address Same as menu 02
06 “TFTP Server “ + IP address Same as menu 02
47 “Direct IP Calling” When entered, you will be prompted a dial
tone, then enter 12 digit IP address
This menu can also be entered by pressing the
button again
86 “Voice Messages Pending”
“No Voice Messages” Enter “9” to dial pre-configured phone number to retrieve VM
99 “RESET” Enter “9” to reboot the phone
Enter encoded MAC address to restore factory
default setting
“Invalid Entry” Automatically returns to main menu
Notes:
•Once the button is pressed, it enters the voice prompt main menu. If the button is pressed again,
while it is already in the voice prompt menu, it jumps to “Direct IP Call” option and a dial tone
is prompted
•“*” 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 of the digits are collected, the input will be processed.
•Key entry can not be deleted but the phone may prompt error once it is detected

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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, and press # (assuming that “Use # as dial key” is selected in web
configuration).
Other functions available during the call are call-waiting/flash, call-transfer, and call-forward.
5.2.2 Direct IP calls
Direct IP calling allows two phones, that is, a HandyTone with an analog phone and another VoIP
Device, to talk to each other in an ad hoc fashion without a SIP proxy. VoIP calls can be made between
two phones if:
•Both HandyTone ATA and the other VoIP device (i.e., another HandyTone ATA or other SIP
products) have public IP addresses, or
•Both HandyTone ATA and the other VoIP device (i.e., another HandyTone ATA or other SIP
produces) are on the same LAN using private or public IP addresses, or
•Both HandyTone ATA and the other VoIP device (i.e., another HandyTone ATA or other SIP
products) can be connected through a router using public or private IP addresses.
To make a direct IP call, first pick up the analog phone or turn on the speakerphone on the analog
phone, follow Section 5.1 with voice prompt 47, followed by the 12-digit target IP address. Destination
ports can be specified by using “*4” (encoding for “:”) followed by the encoded port number.
Following is a table of the encoding scheme for the most commonly used characters:
Input Meaning
00 0
01 1
02 2
03 3
04 4
05 5
06 6
07 7
08 8
09 9
*0 . (dot character)
*4 : (column character)

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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 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: Call Feature has to be set to YES.
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 supported platforms only). This means the
transferee has received a 4xx response for the INVITE and we will try to recover the call. The
busy tone is just to indicate to the transferor that the transfer has failed.
•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 completed
successfully.
5.2.4 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 dial
tone.
2. A dials C’s number then # (or wait for 4 seconds). A and C are now in conversation.
3. A can hang up.

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Note: When intended Transfer failed, if A hangs up, the HandyTone-386 will ring user A again to
remind A that B is still on the call, by pressing FLASH or Hook again will restore the conversation
between A and B.
5.2.5 3-way Conferencing
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 *23 then C’s number then # (or wait for 4 seconds). A and C are now in conversation.
3. A presses FLASH again to begin conference.
5.2.6 Send and Receive PSTN Calls Through PSTN Line Port
User can send and receive calls from PSTN. To receive PSTN calls, simply make phone off hook when
the analog phone rings. To make a PSTN call, simply press *00 (default, can be configured in web
configuration page) to switch to the PSTN line and get dial tone and call out using the PSTN line.
5.3 Call Features
5.3.1 Call Features Table
Following table shows the call features of HandyTone-386.
Key Call Features
*30 Block Caller ID (for all subsequent calls)
*31 Send Caller ID (for all subsequent calls)
*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 calls)
*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 hang 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

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To use this feature, dial “*90” and get the dial tone. Then dial
the forward number and “#” for a dial tone, then hang 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 hang 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.
5.3.2 PSTN Pass Through / Life Line
When HandyTone-386 is out of power, the RJ 11 line jack on the HandyTone-486 side will function as
a pass through jack. The user will be able to use the same analog phone for PSTN calls.
5.4 FAX Support
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 tables show the LED light pattern indication.
RED LED indicates abnormal status
DHCP Failed or WAN No Cable Button flashes every 2 seconds (if DHCP is configured)
HandyTone-386 fails to register Button flashes every 2 seconds (if SIP server is configured)
GREEN LED indicates normal working status
Message Waiting Indication Button flashes every 2 seconds
RINGING Button flashes at 1/10 second
RINGING INTERVAL Button flashes every second

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6Configuration Guide
6.1 Configuring HandyTone-386 WAN IP through Voice Prompt
6.1.1 DHCP Mode
Follow section 5.1 with voice menu option 01 to enable HandyTone-386 to use DHCP.
6.1.2 STATIC IP Mode
Follow section 5.1 with voice menu option 01 to enable HandyTone-386 to use STATIC IP mode, then
use option 02, 03, 04 to set up HandyTone-386’s IP, Subnet Mask, Gateway 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-386 with Web Browser
HandyTone-386 series ATA has an embedded Web server that will respond to HTTP GET/POST
requests. It also has embedded HTML pages that allow users to configure the HandyTone-386 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-386 through section 5.1 with menu option 02. Then access
the HandyTone-386’s Web Configuration Menu using the following URI:
http://Phone-IP-Address
where the Phone-IP-Address is the IP address of the phone.
6.2.2 End User Configuration
Once this HTTP request is entered and sent from a Web browser, the HandyTone-386 will respond
with the following login screen:

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Grandstream Device Configuration
Password
Login
All Rights Reserved Grandstream Networks, Inc. 2005
The password is case sensitive and the factory default password for End User is “123”.
After a correct password is entered in the login screen, the embedded Web server inside the
HandyTone-386 will respond with the Configuration page which is explained in details below.
Grandstream Device Configuration
STATUS BASIC SETTINGS ADVANCED SETTINGS
End User Password: (purposely not displayed for security protection)
IP Address: dynamically assigned via DHCP (default) or PPPoE
(will attempt PPPoE if DHCP fails and following is non-blank)
PPPoE account ID:
PPPoE password:
Preferred DNS server: 0.0.0.0
statically configured as:
IP Address: 192 .168 .1.12
Subnet Mask: 255 .255 .255 .0
Default Router: 192 .168 .1.1
DNS Server 1: 19 .105 .5.1
DNS Server 2: 24 .127 .198 .4
Time Zone: current setting is " GMT-6:00 (US Central Time, Chicago)"
Da
y
li
g
ht Savin
g
s
Time: No Yes (if set to Yes, display time will be 1 hour ahead of normal time)
Cancel Update Reboot
All Rights Reserved Grandstream Networks, Inc. 2004

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End User
Password This contains the password to access the Web Configuration Menu. This
field is case sensitive.
IP Address There are 2 modes under which the HandyTone ATA can operate:
- If DHCP mode is enabled, then all the field values for the Static IP mode
are not used (even though they are still saved in the Flash memory.) The
HandyTone ATA will acquire its IP address from the first DHCP server it
discovers from the LAN it is connected.
To use the PPPoE feature the PPPoE account settings need to be set. The
HandyTone will attempt to establish a PPPoE session if any of the PPPoE
fields is set.
- If Static IP mode is enabled, then the IP address, Subnet Mask, Default
Router IP address, DNS Server 1 (primary), DNS Server 2 (secondary) fields
will need to be configured. These fields are reset to zero by default.
Time Zone This parameter controls how the displayed date/time will be adjusted
according to the specified time zone.
Daylight Savings Time This parameter controls whether the displayed time will be daylight savings
time or not. If set to Yes, then the displayed time will be 1 hour ahead of
normal time.
In addition to the Basic Settings configuration page, end users also have access to the device Status
page. The following is a snap shot of the device Status page. Details will be explained next.
Grandstream Device Configuration
STATUS BASIC SETTINGS ADVANCED SETTINGS
MAC Address: 00.0B.82.00.00.00
WAN IP Address:
Product Model: HT386
Software Version: Program-- 1.0.2.5 Bootloader-- 1.0.1.0 HTML-- 1.0.0.5 VOC-- 1.0.0.10
S
y
stem Up Time: 0 day(s) 4 hour(s) 43 minute(s)
Re
g
istered: Yes
PPPoE Link Up: disabled
detected NAT type is symmetric NAT
All Rights Reserved Grandstream Networks, Inc. 2004

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MAC Address The device ID, in HEX format. This is a very important ID for ISP
troubleshooting.
WAN IP Address This field shows WAN port IP address.
Product Model This field contains the product model info.
Software Version Program: This is the main software release, its number is always used for
firmware upgrade.
Bootloader: This is normally not changed.
HTML: This is the web user interface, normally not changed.
VOC: This is the codec program, normally not changed.
System Up Time This field indicates how long the device has been up since the last reboot.
Registered This field indicates whether the device is registered to the SIP server.
PPPoE Link Up This field shows whether the PPPoE connection is enabled or not.
Detected NAT Type This field shows what kind NAT the HandyTone is connected to via its
WAN port. It is based on STUN protocol.
6.2.3 Advanced User Configuration
To login to the Advanced User Configuration page, please follow the instructions in section 6.2.1 to
get to the following login page. The password is case sensitive and the factory default password for
Advanced User is “admin”.
Grandstream Device Configuration
Password
Login
All Rights Reserved Grandstream Networks, Inc. 2004
Advanced User configuration includes not only the end user configuration, but also advanced
configuration such as SIP configuration, Codec selection, NAT Traversal Setting and other
miscellaneous configuration. Following is a snap shot of the advanced configuration page.
Grandstream Device Configuration

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STATUS BASIC SETTINGS ADVANCED SETTINGS
Admin Password: (purposely not displayed for security protection)
FXS Port 1:
SIP Server: sip.mycompany.com (e.g., sip.mycompany.com, or IP address)
Outbound Prox
y
: (e.g., proxy.myprovider.com, or IP address, if any)
SIP User ID: 3123320 (the user part of an SIP address)
Authenticate ID: 3123320 (can be identical to or different from SIP User ID)
Authenticate
Password: (purposely not displayed for security protection)
Name: (optional, e.g., John Doe)
Register Expiration: 60 (in minutes. default 1 hour, max 45 days)
local SIP port: 5060 (default 5060)
local RTP port: 5004 (1024-65535, default 5004)
Enable Call Features: No Yes (if Yes, Call Forwarding & Call-Waiting-Disable are supported
locally)
Send DTMF: in-audio via RTP (RFC2833) via SIP INFO
DTMF Pa
y
load T
y
pe: 101
FXS Port 2:
SIP Server: sip.mycompany.com (e.g., sip.mycompany.com, or IP address)
Outbound Prox
y
: (e.g., proxy.myprovider.com, or IP address, if any)
SIP User ID: 3125250 (the user part of an SIP address)
Authenticate ID: 3125250 (can be identical to or different from SIP User ID)
Authenticate
Password: (purposely not displayed for security protection)
Name: (optional, e.g., John Doe)
Register Expiration: 60 (in minutes. default 1 hour, max 45 days)
local SIP port: 5062 (default 5062)
local RTP port: 5008 (1024-65535, default 5008)
Enable Call Features: No Yes (if Yes, Call Forwarding & Call-Waiting-Disable are supported
locally)

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Send DTMF: in-audio via RTP (RFC2833) via SIP INFO
DTMF Pa
y
load T
y
pe: 101
General Misc. Settin
g
s:
Preferred Vocoder:
(in listed order) choice 1: current setting is " PCMU"
choice 2: current setting is " PCMA"
choice 3: current setting is " G723"
choice 4: current setting is " G729"
choice 5: current setting is " G726-32"
choice 6: current setting is " iLBC"
Voice Frames per TX: 2(up to 10/20/32/64 for G711/G726/G723/other codecs respectively)
G723 rate: 6.3kbps encoding rate 5.3kbps encoding rate
iLBC frame size: 20ms 30ms
iLBC payload type: 97 (between 96 and 127, default is 97)
Silence Suppression: No Yes
SIP Registration: Yes No
Unregister On Reboot: Yes No
Fax Mode: T.38 (Auto Detect) Pass-Through
Layer 3 QoS: 48 (Diff-Serv or Precedence value)
Layer 2 QoS: 802.1Q/VLAN Tag 0802.1p priority value 0(0-7)
Use DNS SRV: No Yes
User ID is phone
number: No Yes
Early Dial: No Yes (use "Yes" only if proxy supports 484 response)
Dial Plan Prefix: (this prefix string is added to each dialed number)
No Key Entry Timeout: 4(in seconds, default is 4 seconds)
Use # as Dial Key: No Yes (if set to Yes, "#" will function as the "(Re-)Dial" key)
Use random port: No Yes
NAT Traversal: No

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Yes, STUN server is: stun.mycompany.com (URI or IP:port)
keep-alive interval: 20 (in seconds, default 20 seconds)
Use NAT IP (if specified, this IP address is used in SIP/SDP message)
Proxy-
R
equire: (if specified, the content will appear in Proxy-Require
header)
TFTP Upgrade Server: 192 .168 .1.30 (for remote software upgrade and configuration)
SUBSCRIBE for MWI: No, do not send SUBSCRIBE for Message Waiting Indication
Yes, send periodical SUBSCRIBE for Message Waiting Indication
Offhook Auto-Dial: (User ID/extension to dial automatically when offhook)
Send Flash Event: No Yes (Flash will be sent as a DTMF event if set to Yes)
FXS Impedance: current setting is " 600 Ohm(North America)"
Caller ID Scheme: current setting is " Bellcore"
Onhook Voltage (FXS
Port): current setting is " 48V"
Disable Call-Waiting: No Yes
Polarity Reversal: No Yes (reverse polarity upon call establishment and termination)
NTP Server: time.nist.gov (URI or IP address)
Send Anonymous: No Yes (caller ID will be blocked if set to Yes)
Lock keypad update: No Yes (configuration update via keypad is disabled if set to Yes)
PSTN access code: (key pattern to use the PSTN line, default is "*00")
Syslog Server:
Syslog Level: current setting is "NONE"
Cancel Update Reboot
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Admin Password This contains the password to access the Advanced Web Configuration page.
This field is case sensitive.
SIP Server SIP Server’s URI or IP address
Outbound Proxy SIP Outbound Proxy Server’s URI or IP address