Vingtor stentofon Sds-1 User manual

Technical Manual 1490000010

Quick Installation & Conﬁguration Guide

SDS-1

SIP Desk Station with Display & Dual LAN

The SDS-1 is a product developed for Zenitel Norway AS and

is primarily used as part of a Zenitel Norway AS IP Intercom

solution. The SDS-1 is not pre-congured to support or carry

emergency calls to any type of hospital, law enforcement agency,

medical care unit (“Emergency Service(s)”) or any other kind of

Emergency Service. You must make additional arrangements to

access Emergency Services. It is your responsibility to purchase

SIP-compliant Internet telephone service, properly congure the

SDS-1 to use that service, and periodically test your conguration

to conrm that it works as you expect. If you do not do so, it is

your responsibility to purchase traditional wireless or landline

telephone services to access Emergency Services.

ZENITEL NORWAY DOES NOT PROVIDE CONNECTIONS TO

EMERGENCY SERVICES VIA THE SDS-1. NEITHER ZENITEL

NORWAY NOR ITS OFFICERS, EMPLOYEES OR AFFILIATES

MAY BE HELD LIABLE FOR ANY CLAIM, DAMAGE, OR

LOSS. YOU HEREBY WAIVE ANY AND ALL SUCH CLAIMS

OR CAUSES OF ACTION ARISING FROM OR RELATING

TO YOUR INABILITY TO USE THE SDS-1 TO CONTACT

EMERGENCY SERVICES, AND YOUR FAILURE TO MAKE

ADDITIONAL ARRANGEMENTS TO ACCESS EMERGENCY

SERVICES IN ACCORDANCE WITH THE IMMEDIATELY

PRECEDING PARAGRAPH.

Package Contents

Precautions

WARNING: Please DO NOT power cycle the SDS-1 during system boot-up

or rmware upgrade. You may corrupt rmware images and cause the unit to

malfunction.

Overview

• SDS-1 SIP Desk Station with Display and Dual LAN (Item Number: 1490000010)

SDS-1 is a Small Business HD IP phone that features 2 lines with 2 SIP accounts,

132x48 backlit graphical LCD, 3 XML programmable context-sensitive softkeys,

dual network ports with PoE and 3-way conference. The SDS-1 delivers HD

wideband audio, superb full-duplex hands-free speakerphone with advanced

acoustic echo cancellation, advanced security protection for privacy, leading edge

SIP intercom features and integration with the Pulse and AlphaCom IP Intercom

platforms developed by Zenitel Norway AS.

1 x Phone Main

Case

1 x Handset

1 x Phone Cord

1 x Ethernet

Cable

1 x Phone Stand

1 x Quick Installation Guide

1 x GPL License

Phone Installation

Installing the Phone with Phone Stand

1. Insert the hooks on top of the stand into the slots on the back of the phone

• You can either use the upper OR lower slots

2. Firmly slide the stand upward to lock it in place

Installing the Phone with Wall Mount

1. Insert all 4 hooks located at the front of the wallmount into the slots on the back

of the phone.

2. Firmly slide the wall mount upward to lock it in place.

3. Attach the phone to the wall via the wall mount holes.

To setup the GXP1620/1625, follow the steps below:

1. Connect the handset and main phone case with the

phone cord.

2. Connect the LAN port of the phone to the RJ-45 socket

of a hub/switch or a router (LAN side of the router) using the

Ethernet cable.

3. Connect the 5V DC output plug to the power jack on the

phone; plug the power adapter into an electrical outlet.

4. The LCD will display provisioning or rmware upgrade in-

formation. Before continuing, please wait for the date/time

display to show up.

5. Using the phone embedded web server or keypad con-

guration menu, you can further congure the phone using

either a static IP or DHCP.

CONNECTING THE PHONE:

Tips For Using the Keypad:

Line Key

Message Waiting

Indicator

Message

Standard Keypad

Phonebook

Page/Intercom

Home Button

Hold

Softkey

Menu/OK key

Navigation keys

Record

Headset

Speaker

Volume

Send

Transfer

Mute

Conference

1. To access the MENU, press the round MENU button.

2. Navigate the menu by using the UP/DOWN and LEFT/

RIGHT buttons.

3. Press the round MENU button to conrm a menu selec-

tion.

4. The phone automatically exits MENU mode when there

is an incoming call, the phone goes off-hook, or when the

MENU mode is left idle for 60 seconds.

Installing the Phone (Phone Stand):

1. Insert the hooks on top of the stand into a slots, you have options

to use either upper OR lower slots.

2. Firmly slide the stand upward to lock it in place.

Installing the Phone (Wall Mount):

UPPER

LOWER

1. Insert all 4 hooks located in the front of wall mount into the slots.

2. Firmly slide the wall mount upward to lock it in place.

3. Pull out and rotate the tab from handset rest to hold the handset

while the phone is mounted on the wall.

LAN Port Power Handset Port

PC Port Headset Port

Tab with extension up

Handset Rest

Tab with extension down

Connecting the Phone

1. Connect the handset and main phone case with the phone cord.

2. Connect the LAN port of the phone to the RJ-45 socket of a PoE switch using

the Ethernet cable.

• The LCD will display provisioning or rmware upgrading information.

Before continuing, please wait for the date/time display to appear.

3. Using the web conguration interface or the keypad conguration menu, you

can further congure the phone using either a static IP or DHCP.

To setup the GXP1620/1625, follow the steps below:

1. Connect the handset and main phone case with the

phone cord.

2. Connect the LAN port of the phone to the RJ-45 socket

of a hub/switch or a router (LAN side of the router) using the

Ethernet cable.

3. Connect the 5V DC output plug to the power jack on the

phone; plug the power adapter into an electrical outlet.

4. The LCD will display provisioning or rmware upgrade in-

formation. Before continuing, please wait for the date/time

display to show up.

5. Using the phone embedded web server or keypad con-

guration menu, you can further congure the phone using

either a static IP or DHCP.

CONNECTING THE PHONE:

Tips For Using the Keypad:

Line Key

Message Waiting

Indicator

Message

Standard Keypad

Phonebook

Page/Intercom

Home Button

Hold

Softkey

Menu/OK key

Navigation keys

Record

Headset

Speaker

Volume

Send

Transfer

Mute

Conference

1. To access the MENU, press the round MENU button.

2. Navigate the menu by using the UP/DOWN and LEFT/

RIGHT buttons.

3. Press the round MENU button to conrm a menu selec-

tion.

4. The phone automatically exits MENU mode when there

is an incoming call, the phone goes off-hook, or when the

MENU mode is left idle for 60 seconds.

Installing the Phone (Phone Stand):

1. Insert the hooks on top of the stand into a slots, you have options

to use either upper OR lower slots.

2. Firmly slide the stand upward to lock it in place.

Installing the Phone (Wall Mount):

UPPER

LOWER

1. Insert all 4 hooks located in the front of wall mount into the slots.

2. Firmly slide the wall mount upward to lock it in place.

3. Pull out and rotate the tab from handset rest to hold the handset

while the phone is mounted on the wall.

LAN Port Power Handset Port

PC Port Headset Port

4. Pull out the tab from the handset cradle (see gure below).

5. Rotate the tab and plug it back into the slot with the extension up to hold the

handset while the phone is mounted on the wall.

Tips for Using the Keypad

Phone Conguration

For further information on the conﬁguration of AlphaCom, SIP, and Pulse, please go

to wiki.zenitel.com.

Conguring the SDS-1 Using the Keypad

1. Make sure the phone is idle.

2. Press the Menu key to access the keypad menu to congure the phone.

3. Select Phone > SIP > Account to congure settings for SIP Server (AlphaCom

or Pulse), SIP User ID (Extension Number), SIP Auth ID and SIP Password.

4. Follow the menu options to congure the basic features of the phone, e.g. the

IP address if using a static IP.

Line Keys

Status Indicator

Message

Standard Keypad

Phonebook

Page/Intercom

Home

Hold

Softkeys

Menu/OK Key

Navigation Keys

Record

Headset

Speaker

Volume

Send

Transfer

Mute

Conference

1. To access the menu, press the round Menu key.

2. Navigate the menu by using the Up/Down and Left/Right arrow-keys.

3. Press the round Menu key to conrm a menu selection.

4. The phone automatically exits menu mode when there is an incoming call, the

phone goes o󰀨-hook, or when menu mode is left idle for 60 seconds.

Conguring the SDS-1 Using Web Browser

1. Ensure your phone is properly powered up and connected to the Internet.

2. Press the Menu key to enter the menu of the phone.

3. Navigate to Status > Network Status and press the Menu key to check the IP

address.

4. Enter the phone’s IP address in your PC’s browser.

5. Log in by entering the default Username: admin and Password: alphaadmin

6. Register the account on the SDS-1 by selecting ACCOUNTS > Account 1/2

> General Settings to congure Account Name, SIP Server (AlphaCom or

Pulse), SIP User ID (Extension Number).

AlphaCom Conguration

AlphaCom SDS-1 Account Setup

●Select ACCOUNTS >Account 1 >General Settings

● Enter the values shown above for the parameters

Account Active: Check Yes button

SIP Server: IP address of AlphaCom server

SIP User ID: Directory Number of SDS-1 phone

Authenticate ID: Same as SIP User ID

Note: For changes to take eect, it may be necessary to temporarily disable the account.

First check the No button for Account Active, then click Save and Apply. Once this is

done, re-enable the account by checking the Yes button for Account Active followed by

Save and Apply again.

AlphaCom SDS-1 Audio Settings

● Check in AlphaPro under Users & Stations the codec that has been

selected for the SIP phone (normally G722)

●Select Account 1 >Audio Settings

●Set all codecs from the Preferred Vocoder list to the one dened in

AlphaPro, i.e. G722.

Pulse Conguration

Pulse SDS-1 Account Setup

●Select ACCOUNTS >Account 1 >General Settings

● Enter the values shown above for the parameters

Account Active: Check Yes button

SIP Server: IP address of intercom station set as Pulse Server

SIP User ID: Directory Number of the SDS-1 phone

Authenticate ID: Same as SIP User ID

Pulse Group Call

The Pulse Server transmits group calls using IP multicast paging. Each group

call uses its own unique multicast IP address. To nd the multicast IP address:

● Log into the Pulse Server

●Select Server Management >Group Call

Make a note of the multicast addresses under Group audio address.

● Log into the SDS-1

●Select SETTINGS >Multicast Paging

●Set Multicast Paging Codec to G722

●Enter the multicast addresses under Listening Address

● Click Save and Apply and Reset

When a Group Call is activated, the SDS-1 will automatically broadcast the audio

in the loudspeaker. The SDS-1 will display the text of the Group Call as entered

under Nickname. If the SDS-1 is busy in a regular call when a Group Call is

made, it will by default not play the Group Call audio. If Paging Barge is set to

value 2 or higher, the current call will be placed On Hold, and the Group Call

audio will be broadcast. When the Group Call is ended, press the Hold button to

resume the regular call.

Zenitel Norway AS

Sandakerveien 24C

0403 Oslo, Norway

For Warranty and RMA information, please visit www.zenitel.com

DOC NUMBER

Zenitel and its subsidiaries assume no responsibility for any errors that may appear in this publication, or for damages arising from the information therein. Vingtor-Stentofon products are developed and marketed

by Zenitel. The company’s Quality Assurance System is certiﬁed to meet the requirements in NS-EN ISO 9001. Zenitel reserves the right to modify designs and alter speciﬁcations without notice.

ZENITEL PROPRIETARY. This document and its supplementing elements, contain Zenitel or third party information which is proprietary and conﬁdential. Any disclosure, copying, distribution or use is prohibited, if