Eternity Ne
Eternity Ne
System Manual
System Manual
This is a general documentation for all models of the product. The product may not support all the features and
facilities described in the documentation.
Information in this documentation may change from time to time. Matrix Comsec reserves the right to revise
information in this publication for any reason without prior notice. Matrix Comsec makes no warranties with respect
to this documentation and disclaims any implied warranties. While every precaution has been taken in the
preparation of this system manual, Matrix Comsec assumes no responsibility for errors or omissions. Neither is any
liability assumed for damages resulting from the use of the information contained herein.
Neither Matrix Comsec nor its affiliates shall be liable to the purchaser of this product or third parties for damages,
losses, costs or expenses incurred by the purchaser or third parties as a result of: accident, misuse or abuse of this
product or unauthorized modifications, repairs or alterations to this product or failure to strictly comply with Matrix
Comsec's operating and maintenance instructions.
Copyright
All rights reserved. No part of this system manual may be copied or reproduced in any form or by any means
without the prior written consent of Matrix Comsec.
Version 1
Release date: February 24, 2011
Introduction..................................................................................................................................................... 1
Welcome ............................................................................................................................................................. 1
About this System Manual .................................................................................................................................. 1
Table of Contents i
ii Table of Contents
iv Table of Contents
Table of Contents v
Welcome
Thank you for choosing the Matrix ETERNITY NE! We hope you will make optimum use of this intelligent,
integrated IP-PBX. Please read this document carefully to get acquainted with the product before installing and
operating it.
You may refer to the ETERNITY NE Quick Start and the ETERNITY NE User Card shipped with the system.
You may also refer the User Card for the proprietary digital key phone/Extended IP Phone shipped with the system,
for instructions on using the features of the ETERNITY NE.
Intended Audience
This System Manual is aimed at:
• System Engineers, who will install, maintain and support the ETERNITY NE. System Engineers are
persons who customize the system configuration to meet the requirements of the organization/users. It is
assumed that they are experienced in installing PBX, are familiar with telecom wiring technology, how it
works, and the various technical terms and functions associated with it.
No one, other than the System Engineer is permitted to make any alterations to the configuration of the
ETERNITY NE.
• System Administrators, who are persons who will administer the ETERNITY NE. Generally an operator/
receptionist in an organization, or the staff manning the reception or front desk area of the establishment
are selected as System Administrators.
It is assumed that the System Administrators have some previous experience in administering a PBX and
its Terminals and Consoles. The System Administrators are not expected to install and configure the PBX,
but only the routine jobs and features that are specific to them like generating SMDR reports, Setting
report filters, configuring the features, Setting Alarms, reminders, etc.
• Chapter 1: Introduction: gives an overview of this document, its purpose, intended audience,
organization, terms and conventions used to present information and instructions.
• Chapter 2: Know Your ETERNITY NE: describes the system’s hardware, its interfaces, application
scenarios, and the available configurations.
• Chapter 3: Installing ETERNITY NE: gives step-by-step instructions for preparing for and installing the
ETERNITY NE, connecting the cable lines, and powering ON the system. It also recommends safety
measures for protecting the system and persons handling the installation and maintenance.
• Chapter 4: Configuring ETERNITY NE: contains description of the different tools available to configure
ETERNITY.
• Chapter 5: Configuring Basic Settings: describes the basic configuration required for the trunks and
extensions.
• Chapter 6: Voice Mail System: describes the Voice Mail System configuration, when the voice mail
module is installed in the ETERNITY.
• Chapter 7: Features and Facilities: describes in detail each feature and facility offered by the ETERNITY
NE.
The features are arranged alphabetically by Name to make it easy for you to locate the description you
want to look up.
• Chapter 8: Maintenance: provides instructions for back-up, generating system reports, debugging, and
trouble shooting.
• Chapter 9: Voice Mail Features: describes the features and facilities offered by the VMS installed in
ETERNITY NE.
• Chapter 10: System Status: displays status of the System, Network, Mobile ports and SIP Trunks.
This System Manual is presented in a manner that will help you find the information you need easily and quickly.
You may use the table of contents and the Index to navigate through this document to the relevant topic or
information you want to look up.
Access Codes
Access codes are strings of digits dialed by an extension to
• call another extension, Department Group.
• grab a trunk line
• use a Feature, e.g. Do Not Disturb, Call Forward
The Access Codes provided in the instructions throughout this document, are default access codes. It is possible to
change the Access Codes according to user requirement and preferences. Verify with the Installer, if the default
Access Codes have been changed, and use the codes configured by the Installer. For more information, read the
topic “Access Codes” in this document.
Notices
The following symbols have been used for notices to draw your attention to important items.
Important: to indicate something that requires your special attention or to remind you of
something you might need to do when you are using the system.
Caution: to indicate an action or condition that is likely to result in malfunction or damage to the
system or your property.
Warning: to indicate a hazard or an action that will cause damage to the system and or cause
bodily harm to the user.
Tip: to indicate a helpful hint giving you an alternative way to operate the system or carry out a
procedure, or use a feature more efficiently.
Acronyms have been defined in the text and a list of the same is appended.
Some of the terms specific to this Manual that you will encounter are defined below:
• 'ETERNITY', 'System', 'PBX': These words are used interchangeably and synonymously to mean all
models of ETERNITY NE.
• CO Lines: The lines subscribed from the CO Network. In this document it refers to Analog, two-wire Trunk
Lines.
• Digital Key Phone (DKP): refers to EON, the proprietary digital key phone of Matrix supplied with the
ETERNITY. The term 'Digital Key Phone' refers to all models of EON.
• External Calls: calls made by users of ETERNITY to subscribers of PSTN, PLMN, ITSPs, etc.
• External Numbers: numbers of parties/individuals outside the PBX or PBX network. The unique number
string given to subscribers of PSTN, PLMN, ITSP, etc.
• Internal Calls: calls made from and received by one extension to another extension of the ETERNITY.
• Mobile Extension: A mobile/landline phone used as a remote extension of ETERNITY. You can access all
the features of an extension of ETERNITY from the mobile/landline phone.
• Port: the physical interfaces on the cards for trunk lines and extension lines.
• Service Provider: the providers of telecom network lines/Internet - POTS, PSTN, GSM, and Internet
Telephony Service Providers (ITSP).
• Single Line Telephone (SLT): any standard two-wire telephone attached as extensions of the ETERNITY.
• System Administrator Commands/SA Commands: number strings dialed from the System
Administrator access/mode to operate features or set/cancel features for other extensions.
• System Commands/SE Commands: number strings dialed from the System Engineer access/mode to
configure the system features/functions.
• CO trunks: Two-wire trunks, i.e. analog trunk lines from the POTS network.
Using this Manual, you will be able to set up, operate and make optimum use of this feature-packed PBX.
If you encounter any technical problems, please contact your Dealer/reseller or the Matrix Support team.
Introduction
The Matrix ETERNITY NE is an all integrated IP-PBX with seamless mobility. It caters to the communication
requirements of small businesses.
ETERNITY NE offers connectivity to analog and digital networks: CO, Mobile and VoIP. So, you have access to
multiple telecom networks on a single platform. The system's intelligent Least Cost Routing logic diverts your calls
through the appropriate network, ensuring least possible call cost.
Besides Operator consoles (Digital Key phones, Direct Station Selection consoles) and standard telephones, you
can also connect a Fax machine, a door phone, and a door lock release relay to the ETERNITY NE.
Least Cost Routing and Call Budgeting help reduce communication cost and enhances productivity.
ETERNITY NE can route a VoIP call on to GSM. In the same way, a call can be routed either on VoIP, GSM or CO.
Further, you can select Fixed or Least Cost Routing to route outgoing calls. ETERNITY NE can handle calls on all
ports simultaneously, allowing full traffic on all ports.
ETERNITY NE can work as an adjunct to your existing telephony infrastructure, as a Gateway, saving you the cost
of equipment replacement, wiring and installation, while giving you multiple network connectivity and a host of
intelligent features.
The system is built on PCM/TDM, 100 percent non-blocking, digital technology, providing high density switching,
and is powered by a 32-bit RISC processor. Ensuring reliable, efficient, and unrestricted simultaneous
communication (incoming and outgoing) by all users.
Key Features
• Auto Attendant
• Call back on Mobile Phone
• Call Budget on Trunks and Extensions
• Call Forking
• CDR Reports
• Conference Dial-in
• Closed User Group (With/without Exchange ID)
• Conversation Recording
• Door Phone with Relay Connectivity
• Fax over IP (FoIP)
• IP Trunks and Extensions
• Mobile Twinning (Dual Ring)
• Multi-Party Conference
• Presence Indication and IM
• Remote Management
• Return Call to Original Caller (RCOC)
• Scheduled Call Forward
• SIP Registrar and Proxy Server
• Voice Mail to Email Notification
• VoIP calls over UMTS (3G)
Also refer “Appendix” for a complete list of Hardware and Software features and technical specifications.
ETERNITY NE is easy to install and operate. The built-in web server Jeeves allows you to configure the system
parameters and features on site and also from a remote location using any Internet browser.
The ETERNITY NE supports the following interfaces for connecting to different telecom networks, digital key
phones, standard telephones and external devices such as a Door Phone and a Door Lock release.
The CO Interface
The CO Interface enables the ETERNITY NE to be connected to the PSTN Network. The PSTN Networks across
the world support various standards and differ in features. For example, some networks support Caller ID
Presentation using DTMF signaling, while some support Caller ID Presentation using FSK signaling; some
networks offer 600 Ohms Impedance, while others offer complex impedance.
The versatile architecture of ETERNITY NE allows it to be connected to such networks differing in their
characteristics.
The VoIP Interface supports Session Initiation Protocol (SIP), the industry standard VoIP.
With SIP Trunks users can make IP calls using the SIP Server of the Internet Telephony Service Providers (ITSPs).
The VoIP Card has an in-built Registrar Server that allows any SIP enabled device like a Wi-Fi mobile handset, a
PDA or an IP-Phone to be registered with it and function as the 'SIP Extension' of the ETERNITY. The SIP
Extension users can make and receive calls to any extension user of the ETERNITY as well as any external
numbers over PSTN, GSM, VoIP. With SIP Extensions, organizations can communicate and stay connected at the
lowest cost without any geographical restrictions.
The VoIP Interface supports adaptive jitter buffer for reducing delay and improving speech quality.
The key auto attendant and voice mail features supported by ETERNITY's Voice Mail System are:
• Programmable Mailbox Size.
• Programmable Message Length.
• Welcome greetings according to the time of the day.
• Different voice greetings for different time zones.
• Special greetings for holidays.
• Five call transfer types: none, blind, wait for ring, wait for answer, and screened.
• Dial by extension.
• Dial by name.
• Personalized greetings for each mailbox.
• Individual mailbox size.
• Call forward to Voice Mail.
• Message forwarding.
• Distribution lists.
ETERNITY's Voice Mail System also forms the basis of other features like:
• Conversation Recording
• Call Taping
LAN Interface
ETERNITY NE has a single Ethernet port, to which you can connect a standalone computer or to a LAN Switch.
WAN Interface
ETERNITY NE supports WAN interface over Ethernet port and Wireless WAN over the UMTS Mobile Port with 3G
SIM.
When you use Wireless WAN over UMTS Mobile port, you can use the Ethernet Port as LAN.
Door Phone
You can connect any standard 4-wire door phone to the Door Phone port of the ETERNITY. The door phone can be
operated in conjunction with a Door Lock connected to the Digital Output Port.
Refer System Capacity and Resources in the Appendix, for an at-a-glance view of the ports available for each of
the aforementioned Interface options on all configurations of ETERNITY NE.
ETERNITY NE6
ETERNITY NE4
ETERNITY NE2
Illustrated below are design of the enclosure and the position of connectors on each configuration of ETERNITY
NE.
ETERNITY NE6
ETERNITY NE4
ETERNITY NE
The Matrix ETERNITY NE is designed for small and home offices, and nascent enterprises.
Illustrated below is an example of how ETERNITY NE can be optimally used by small businesses, taking full
advantage of its enterprise grade features and facilities.
• Necessary telecom wiring in place, with wall jacks for extension lines at the required locations.
• Standard, good quality, twisted pair telephone cables with 0.5 mm conductor diameter, with RJ11 plug.
• For the SLT ports, arrange for as many standard analog telephone instruments as required to connect as
SLT extensions. You may select any standard telephone instrument like rotary phone, Pulse/Tone
switchable push-button phone, Feature phone or Cordless phone.
• One or two digital key phones (Matrix proprietary phones), as required, to connect as DKP extensions.
• For the CO ports, arrange for one or more active Analog, two-wire trunk lines, as required.
• For the Mobile ports, A SIM card to test mobile network connectivity, if GSM/UMTS module is present.
• SIP Account information to be configured in the system to test SIP calls, if VoIP module is present.
• Any standard SIP Phone or Matrix Extended IP Phone to register as SIP Extension of ETERNITY NE, if
required. The first 8 SIP Extensions you connect are free.
To connect additional 8 SIP Extensions, you would need to purchase the IP8 License.
• If Voice Mail System module (VMS) is present, the Pen Drive and the License Voucher (provided by Matrix
with the VMS package).
• A standalone computer or a computer connected in a LAN to access Jeeves, the web-based configuration
tool of ETERNITY NE.
• The site of installation should be well-ventilated, moisture and dust free, and not exposed to direct sunlight,
heat or excessive cold.
• The site should not be near any source of electromagnetic noise such as any radio equipment, heavy
transformers, faulty electric chokes of tube-lights, any device having faulty coil, etc.
• The site should be at equidistant from all the extensions to simplify cabling network and reduce cabling
costs.
• The site should allow installation of the system at a clearance of at least 3.5 feet off the ground. Installation
at this height makes preventive or corrective maintenance tasks easy.
• If the system has GSM/UMTS module, the site should have sufficient network coverage available.
Cables
• Select standard, good quality telephone cables with 0.5 mm conductor diameter for the internal as well as
over-head cabling.
• The length of the cables must not be too long. They must have minimum number of joints. This will help
you detect cable faults easily.
Extension Telephones
Select appropriate telephone instruments to be connected to the SLT ports. You may connect any standard
telephone instrument like rotary phone, Pulse/Tone switchable push-button phone, Feature phone or Cordless
phone. So, you can also use your existing telephone instruments.
• Arrange for a separate power point and switch, close to the system.
• Power supply for the system must be separate from other heavy electrical loads like Air-conditioners,
heaters, welding machines, electrical motors, etc.
• The use of a UPS is recommended in case you experience frequent power failures.
Package Contents
• ETERNITY NE with side clamps.
• Two self tapping screws (M7x30 PAN PH).
• Screw Grips
• Power Adapter 24VDC, 2Amp
• Ethernet cable
• CD ROM (System Manual, Quick Start, User Card)
• ETERNITY NE User Card
• ETERNITY NE Quick Start
• Mounting Template
• Warranty Card
The ETERNITY NE is an electronic device. When you handle any electrical or electronic equipment, you are in a
situation that could cause you bodily harm, besides damage to the product. When handling any electronic
equipment, you must be aware of the safety hazards involved in electrical circuitry and the standard practices for
accident prevention.
Take every safety precaution to reduce the risk of fire, electric shock and injury to persons. Read and understand
the precautions, dos and don'ts of handling this product listed below.
These instructions are by no means exhaustive. So, take all the necessary precautions for handling electronic and
electrical appliances. Your safety and that of the others lies in your hands.
Location
• Do not place this product in locations that are close to a water source, on moveable or unstable surfaces,
near high frequency generating devices, and areas where it may be exposed to dust, direct sunlight, heat,
excessive cold or humidity, where shocks or vibration are frequent or strong.
• Do not leave cables exposed on the ground where they may be trampled upon, or get damaged by
entangling with feet or pressure from other heavy objects.
Power Supply
• This product should be operated with proper supply voltage. The DC Adapter (24VDC-2 Amp) provided
with ETERNITY NE works with input voltage ranging between 100-240VAC.
• ETERNITY NE has trunk and extension interfaces. So there are chances of heavy voltages entering the
system from trunk lines or from overhead extensions due to:
• Heavy voltage line falling on the CO line or on the overhead extensions cable. A dangerous surge can
occur if a telephone line comes in contact with a power line.
• Lightning/Thunderbolts.
• Heavy voltage on the cable connecting the relay device to the Digital Output Port of the system.
• Heavy voltage on the cable connecting the door phone to the Door Phone Port of the system.
• Protect ETERNITY NE from lightning and electrical surges by installing Primary Protection/Surge
Protectors on the trunk and long-distance or off-premise extension lines. The product warranty does not
cover damages resulting from lack of primary protection on trunk lines.
Battery
• There is a lithium manganese rechargeable battery installed on the mainboard of the system. It is
recommended that the battery be replaced by the authorised dealer/reseller only. End customers are
advised to have the battery replaced by their dealer/reseller.
• The Earth (Ground) is the most important safety procedure to prevent electrical shocks and fires. It
protects from lightning strikes, electrical transients, static discharges, electromagnetic interference and
electrical hazards.
• The ventilation openings on the sides of the product’s enclosure must not be blocked or covered to prevent
overheating.
• Never insert or push objects of any kind into this product through the openings as they may touch
dangerous voltage points or short out parts which may result in fire or electric shock.
• Do not overload wall outlets and extension cords as this can result in the risk of fire or electric shock.
• This product is equipped with a plug having a third (ground) pin, which fits only into a grounding-type
outlet. This is a safety feature. If you are unable to insert the plug into the outlet, ask an electrician to
replace the obsolete outlet. Do not defeat the purpose of the grounding type plug.
• Do not overload wall outlets and extension cords as this can result in the risk of fire or electric shock.
• Avoid using a telephone (other than a cordless type) during a storm, to prevent electric shock from
lightning.
• Do not use the telephone to report a gas leak in the vicinity of the leak so as to prevent the risk of fire.
External Devices
• When you connect external devices like a door phone or a door lock release, telephone instruments,
cables, connectors, etc., ensure that they are of standard make and good quality, so that the functioning of
the system is not affected.
• Matrix does not guarantee the performance of external devices that are not supplied by it.
• This product must be serviced by a qualified technician only. Call your dealer, if:
• the power supply cord or plug is damaged or frayed.
• liquid has been spilled into the product.
• the product has been exposed to rain or water.
• the product has been dropped or the cabinet has been damaged.
• the product exhibits a distinct change in performance.
Disposal
• This product must be disposed according to the national laws and regulations prevailing in the country
where it is installed.
• Make sure that the RF Antenna is installed at least 20cm away from other electronic and radio
transmission devices.
• Make sure that the RF Antenna is installed atleast 20cm away from people's vicinity.
• People carrying medical implants like cardiac pacemakers are advised to maintain appropriate distance
from the system. They are also advised to avoid being in the vicinity of the product for a long time.
The CO ports of ETERNITY NE provides the interface to connect the ETERNITY NE with the CO Network. The CO
interface supports different standards and features of networks across the world.
• Use standard, good quality, twisted-wire pair telephone cables with RJ11 plugs to connect the CO ports of
ETERNITY NE to the Trunk Lines from your exchange.
The Mobile Port interfaces the ETERNITY NE with GSM/UMTS network. It routes calls made and received over
mobile networks, like a mobile handset2.
The UMTS Mobile Ports of ETERNITY NE also serve as the WAN Interface of ETERNITY NE for VoIP calls, when
used with 3G.
The Mobile Port does not support Fax and network supported services, except CLIR and USSD.
• Make sure power supply is turned off before you begin installation.
• Unscrew and remove the top cover of the enclosure. Keep the screws and the cover aside.
• Select the Mobile port you want to install the GSM/UMTS module.
• The slots for the GSM/UMTS module are located on the side panel and are covered with a flap. Use your
finger or any blunt object to press the slot cover from the outside of the panel to release the flap.
• Now, grasp the flap from inside the enclosure and pull up the flap.
• Gently seat the GSM/UMTS module on the connector on the mainboard such that the SIM holder and the
antenna connector emerge from the respective slots. The connector pins on the module must make
complete contact with those on the mainboard. Do not apply pressure.
• When the module is seated firmly on the connector on the mainboard, secure the module with the screws
on the studs.
• If you have no other module(s) to install, replace the top cover and secure the cover with the screws.
• Get the SIM Card from the GSM service provider of your choice ready.
• Protect the SIM card from unauthorized use with a Personal Identification Number (PIN) on the SIM (in
consultation with the customer/owner of the SIM).
2. Just like mobile handsets, each Mobile Port has a unique IMEI (International Mobile Equipment Identity) number, pasted on the
mobile engine.
• get a mobile handset. Insert the SIM into the mobile handset.
• change the SIM PIN to 1234 (this is the default PIN for both SIM cards used in the system). You can change this
SIM PIN later from ETERNITY NE when configuring the mobile port.
If you do not want to use PIN protection, insert the SIM in the mobile handset and disable PIN protection.
Remove the SIM Card from the mobile handset and insert it in the mobile port’s SIM holder tray.
• press the SIM holder lock (the small yellow button). You may use a blunt pin to press the lock button.
• now, fit the SIM Card on the tray, with its contact side facing up.
• Switch On power supply, if you have finished connecting all the required ports.
• At every power up of the system, it takes about 3 minutes for the Mobile ports to get registered with the
network. Once registration with the mobile network is completed, the mobile port can be used.
The VoIP Interface enables the extensions of ETERNITY NE to connect to the IP network and make Proxy as well
as Non-Proxy (Peer-to-Peer) VoIP calls.
The Mobile Ports of ETERNITY NE can also used as the WAN Interface to connect the system to the
Internet.
Voice Channels
ETERNITY NE supports 8 Voice Channels, allowing you to make and receive 8 simultaneous VoIP calls.
Please note that a SIP Extension-to-SIP Extension call will consume 2 channels, while a SIP Extension-to-
SLT/DKP Extension/Trunk will consume a single voice channel only. SIP Extension to SIP Trunk calls
consume 2 channels
SIP Trunks
In countries, where the provision and use of Internet telephony services and products is prohibited and or
subject to laws, regulations or licenses, the User is advised to comply with such laws and regulations when
installing and using this product.
• Make sure power supply is turned off before you begin installation.
• Unscrew and remove the top cover of the enclosure. Keep the screws and the cover aside.
• Locate the connector of the VoIP module on the mainboard. Remove the screws on the studs for the
module and keep them aside.
• Gently seat the VoIP module on the connector on the mainboard. The connector pins on the module must
make complete contact with those on the mainboard. Do not apply pressure.
• When the module is seated firmly on the connector on the mainboard, secure the module with the screws
on the studs.
• If you have no other module(s) to install, replace the top cover and secure the cover with the screws.
3. SIP Trunks are the same as SIP Accounts. A SIP Account is an account you would get from your VoIP/Internet Telephony Service
Provider much like you would get an email account from your Internet Service Provider.
• You can connect ETERNITY NE to WAN either over Ethernet port (Ethernet WAN) or over Mobile Port 1 or Mobile
Port 2 (Wireless WAN).
While several installation scenarios are possible, only three most common and most typical scenarios are depicted
here.
Ethernet WAN
• Use the RJ45 Ethernet cable supplied for the Network port of ETERNITY to connect the system to the IP network,
which may be Public Internet or a LAN.
• Plug one end of the RJ45 Ethernet cable into the Network Port of ETERNITY and the other end into the Broadband
Router/Modem.
• Plug one end of the RJ45 Ethernet cable into the Network Port of ETERNITY and the other end into the LAN
Switch/Hub.
Mobile WAN
UMTS
LAN Switch/Hub
Ethernet
LAN
The Single Line Telephone (SLT) ports provide the interface to connect as extension phones, any standard, two-
wire, analog single line telephone instrument-rotary, pulse-tone, cordless, feature phones with or without Calling
Line Identification.
1. Decide the number of SLT extensions required and arrange for as many telephone instruments.
Use SLTs equipped with a 'Flash' key, as several of the features and facilities of the ETERNITY require you
to press Flash. If any of the SLTs you have selected does not have a Flash key, tap the Hook switch of the
phone to dial Flash.
2. Use standard twisted wire pair the cables of good quality with RJ11 plugs to connect the analog single line
telephone instruments to the SLT ports SLT1 - SLT14 of ETERNITY NE.
4. Terminate the cables from the SLT ports of ETERNITY on the wall jacks.
ETERNITY NE supports 2 DKP ports, DKP1 and DKP2. You may connect EON48, EONSOFT; or you may connect
EON48 on one port and DSS64, the DSS Console, on the other port.
EONSOFT has two-PC based DSS Consoles that do not consume any physical DKP Port. So, you may use the
DKP ports as follows:
• On DKP 1 connect EONSOFT and use DSS1 and DSS1
• On DKP 2 connect EON48 or another EONSOFT with DSS1 and DSS2.
1. Use standard twisted wire pair cables of good quality with RJ11 plugs to connect the DKP(s).
2. Lead and terminate the telephone cables from the DKP ports of ETERNITY NE to the wall jacks at the
desired locations.
Installing EON48
1. Unpack the box and verify the package contents.
• To mount EON48 on a wall, detach the Foot Stand on the bottom of the phone, as illustrated below.
• Fix two screws of appropriate diameter on the wall, ensuring that they are aligned with the Keyhole
Slots 1 and 2 of EON48. The screws should protrude from the wall to fit into the Keyhole Slots.
• Now, mount the phone with the screws fitting into the keyhole slots.
F ri 1 0 Oc t 1 5 :4 0
2 0 1 R e c e pt io n
Foot Stand
Keyhole Keyhole
Slot 1 Slot 2
CA03 * 0 #
CA02
CA01
• Connect the handset of the EON48 to the phone body using the spring cord.
• To use a Headset (not supplied with the phone), plug any standard stereo headset with 2.5mm single
connector into the headset jack on the left side panel of the phone.
You can also plug in a headset with RJ11 connector into the Headset port at the bottom of the phone.
3. Connect EON48 to the wall jack. Plug one end of the RJ11 cable into the connector of EON48 labeled as
‘Line’. Plug the other end into the wall jack.
4. When the ETERNITY NE is powered ON, the EON will get reset. The EON communicates with the
ETERNITY. The handshaking lasts for 5-6 seconds. The EON model, version and revision number, along
with the message 'Please wait'… appear on the LCD display.
M AT R I X E O N 4 8 - S V 2 R 2
PLEASE WAI T .. .
5. After successful handshaking and reset cycle, if the DKP Parameters have been configured, the LCD
display of the EON will show the extension number and the extension name in a line. The day, date and
time, time zone in the other line.
202 Reception
M on 2 4 A U G 1 2 : 0 0
2. Place the DSS Console next to the DKP, EON, to which it is to be attached.
3. Plug the RJ11 end of a telephone cable into a DKP port of ETERNITY.
Terminate the other end into a wall jack.
4. Plug the RJ11 end of a second cable into the connector on DSS64. Plug the other end into the wall jack.
Installing EONSOFT
To install EONSOFT, you must have a computer with Windows as the operating system. The EONSOFT is
compatible with the following Operating Systems of Windows:
• Windows 98
• Windows XP
• Windows NT
• Windows 2003
• Windows Vista
1. Unpack the box and verify the package contents.
2. Connect the Handset to the dongle. If using a headset, plug the microphone and the speaker connectors
into the dongle.
3. Connect one end of the Communication cable to the COM port of the dongle. Connect the other end of the
communication cable into the COM port of the computer.
5. Switch ON the computer to which EONSOFT dongle is connected. The computer must have Windows
Operating System installed on it.
6. Now, insert the EONSOFT CD-ROM into the CD drive of your computer. The EONSOFT has a self-
executing program, which will automatically install itself on your computer.
7. If the software does not perform auto install on your computer, browse to CD-ROM.
8. The software program will appear, with the Matrix Icon and labeled as 'Matrix-EONSOFT'.
10. After the program has been installed and run, a shortcut will be automatically created and appear on your
desktop.
11. Click the shortcut to open the program. The EONSOFT window will open:
14. Select the COM Port to which the communication cable is connected.
If you select the wrong COM port, a dialog box will pop up on your screen with the message: “COMx is
invalid or busy, please select another COM Port”. Select the right COM Port.
The default DKP settings will appear, i.e. default DKP extension number, date and time will appear on the
EONSOFT display. However, if you have configured DKP port parameters, Name and extension number,
these will appear on the EONSOFT display.
• If this dialog box does not appear on the screen after you selected the COM Port, you may test the
COM Port for data transfer. It is recommended that you test the functioning of the COM Port of the
computer and the communication cable, before you install the EONSOFT.
ETERNITY NE supports up to 16 SIP Extensions. The SIP Extensions function as any normal DKP/SLT extension
of the ETERNITY NE. SIP Extension users can make and receive calls to any extension user of the ETERNITY and
to external numbers over CO, Mobile and VoIP4 networks.
You may register any SIP-enabled device, like an IP-phone, a Soft phone, Analog Phone Adapter, as the 'SIP
Extension' of the ETERNITY NE.
You may also connect the Matrix Extended IP Phone, SETU VP248, as SIP extension of ETERNITY NE. The
Extended IP Phone takes on all the functions of EON48, the proprietary digital key phone of Matrix.
The first 8 SIP extensions are free. To increase the number of SIP extensions, you will require the IP8
License. Make sure you have a valid License Key when connecting more than 8 SIP Extensions. See
“License Management” to know more.
The SIP Extensions may be registered either over Ethernet WAN or over Wireless WAN (Mobile 1/Mobile 2), according
to your preference and your IP network installation scenario.
Ethernet WAN
If ETERNITY NE is connected to a Public Network,
• Connect SETU VP248, the Extended IP Phone, or any Open IP SIP device to the LAN Switch.
When you register any SIP device, other than the Extended IP Phone, on the public network as SIP
Extension, you must configure the Registrar Server Address of ETERNITY NE, the Registrar Server Port,
the SIP ID, Authentication ID and Password in the SIP device.
4. VoIP-to-PSTN routing may be subject to Regulation in your country. Read “Logical Partition”.
DSL Modem/
Router
PSTN
CO
IP
IP
Ethernet
WAN
LAN Switch/Hub
• Connect SETU VP248, the Extended IP Phone, or any standard IP Phone to the LAN Switch.
• You may also register any SIP device (Extended IP Phone or open SIP phone) on the public network as
SIP Extension. In which case, configure Port Forwarding for SIP and RTP on the Router.
GSM/UMTS
LAN Switch/Hub
CO IP
Ethernet DSL Modem/
WAN Router
UMTS
LAN Switch/Hub
CO
Ethernet
LAN
• Connect SETU VP248, the Extended IP Phone, or any standard IP Phone to the LAN Switch.
• You may also register any SIP device on the public network as SIP Extension.
• Decide the location of the Extended IP Phone—within the same network or outside—according to your
installation scenario.
• Log into the web-browser Jeeves. Read the topic “Configuring using the Web-based GUI: Jeeves” under
Configuring ETERNITY NE for instructions.
• Configure DHCP Server on the “Network Parameters” page under Basic Settings of Jeeves.
• Assign an extension number to the Extended IP Phone on the “Extension Numbering Plan” page under
Basic Settings of Jeeves.
• For the SIP extension number you assigned to the Extended IP Phone, configure these parameters on the
“Matrix Extended IP Phone Settings – Location 1” page of Jeeves:
• Enable Matrix Extended IP Phone Mode.
• Enter the MAC Address of the Extended IP Phone.
• Assign the Registrar Server IP Address, as per you installation scenario.
See the topic “SIP Extensions”, under Basic Settings, for instructions.
Now, follow the steps described below to install the Extended IP Phone. The instructions are common for all models
of the SETU VP248. For the purpose of illustration, the premium model, SETU VP248P, has been used.
• Fix two screws of appropriate diameter on the wall, ensuring that they are aligned with the Keyhole
Slots 1 and 2.
• Use wall plugs, if required, to fix the screws. Leave the screw heads protruding from the wall to fit
into the Keyholes.
• Now, mount the phone on the wall, with the screws fitting into the Keyhole slots.
• When you mount the phone on a desk, you can attach the Foot Stand in two ways as illustrated below.
If you attach the Foot Stand at 45°, the phone will be placed in an almost upright position on your
desk.
• Decide which of these positions would work for you best and accordingly attach the Foot Stand.
• Plug the long straightened end of the phone cord into the handset jack at the bottom of the phone
marked with the handset symbol.
• Plug the other (short straight) end of the phone cord into the jack at the bottom of the handset.
4. If you want to use a Headset (not supplied) with your phone, you may plug a headset with a 2.5 mm single
connector into the headset jack on the left side panel of the phone.
OR
You may plug a headset with an RJ11 connector in to the headset port at the bottom of the phone.
6. To connect your phone to a computer on your desk, use an Ethernet cable (not supplied with this phone) to
connect the PC Port of the phone with LAN Port of the computer.
7. Plug the connector of the Power Adapter in to the power jack at the back of the phone5. Use only the
adapter provided with the phone to prevent any damages that may arise from the use of other adapters.
If you want to use Power over Ethernet (PoE), ensure that your LAN supports PoE. Supply power through
an 802.3af connection on the LAN Port of the phone. Do not connect the Adapter!
• All keys with LED, including the Speaker key and the Ringer LED will glow.
• The LCD display will light up and the following message will appear on it, as the phone boots:
Welcom e to M atrix
B ooting ...
• As soon as the ‘Loading...’ message appears on the phone display, press # key.
W e l c o m e t o M a t ri x
L oad ing ...
• Select the firmware Extended - IP Phone. Move the cursor by pressing the Down navigation key V.
• When the cursor is placed under the Extended IP Phone, press Enter key.
Welcom e to M atrix
L oa ding V05R 01_ExtSI P
• After loading the firmware, the phone will prompt you to change Network settings.
If you want to change the Network Settings, you may select ‘Yes’, by moving the cursor to Yes with the Up
key and pressing the Enter key.
• The phone makes DHCP Discovery and fetches its IP Address and Server IP Address from the ETERNITY
NE.
D H C P d i s c o v e r y. . . !
On getting the IP Address, the phone initiates Auto Configuration to download the configuration files from
ETERNITY NE.
T r y i n g f o r C o n f i g. f i le
L a n g u a ge S t r . x m l
• On successful download of all configuration files, the phone attempts to register with ETERNITY NE.
• On successful registration, the phone will display the current day, date and time, the extension number and
name assigned to the Extended IP Phone.
M on 2 0 D E C 1 6 : 5 8 N A
3 0 3 Re c e p t i o n 2
The Door Phone port is available as an optional module on the ETERNITY NE. The Door Phone module contains a
Door Phone Port and a Digital Output Port, providing the interface for connecting and operating a Door Phone and
a Door Lock Release Device with the ETERNITY NE.
• Make sure power supply is turned off before you begin installation.
• Unscrew and remove the top cover of the enclosure. Keep the screws and the cover aside.
• Locate the connector of the Door Phone module on the mainboard. Remove the screws on the studs for
the module and keep them aside.
• The slot for the Door Phone module is located on the back panel and is covered with a flap. Use your
finger to press the slot cover from the outside of the side panel of the enclosure, and release the flap.
• Now, grasp the flap from inside the enclosure and pull up the flap.
• Gently seat the Door Phone module on the connector on the mainboard such that the RJ45 connector of
the module is aligned with the slot, and the connector pins on the module make complete contact with
those on the mainboard. Do not apply pressure.
• When the module is seated firmly on the connector on the mainboard, secure the module with the screws
on the studs.
• If you have no other module(s) to install, replace the top cover and secure the cover with the screws.
The 4-wire Door Phone is to be connected to the Door Phone port of ETERNITY NE. Matrix does not supply Door
Phones.
Specification Value
Specification Value
1. Make sure that the power supply is switched off, before you begin connecting the door phone.
2. Now, plug in the cable supplied with the card into the Door Phone connector. Terminate the free end of the
cables into the main distribution frame.
Refer to the pinout details of the Door Phone port to connect the wires correctly.
1 Relay A Orange-White
DOP-1
2 Relay B Orange
3 Status Green-White
3. The wires of the devices, the Door Phone and the Door Lock, which you want to connect to the ETERNITY
NE, should be terminated in to the Distribution Frame. Refer the pinout details of the ports to connect the
cables.
• You can connect any standard 4-wire door phone. Refer to the pin out details given above to make the
connections.
• Make sure that the door phone you connect conforms with the Technical Specifications of the door
phone port.
• If a Door Lock is to be used in conjunction with the Door Phone, connect the Door Lock to the Digital
Output Port (DOP) of the ETERNITY NE.
The optional Voice Mail System (VMS) module of the ETERNITY NE provides mailbox facility to all extensions of
ETERNITY NE.
Each Mailbox has the capacity of storing 254 messages. The size of each Mailbox is set by default to 5 minutes.
The maximum message length for each mailbox is set by default to 15 seconds.
The VMS module requires a license. The VMS module is delivered as Pen Drive, containing VMS configuration
files, and voice messages for prompts and greetings. The Pen Drive is also the storage device for mailbox
messages.
The Pen Drive is shipped separately as per the order of individual customers, along with the License Voucher.
• Make sure power supply is turned off before you begin installation.
• Unscrew and remove the top cover of the enclosure. Keep the screws and the cover aside.
• Locate the USB port on the mainboard. Connect the Pen Drive to the upper USB port on the mainboard.
• If you have no other module(s) to install, replace the top cover and secure the cover with the screws.
• Connect a computer to the Ethernet Port of ETERNITY NE the ethernet cable supplied for the port.
• Open a Web browser on the computer to access the embedded Web server, Jeeves.
• Activate the License Voucher for the VMS. See “License Management” under Advanced Settings for
instructions.
• Configure VMS. To know more, see “Voice Mail System” to for detailed instructions.
Power ON
• Connect the power adapter to ETERNITY NE.
• Switch on power supply and wait for the Reset Cycle to complete.
Reset Cycle
• Reset Cycle (Power-ON Self Test) takes about 2 minutes to finish.
• When the system becomes stable after power on, the status LED, STS, blinks Green (1 second ON, 1
second OFF).
• The LEDs of the DSS keys of the Digital Key Phone(s) attached to the system are turned on in a
sequence. The handshaking between DKP and ETERNITY lasts for 5-6 seconds.
• At the end of the Reset Cycle, the default extension numbers loaded by the system, the date and time of
the Real Time Clock of the system will appear on the display of the DKP connected.
• Mobile Ports take about 3 minutes to get registered with the network.
You may now access the web-based programming tool, Jeeves, and configure ETERNITY NE.
Configuration Modes
ETERNITY NE can be configured at two levels: System Engineer and System Administrator.
A distinct set of features and facilities can configured at each of these levels.
Only the System Engineer, the person who installs, configures and maintains the PBX must be allowed access to
this mode.
Access to the SE mode is protected by means of a password, referred throughout this document as the SE
Password.
SE Password
The SE password is a 4-digit code used to prevent unauthorized access and alterations or misuse of the features
and facilities. As this password is meant for restricting access to the SE mode, it must be kept strictly confidential
with the System Engineer. Default password:1234.
The SE password can be changed. Refer the topic “System Security” for instructions on how to change the SE
password.
If you forget the SE password, you can restore the default SE password. Read the topic “Default Settings” for
instructions on restoring the default SE password.
You are advised to record the SE password at a safe place, where it can be accessed by you (the System
Engineer) to avoid the inconvenience of restoring the default SE password.
SA Password
The access to SA mode may be protected by means of a password, referred to as the SA password in this
document.
The SA password is 4-digit code for preventing unauthorized access to the SA mode. The default, SA Password is
1111.
The SA password can be changed by the System Administrator and the System Engineer. However, if the System
Administrator forgets the password, the default SA password or an entirely new password can be issued only by
the System Engineer.
Refer the topic “System Security” for instructions on how to change the SA password.
Configuration Tools
The ETERNITY NE can be configured using the following tools:
ETERNITY NE can be configured both, 'on site' and from a remote location without its functioning being affected.
ETERNITY NE provides a Graphic User Interface (GUI), Jeeves, the proprietary Web-based configuration software
of Matrix. It is an HTTP server built into the ETERNITY NE.
Connecting a computer
You may connect a standalone computer to ETERNITY NE or grab any computer connected in the LAN as
ETERNITY NE.
Connect a standalone computer to ETERNITY NE, when installing the system for the very first time. You
may connect it to a computer on LAN at a later stage, once you have finished installation and configuration
of the system.
• Plug one end of the RJ45 cable supplied into the Ethernet Port of ETERNITY NE. Plug the other end into
the LAN port of the computer.
• Make sure the IP Address of the computer and the Ethernet Port of ETERNITY NE do not conflict and both
are in the same Subnet.
The default Subnet Mask of the Ethernet Port of ETERNITY NE is: 255.255.255.0
• Make sure a web-browser, either Internet Explorer 7 or 8 or Mozilla Firefox, is installed on the computer.
• Enter the default IP address 192.168.1.101 of the Ethernet Port ETERNITY NE in the address bar of the
browser.
Set the Appearance Settings of your computer's screen to the resolution of 1024 x 768 pixels for full view
of the pages.
• From the Login as box, select the login mode you want to enter: System Engineer or System
Administrator.
• Enter 1234, the default System Engineer Password, in the Password field.
If the password has been changed, use the new password.
• On the left pane are the links, Basic Settings, Advanced Settings and Administration.
Basic Settings break down the complexities of configuration and are sufficient to get your system into
operation.
Advanced Settings, enable you to configure the advanced features and facilities of ETERNITY NE.
• Click the link of the desired parameter. The page of the respective parameter opens.
• Change the settings on the page to the desired values, and click Submit button to save.
• To exit the SE configuration mode, click the Logout on the top of the page.
• As many as four System Engineers can simultaneously log in and configure the Basic Settings and
Advanced Settings. However, it is recommended that multiple login be avoided.
• Enter 1111, the default System Administrator Password, in the Password field.
If the password has been changed, use the new password.
• On successful login, the Feature Menu for the SA mode appears on the left pane as links.
While four persons can simultaneously log in to the SA mode, it is recommended that you avoid multiple
simultaneous SA login sessions.
It is possible to configure the ETERNITY NE from any location using Jeeves. You can use Jeeves to configure the
system On site (where it is installed) and Off site, from a remote location.
A few parameters of ETERNITY NE may be configured by dialing the relevant command strings from a telephone.
The telephone may be any Single Line Telephone (SLT) or the Matrix proprietary Digital Key Phone (DKP), or the
Extended IP Phone, connected as an extension of the ETERNITY NE.
For the ease of operation, you may use a DKP/Extended IP Phone instead of an SLT. Using a DKP/Extended IP
Phone has the advantage that
• you can view the command strings that you have keyed in on the phone’s display;
• you will get prompts and confirmatory messages on the phone's LCD display, in addition to confirmation
tone played to you.
You can enter the System Engineer as well as the System Administrator mode to configure the system using a
telephone.
It is possible to configure the ETERNITY NE from any location using a Telephone. You can use a
Telephone connected as an extension of the ETERNITY NE to configure the system On site (where it is
installed).
You can also configure the system using a Telephone Off site, i.e. from a Remote location, using the
“Direct Inward System Access (DISA)” feature of the ETERNITY NE. You can access both the System
Engineer mode as well as the System Administrator mode from the remote location.
6. You may use the default SE password, 1234, if the password has not been changed already. If the password has been changed,
use the new password.
• The system accepts and executes the command immediately, but it takes approximately 2 minutes to
save a command. So, it is advisable that you do not turn OFF the system for 2 to 3 minutes after
entering the last command.
• There is no restriction on the number of persons who can simultaneously enter SE mode from a
Telephone and configure the system.
The Operator or Receptionist, who usually administer the system, must enter the System Administrator (SA) mode
by dialing 1#92-SA Password.
After entering SA mode, you may dial command strings referred to as SA Commands from a telephone.
SA Commands
SA commands consist of a prefix string 1072, followed by the Command string. For example: the SA command for
setting Do Not Disturb for an extension is 1072-001-extension number-1, where 1072 is the prefix string and 001
is the command string.
The Prefix string in the SA Command (1072) can be changed by the System Engineer. However, the command
strings of the SA Command (001 in the above example) cannot be changed.
The command for entering SA mode (1#92) is also non-configurable. See the topic “Access Codes” under Features
and Facilities.
To know how to use change feature settings with SA Commands, please refer the description of individual features
under “Features and Facilities”.
• You can enter SA mode only from extensions which have the features 'SA mode' and or 'SA Extension'
enabled in their Class of Service.
• When the feature 'SA Extension' is enabled in the Class of Service of an extension, the extension will
always be in SA mode. You do not need to enter SA mode by dialing the SA password. You can enter
the SA mode by dialing the SA command prefix string.
• When the feature 'SA Mode' is enabled in the Class of Service of an extension, dialing of the SA
Password is required to enter the SA mode. SA Commands can be dialed only after successfully
entering the SA Mode.
• There is no restriction on the number of persons who can simultaneously enter and operate from the
SA mode using a telephone.
You can exit from the SA mode automatically or manually. To exit the SA mode automatically, you must
configure the SA Mode Timer. This Timer can be set to a desired duration between 000 and 255 minutes.
On the expiry of the set time, the system disconnects the extension phone from the SA mode. By default,
the SA Mode Timer is set to 003 minutes. The Timer is loaded every time a new SA command is issued.
• You can also exit the SA mode before the SA Mode Timer expires by dialing this command. If the SA
Mode Timer is set to 000 minutes, you can exit the SA mode only by dialing the command 1#92.
The SA Mode TImer can also be configured from the SA pages of Jeeves.
7. You may use the default SA password, 1111, if the password has not been changed already. If the password has been changed,
use the new password.
When you dial this command, the system will check if the facility 'SA Mode' is enabled in the Class of Service allowed to the exten-
sion from which you have dialed this command. If the SA mode is not allowed, an Error Tone will be played. The Error Tone will be
played also when the SA password is entered incorrectly.
If the facility 'Allow SA Commands' is enabled in the Class of Service of the extension you are using, you can skip this step and
directly dial the SA Command (1072-<Command String>).
File Transfer Protocol (FTP), is a standard network protocol, used to exchange and manipulate files over a TCP
computer network such as the Internet.
FTP is the simplest way to exchange files between computers on the Internet. Like the Hypertext Transfer Protocol
(HTTP), which transfers displayable Web pages and related files, and the Simple Mail Transfer Protocol (SMTP),
which transfers e-mail, FTP uses the Internet's TCP/IP protocols, and is commonly used to transfer Web server for
everyone on the Internet, download program and other files to your computer from other servers.
Using FTP, you can also update (delete, rename, move and copy) files at a server. For this, you need to log on to
an FTP server.
ETERNITY supports an embedded FTP server which can be used for these purposes:
Configuring ETERNITY using FTP is meant for Installers who want to complete system configuration at their end or
are unable to configure the system on-site at the customer's end.
The advantage of using FTP for configuring the system is that Installers can complete the entire system
configuration as per their customer's requirement at their end, copy these configuration files, and then upload these
configuration files on to the customer's system.
Further, once the configuration file has been created, Installers only need to make the desired changes in the
relevant files and upload the updated files again.
If you are the Installer, before you configure the users’ system using the FTP server, make sure you complete the
following tasks at your end:
• Now, copy the system configuration you just completed for the user’s system on a CD or a Pen Drive using
the embedded FTP server. To do this, you may use either Windows FTP or FireFTP. See “Configuration
Upload” for step-by-step instructions.
If you have multiple customers and you want configure your customers' systems using FTP, you are
recommended to tag the names of the configuration folders you create for your customers with some
identification, like name and date.
• Open the web browser on the computer connected to the customer’s ETERNITY NE.
Basic Settings help you with the basic configuration of the system in easy steps. The Basic Settings break down the
complexities of configuration and cover as much as 80 per cent of all your basic installation and configuration
requirements.
The country-specific default values assigned to the parameters in Basic Settings are sufficient to set your system in
operation. You may need to change the settings of only those parameters that are peculiar to your installation
requirement.
1. Open Jeeves (see “Configuring using the Web-based GUI: Jeeves”). The Login page will open.
The links Basic Settings and Advanced Settings appear on the left pane.
• using the Wizard. A special configuration Wizard will lead you logically, step-by-step through the
configuration of the parameters listed above.
Or
• using the Basic Settings links for selective configuration. You can choose the parameters you want to
configure, the order you want, and accordingly select the parameter links and configure the settings.
• To navigate the Wizard pages use the Next and Back buttons.
• When you press the Next button, the changes on the current page are saved and the Wizard takes you to
the next page.
• When you press the Back button, you will be prompted to save changes made on the current page.
• The More button and the Less button on the page allow you to expand and collapse respectively,
the parameters on the page.
• You may exit the Wizard at any time by clicking the Quit button . The changes you made before you
exit will be saved.
• Get familiar with the buttons and icons listed below before you begin to change the settings of the
parameters on each page.
More: displays all the parameter links on the page.
• Set the desired values on this page and click Submit button to save.
You may use the Wizard or selectively configure the Basic Settings pages, whichever works best for you.
When configuring the system for the first time, you may use the Wizard. When you want to make changes
after the configuration has been done, you may selectively configure the Basic Settings parameters using
the links and sub-links.
Do not allow simultaneous login and system configuration using the Basic Settings pages. As the system
updates configuration changes last submitted, configuration changes made by one person may be
overwritten by those made by another.
Read on for instructions to configure the Basic Settings using the links.
The ETERNITY NE can operate anywhere in the world. It provides “Default Settings” to match country/region-
specific requirements of users around the world.
The Default Settings are factory-set values for system and feature configuration.The system is designed to work
efficiently in any country with these default settings.
Default Settings also speed up the process of system configuration, as these are sufficient for getting the system
into operation.
To load the country-specific Default Settings, you must select ‘Region’, i.e. the country/region in which the system
is installed.
India is selected as the default ‘Region’. So, if you are installing ETERNITY in a country other than India, change
Region.
To configure.
1. Select the 'Region', i.e. the country where the system is installed, from the list.
System resources (number of trunk and extension ports supported) vary by configurations of ETERNITY NE. It is
common for system resources to be used below capacity in a first-time installation.
To make the task of configuring easier and more focussed, ETERNITY allows you to define System Pre-requisites:
the system configuration you are using (NE2, NE3, NE4, NE6), the number of trunk and the number of extension
ports you want to configure. Accordingly, Jeeves will show only as many trunk and extension ports that you have
specified.
To configure,
The Customer Name you enter in this field will appear as header on the various System Reports generated
and printed by the ETERNITY NE like SMDR Incoming, Outgoing and Internal Call Reports, Alarm Status
reports, etc.
The Customer Name may consist of a maximum of 80 alphanumeric characters, including punctuation
marks. So, you can enter also the organization's address along with the Customer Name.
3. Enable On-Site Configuration, if you want Basic Settings pages to display only those port types and ports
that are on-board (connected). Default: Disabled.
Enable the 'On-Site Configuration?' flag when you are configuring the system at the installation site.
4. Select as Model Type, the configuration of ETERNITY NE that you are using: NE2, NE3, NE4, NE6.
When you select the Model Type, Jeeves displays on relevant pages, the port types according to the
system capacity (the maximum number of ports) supported by your model.
For example, if you selected ETERNITY NE6 as Model Type, the page will show maximum CO Ports as 6,
maximum DKP extensions ports as 2, and maximum SLT ports as 14. If you selected ETERNITY NE2 the
page will show maximum number of ports as 2 for CO Ports, 2 for DKP ports, and 4 SLT ports.
5. Define the Number of Ports Used for each Port Type: CO, DKP, SLT, Mobile, SIP from the respective
combo boxes.
For example, if you want 4 CO Trunks, 1 DKP extension, and 8 SLT extensions to be used, select the
same numbers from the combo box.
If you want to use voice mail, or connect a device (Door Lock Release) to the Digital Output port, select
'Yes'.
If you have enabled On-Site Configuration, the fields for 'Number of Ports Used', will be populated with
exactly the number the system has detected. The fields for port types which are not available on-board the
system are displayed as non-editable fields.
The default Extension Numbers and Feature Access Codes8 appear on this page.
The default Feature Access Codes that appear on this page are country-dependant.
To configure,
2. You may re-assign extension numbers to SLT, DKP, SIP Extensions, and the Door Phone port.
The length of the extension number may be 6 digits. The digits 0 to 9, # and * are allowed.
When you change the extension numbers, make sure that they do not clash with any other Feature Access
Codes in the dialing phase (codes starting with 1, 9, etc.). To know more, refer the topic “Access Codes”
and “Conflict Dialing” to know more.
If you have installed Voice Mail System, do not configure * (star) in the Extension numbers.
8. Feature access codes are short digit codes used invoke a feature or function. See “Access Codes”.
3. You must resolve the conflict. Jeeves will not allow you to continue if you do not resolve conflict.
If you want to change the Feature Access Code that is conflicting with the extension number you have
assigned, go to Advanced Settings and change the Feature Access Code.
4. Assign names to extensions. The name you assign may be the name of the person who will use the
extension. Extension names may consist of a maximum of 18 characters.
This page displays only those extension ports available in the system and the number of ports you have
defined as Number of Ports Used in the Pre-requisites page.
In the case of SIP Extensions, the number you configure as Extension Number will be considered as the
SIP ID, the Authentication ID and the Authentication Password.
5. If you have finished assigning numbers and names, click Submit button.
To assign the extension numbers and names all over again, click the Clear Button on this page.
To default all extension numbers, click the Default button on this page.
6. Reassign Feature Access Codes, if required. The new access code may be a single digit or a sequence of
a maximum of 6 digits. Digits 0 to 9,# and * are allowed.
You can assign names to trunk ports for easy identification of the trunks. On this page, trunk ports appear with their
default names and port numbers, in ascending order of the port number.
The default trunk name consist of the trunk type (CO Trunk, MOBILE, SIP) and port number. For example, the
name of the Two-wire Trunk connected to port number 1 is displayed as CO-1.
The Wizard will display only those Trunk port types available in the system and the number of Trunk ports
you have defined for each trunk port type earlier on the Pre-requisites page.
To configure,
2. You may assign names to each trunk port type for easy identification. The name be derived from the name
of the Service Provider.
Certain features of the ETERNITY NE like Operator, Class of Service, Toll Control, Outgoing Trunk Access, among
others9, require its trunks and extensions to behave differently according to the time of the day: Day (working
hours) and Night (non-working hours).
For example,
• incoming calls are to be routed to the extension of security personnel, instead of the Operator when the
office is closed.
• certain features in the Class of Service are to be allowed during the day (working hours), but not at night
(non-working hours).
• access to outgoing long distance calls is to be denied in the night hours.
• trunks must play a different greeting message to the callers during the day and a different message in the
night hours.
A Time Table is a schedule of Working Days and Day time (working hours) for the entire week. In a Time Table, you
can define the working days and working hours for the entire week.
Time Tables are applied to extensions and trunks defining the working hours and Day time for the entire week, so
that the system can execute the time-dependent features and facilities according to the Time Table.
ETERNITY NE offers 4 different Time Tables: the System Time Table (applied to all trunks and extensions and
Operator by default), and customized Time Tables - 1, 2, and 3 which you can set to your preferences and apply to
trunks and extensions.
You can assign different Time Tables to different trunks and extensions.
To configure,
1. Click the Time Table link.
9. Direct Inward Dialing, Direct Inward System Access, Trunk Landing extensions, Door Phone Call Routing.
By default, System Time Table has Monday to Saturday as Working days, and working hours are defined
as: 09:00 to 18:00 for all working days. Sunday is a non-working day, with working hours set to 00:00.
The system considers the time period other than the working hours you define as Night (non-working
hours).
If the System Time Table suits you, retain it. You may redefine the Working days and Working hours. You
can also define different working hours for each day of the week. For example, you may define the
Working hours from Monday to Friday as 09:30 to 18:30, and for Saturday, from 09:30 to 15:00. If your’s is
a 24x7 business, you may select Sunday as working day and set the working hours.
4. To define the working hours, select the Start Time and the End Time from the combo boxes.
5. Click Submit.
6. If you want to use a Customized Time Table, click the tab Customized Time Table-1.
7. Define the working days and the working hours - start time and end time - for each working day.
8. Click Submit.
9. If you want to use the other customized time table, click their tabs and follow the same instructions as
above.
In the context of a PBX, users understand the term 'Operator' as a person who handles multiple simultaneous calls
and functions as the link between callers and called parties.
For the PBX however, an 'Operator' is a Routing Group, i.e. a group of extensions to which calls made by
extensions by dialing '9' are to be landed. This also includes Direct Inward Dialing calls on trunks during which the
caller dials '9'.
Depending on the amount of call traffic to be managed, there may be more than one Operator extension. Also,
different Operator extensions may be assigned according to the time of the day. For instance, during the day
(working hours) call may be landed on the extension of the Receptionists/Front Desk Personnel. During the night
(non-working hours), calls may be landed on the extension of the Security Personnel.
To meet this requirement, ETERNITY allows you to configure the Operator for Day and for the Night.
3. Select the Extensions which are to be used as Operator. These may be SLT, DKP or SIP extensions.
4. For each extension you have selected, set the Ring Timer. This timer defines the time for which the
extension, on which the call lands, should ring. Default: 015 seconds.
5. For each extension you have selected, you may set Continuous Ring, if you want the extension to ring till
the incoming call is answered. Default: Disabled.
When Continuous Ring is selected, the first extension in the Operator group you have created will continue
to ring, even as the system hunts for other extensions in the group to land the call. If the call still remains
unanswered, the system will return the call to the first extension once again. This flag has no relevance, if
there is only one Operator extension.
6. Enable Rotation, if you have selected more than one extension as Operator. Default: Disabled.
When you enable Rotation, each new call lands on the subsequent extension10 in the group next to the
one that received the last call. This ensures equal distribution of incoming calls to all the destination
extensions in the Operator extension group.
Rotation has no relevance if the Operator group has only one member extension.
10. The extension next to the one that received the previous call.
9. Click Submit.
For each extension you have selected as Operator for the Day and Night,
• define the Class of Service11 for the Day and for Night.
• set Toll Control12 for the Day and for Night.
• select Outgoing Trunks13 for the Day and for Night.
• set Priority14 for the Operator extension for the Day and for Night.
You can set Class of Service, Toll Control, Outgoing Trunks and Priority for the Operator extension(s)
when you configure the parameters of different extension types: SLT, DKP and SIP Extensions.
The ETERNITY NE supports a maximum of 14 Single Line Telephone (SLT) extension ports, depending on the
configuration you are using.
The number of SLT extensions available to you for configuration depends on the number of SLT ports supported by
your configuration of ETERNITY and the number of SLT ports you have specified on the “Pre-requisites” page.
If you have enabled 'On-Site Configuration', the system will provide you only those SLT ports that are actually
present in the system for configuration on this page.
On this page,
More: Click this button to view all parameter links on the page.
Less: Click this button to view only the essential parameter links on the page.
Expand: Click to expand a link to display all parameters under the link.
Collapse: Click to collapse a link. Hides all parameters under the link.
To configure another SLT extension, click the SLT Extension number tab.
Class of Service
Define the Class of Service for the SLT extension for Day time and Night time.
Toll Control
1. Click Toll Control to expand options. Set the desired Toll Control for the SLT extension for the Day and
Night.
2. Select the type of Calls Allowed during Day: All Calls, No Calls, Local Calls, Regional Calls, National
Calls, and Limited Calls 1, 2, 3. Default: All Calls.
3. Select the type of Calls Allowed during Night: All Calls, No Calls, Local Calls, Regional Calls, National
Calls, and Limited Calls 1, 2, 3. Default: All Calls.
4. If you have not configured the allowed and denied number list for the Type of Calls you selected as Toll
Control or if you want to add to the existing list, you may do it now.
• Click the settings icon.
• The Number Patterns page will open in a new window.
• Configure the Allowed and Denied Numbers.
• Click Submit.
• Close the window after you have configured the list.
For each Toll Control Level from 0 to 3, you must assign 'Call Privilege’15. For each Call Privilege, you need to
configure the corresponding numbers strings to be allowed and number strings to be denied. See “Dynamic Lock”
to know more about this feature.
5. Select the call privilege for Calls allowed for Lock Level 1: No Calls, Local Calls, Regional Calls, National
Calls, All Calls.
6. Select the call privilege for Calls allowed for Lock Level 2: No Calls, Local Calls, Regional Calls, National
Calls, All Calls.
7. Select the call privilege for Calls allowed for Lock Level 3: No Calls, Local Calls, Regional Calls, National
Calls, All Calls.
The Lock Levels on this page are based on the allowed and denied number lists of Local, Regional,
National, and Limited Call numbers you configured on the “Number Patterns” page.
8. If you have not configured the allowed and denied number list for the Calls allowed/denied for the selected
Lock Level or if you want to add to the existing lists, you may do it now.
• Click the settings icon.
• The Number Patterns page will open in a new window.
• Configure the Allowed and Denied Numbers for Local, Regional, National, International, and Limited
Calls.
• Click Submit.
• Close the window after you have configured the list.
15. The Call Privilege types are: No Calls, Local Calls, Regional Calls, National Calls, International Calls and Limited Calls.
Outgoing calls (to external numbers) are made by dialing Trunk Access Codes (TAC).
• TAC for users worldwide are: 0, 5, 61, 62, 63, 64.
• TAC for users in USA are: 9, 5, 81, 82, 83, 84.
For each TAC, you need to select the Outgoing Trunks. All external calls made by dialing a particular TAC will be
routed through the outgoing trunks you selected for that TAC.
You can also apply Least Cost Routing logic on the selected trunks, so that ETERNITY will route the outgoing call
through the trunk that costs the lowest for the call.
2. Select Trunks allowed for ‘0’ dialing. The outgoing call will be routed through the selected trunks when
the extension user dials TAC ‘0’.
On the left, the trunks appear with their names (if configured in “Naming Trunks”) and port numbers in a
sequence, starting with CO trunks, followed by Mobile trunks and SIP trunks.
If you have not assigned any names to the trunks, they will appear with their default names (CO, MOB,
SIP) and port numbers.
If you have enabled On-Site Configuration, only those trunks that are connected will appear in the box.
• To select a trunk, place your cursor on the desired trunk, and click the Select>> button.
Or
• Press the ctrl key and click the left mouse button to select multiple trunks.
• You may change the sequence of the trunks you selected, if required, using the Up and Down arrow
buttons on the right display box.
• You can also delete trunks from the ones you have selected.
• You may enable Rotation, if you have selected more than one trunk. Default: Disabled.
When you enable Rotation, each new outgoing call is routed through the subsequent trunk in the
group16. This ensures equal distribution of outgoing call traffic on all trunks.
16. The first call through the first trunk, the second through the second, the third through the third trunk, and so forth. Thus each new
call is routed through the trunk next to the one that routed the previous outgoing call.
All the trunks appear in the field Trunks allowed for ‘0’ dialing, in the sequence you selected, separated
by commas.
3. To apply Least Cost Routing on the Trunks allowed for 0 dialing, select the desired LCR method from the
combo box:
• Number Based: Choose this option if the service providers of the trunks you selected offers different
tariffs according to area or distance, or phone numbers dialed.
• Time Based: Choose this option if the service providers of the selected trunks offer a different tariff
according to the time of the day.
• Time + Number Based: Choose option if the service providers of the selected trunks offer different
tariffs according to the time of the day as well as area/distance.
• Service Provider Based: Choose this option if the same service providers of the selected trunks offer
different rates for calls made to numbers within their own network and for calls made to numbers of
another Service Provider's network.
Configure LCR method that you selected for the trunk group.
4. To configure Least Cost Routing method you selected, click the link LCR settings. A new window opens.
• Go to the LCR method you selected for Trunks allowed for 0 dialing.
• Configure the LCR method. See Least Cost Routing under Advanced Settings for instructions.
• Click Submit. The window closes.
5. Select Trunks allowed for ‘5’ dialing. The outgoing call will be routed through the selected trunks when
the extension user dials TAC ‘5’. Follow the same steps as described above.
• Double-click the field. A multiple selection box opens.
• Select trunks, placing your cursor on the desired trunk, and clicking the Select>> button.
• Change the sequence of the trunks you selected, if required, using the Up and Down arrow buttons on
the right display box. Delete trunks from the ones you have selected, if required.
• Enable Rotation, if you have selected more than one trunk. Default: Disabled.
• Click OK.
All the trunks appear in the field Trunks allowed for ‘5’ dialing, in the sequence you selected, separated
by commas.
• Choose a Least Cost Routing method, if you want to apply it on the trunks. Default: OFF.
• Configure the Least Cost Routing method you selected by clicking LCR Settings link.
6. Select Trunks allowed for ‘61’ dialing. The outgoing call will be routed through the selected trunks when
the extension user dials TAC ‘61’.
• Follow the same steps as described above to select the trunks, enable Rotation, and apply Least Cost
Routing.
8. Select Trunks allowed for ‘63’ dialing. The outgoing call will be routed through the selected trunks when
the extension user dials TAC ‘632’.
• Follow the same steps as described above to select the trunks, enable Rotation, and apply Least Cost
Routing.
9. Select Trunks allowed for ‘64’ dialing. The outgoing call will be routed through the selected trunks when
the extension user dials TAC ‘64’.
• Follow the same steps as described above to select the trunks, enable Rotation, and apply Least Cost
Routing.
3. Keep the option System should ask Password while Accessing Mail Box enabled, if you want
password protection for the mailbox.
By default, access to the mailbox is password protected. The “User Password” is required to access the
mailbox. Whenever the mailbox owner accesses the mailbox, the VMS will ask for the (user) password.
Since a Mailbox can be accessed using the default User Password, 1111, extension users who are
assigned a mailbox are recommended to change their User Password to a unique 4 digit number to
prevent unauthorized access to their mailbox.
4. Define the Mail Box Size (minutes) for the extension. You may change the mailbox size to any desired
value from 001 to 999 minutes. Default: 5 minutes.
The VMS stops recording the message of the callers if it exceeds the maximum message length, and
stores only that part of the message that was recorded within the maximum message length limit.
6. Select the option for delivery of new messages When Mail Box is Full. You may select any of the
following message delivery options:
• Do not offer to Record a Message: The VMS will not allow the caller to record a message by
declining delivery of the message.
• Deliver New Message in General mailbox: The VMS will record the message in the General
mailbox. A General mailbox is a shared mailbox between extension users.
Only extension users who have General Mailbox in their “Class of Service (COS)” are allowed to
access it.
When you select this option, make sure that General Mailbox is enabled in the Class of Service of
the SLT extension. Refer “Class of Service (COS)” for instructions.
• Overwrite Old messages: The VMS will overwrite the old messages to record the new message in
the mailbox. The VMS starts overwriting the oldest message first.
7. To play to the SLT extension user (mailbox owner), message details such as Date and Time (when the
message was recorded by the caller), the caller’s number17, the extension number dialed by the caller18,
select Play Message Detail after Delivery of Message.
You may select from the following options for Play message details:
• Never: The VMS will not play message details to the extension user after playing the message.
• Always: The VMS will play message details to the extension user after playing each message.
• On Demand: The VMS will play message details to the extension user only when the user requests
it. On completion of each message, the VMS will prompt the extension user to press a digit for date
and time stamp. When the mailbox owner presses the digit, the VMS will play the message details.
Default: On Demand.
8. Select the Voice Mail/Message Wait Notification Type for the extension. The VMS will notify the
extension user of the new messages in the mailbox and the message wait set by another extension user.
• Stuttered Dial Tone/Voice Message: When the extension user goes OFF-Hook, s/he will hear a
voice message, if a pre-recorded Voice Module has been assigned for Message Wait Notification. If
no voice module is recorded and assigned, the extension user will hear a stuttered dial tone
instead.
17. The number of person who left the message in the mailbox.
18. The number of the extension user for whom the message is intended.
ETERNITY can play only 4 Voice Modules simultaneously. The Voice Module for Message Wait Notification
will not be played if there are already 4 being played simultaneously. In which case, Stuttered Dial Tone will
be played for Message Wait Notification, when the extension user goes OFF-Hook.
• LED Lamp: If the SLT has a 'Message Wait' lamp, you may select this option as Message Wait
Indication. The lamp will blink continuously and will be turned off when the extension user has
retrieved all the waiting messages.
• Ring: The extension will ring for the duration of the Message Wait Ring Timer (configurable;
default: 30 seconds), for as many times as the Message Wait Ring Count (configurable; default: 10
times), at the interval set as the Message Wait Ring Timer Interval (configurable; default: 30
minutes).
When the extension user answers the call, the VMS informs the user of the new message and
allows the extension user to access it.
9. To notify the extension user via e-mail about new messages in the mailbox, select the check box Voice
Mail Notification via E-mail.
• To mail the new voice message to the extension user as attachment, select With Attachment (.wav
file).
• To mail the notification only to the extension user, select Without Attachment.
• Type the E-mail Address of the extension user to which the Voice Mail notification is to be sent. The e-
mail ID may consist of up to 64 characters. Default: blank.
10. To assign Department Group mailbox to the SLT extension, select the check box Assign Mail Box of
Department Group. Default: Disabled.
You can assign the Mailbox of a Department Group to SLT extensions, even to those SLT extensions that
are not included in the Department Group. See “Department Call”.
• Select the Department Group whose mailbox you want to assign to the extension from the combo box
Specify Department Group.
2. Set AC Impedance for the SLT port to match the AC impedance of the telephone instrument connected to
it. Default:
However, the ETERNITY allows you to connect instruments with AC impedance other than 600.
Flash is generally required to enable the SLT user to use features like Hold, Transfer, etc. Flash can be
dialed using the Flash key (if available) on the SLT or by tapping the hook switch.
Flash is breaking of Loop Current for a specific time period. ETERNITY considers that an extension has
dialed flash, if it detects breaking of the loop current for duration of the Flash Timer. The Flash timer is
configurable. The range of this timer is from 83 (minimum) to 999 (maximum) milliseconds. By default,
Flash Timer is set to 600 milliseconds.
You can program the Flash Timer according to the Flash Timer supported by the SLT instrument. If the
SLT instrument breaks the loop current for more than the programmed Flash Timer, the system will
consider it as a call disconnection.
4. Select the appropriate CLIP Type according to the CLIP Type supported by the telephone instrument
connected to the SLT port.
ETERNITY NE supports 3 signaling protocols for CLI on the SLT port: DTMF, FSK-V.23, and FSK-
BellCore. Default: DTMF.
5. Define the Minimum Caller ID Digits according to number of digits supported by the telephone instrument
connected to the SLT port.
6. Select the appropriate Ringing Signal to match the type of ringing current expected by the telephone
instrument connected to the SLT port.
You may select from these Ring Types: Sinusoidal, Trapezoidal, Low Sinusoidal, Low Trapezoidal.
Default: Trapezoidal.
7. Increase or decrease the Loop Current (mA) according to the Loop Length19, i.e. the length of the
telephone cable between the wall jack (into which the SLT telephone instrument is plugged) and the SLT
port.
The SLT Port provides the telephone instrument connected to it Loop Current of 25, 30, 35 and 40 mA.
Default: 25mA, which is sufficient to support Loop Length of 1 kilometer.
8. Set the value of the OFF-Hook Current (minimum) according to the current drawn by the SLT instrument
connected to the SLT port.
ETERNITY NE supports OFF-Hook detection for all types and brands of SLT instruments by providing for
configurable values for threshold current for OFF-Hook detection: 10mA, 12mA, 14mA, 16mA and 18mA.
Default: 12mA.
When an SLT instrument draws current equal to or greater than the configured threshold value of current
for OFF-Hook detection, ETERNITY will consider the SLT instrument as OFF-Hook and will offer dial tone
to the SLT.
9. Set the value of ON-Hook Detection (maximum) according to the current drawn by the SLT instrument
connected to the SLT port.
ETERNITY detects ON-Hook state of for all types and brands of SLT instruments by providing for
configurable values for threshold current for ON-Hook detection: 10mA, 12mA, 14mA, 16mA and 18mA.
Default: 10mA.
When an SLT instrument draws current equal to or lower than the configured threshold value of current for
ON-Hook detection, ETERNITY will consider the SLT instrument as ON-Hook and routes calls to this port.
SLT instruments also vary by the level of current they draw during the normal 'idle' state and when Flash is
dialed20 (the simulated idle state). So, when the Flash key of an SLT instrument is pressed, and if the
instrument draws a higher current than the threshold defined for the 'idle' state, the system will not be able
to detect Flash (i.e. ON-Hook state).
Consider this when changing the value of ON-Hook Detection Current. Define the value considering the
current drawn by your SLT instrument in idle state, as well as when Flash key is pressed.
19. The longer the Loop Length of the SLT port, the greater the likelihood of current dissipation, affecting speech quality of the tele-
phone instrument connected to the SLT port. Change the Loop Current according to the Loop Length of the SLT instrument con-
nected to the port.
20. Dialing 'Flash' either with the 'Flash Key' or by pressing the Hook-switch causes the phone to go in ON-Hook state briefly for 600-
800 milliseconds. Thus ON-Hook state is simulated briefly. The SLT may draw a higher current when 'Flash' is dialed.
Select None if no Answer Signaling is to be generated on the SLT port. Default: None.
11. Select the Disconnect Signaling22 on the SLT port as Polarity Reversal or Open Look as appropriate.
Select None if Disconnect Signaling is not generated on the SLT port. Default: None.
• Polarity Reversal: Call Disconnection is signaled in the form of Polarity Reversal. The Battery polarity
of the SLT port will be reversed. For example, if the battery polarity of the SLT port is '+ve' for TIP and '-
ve' for RING in speech condition then on disconnection on other port, TIP will become '-ve' and Ring
'+ve'. When call is disconnected, user will get Error tone.
• Open Loop: Call Disconnection is signaled in the form of Open Loop Disconnect Pulse, whereby the
Battery voltage on the SLT port is removed for the duration of the Open Loop Disconnect Timer
configured for that SLT port and will be restored on the expiry of this Timer. However, the Polarity of
Battery Voltage on the SLT port is not changed. When call is disconnected, the SLT extension user
gets an Error tone.
12. Set the Open Loop Disconnect Timer (msec), if you selected Open Loop Disconnect as Disconnect
Signaling on the SLT port.
Open Loop Disconnect Timer is the time period for which the system will remove Battery Voltage on the
SLT port and restore Battery Voltage on the expiry of the Timer to signal Call Disconnection. The range of
this timer is: 001 to 999 milliseconds. Default: 500 msec.
13. To increase the volume of the outgoing speech on the SLT, set the Tx Gain. Default: 0dB.
14. To increase the volume of the incoming speech on the SLT, set the Rx Gain. Default: 0dB.
15. To increase incoming speech volume level of calls from SIP Trunks to the SLT extension, set Rx Gain at
SIP Trunk (dB). Default: -9dB.
16. To increase outgoing speech volume level of calls from SIP Trunks to the SLT extension, set Tx Gain at
SIP Trunk (dB). Default: 0dB.
21. Answer Signal is a signal generated on the SLT port to indicate that the called party (remote party) has answered the call and the
call is now mature. Answer Signaling on the SLT port is particularly useful when there is a PCO machine or any Billing equipment
connected to the SLT port. With Answer Signaling enabled on an SLT port, during an outgoing call is made from that SLT port to
any other port - CO/Mobile/SIP- when the called party (remote party) answers, the Public Network provides an Answer Signal to
the trunk port to indicate call maturity. This information can be passed on to the PCO machine billing equipment in the form of
Answer Signaling. On detecting Answer Signaling the PCO machine billing equipment can start billing.
Answer Signaling is generated in the form of Polarity Reversal or Battery Reversal, whereby the Battery polarity of the SLT port
gets reversed. For example, if the battery polarity of the SLT port is +ve for TIP and -ve for RING in speech condition, then on call
maturity, TIP becomes -ve and Ring becomes +ve.
22. A 'Disconnect Signal' is the signal generated on the SLT port to indicate that the called party (remote party has disconnected the
call.
Disconnect Signaling on the SLT port is useful when there is a PCO machine or any Billing equipment connected to the SLT port.
With Disconnect Signaling enabled on an SLT port, during an outgoing call is made from that SLT port to any other port - CO/
Mobile/SIP - when the called party (remote party) disconnects, i.e. goes ON Hook, the Public Network provides a Disconnect Sig-
nal to trunk port indicate call disconnection. This signal can be generated on the SLT port to indicate to the PCO machine/Billing
equipment connected to this port to consider the call as disconnected and stop billing. Thus, Disconnect Signaling on the SLT port
helps prevent excessive billing.
2. Select a Profile from 1 to 4 as Voice Mail Auto Attendant Profile for the extension. Default: 1.
The Voice Mail Auto Attendant Profile determines the welcome message to be played to mailbox owner
(extension user) when they reach the home node. It also determines whether or not the user should be
taken to the root node directly.
Make sure you also configure the Voice Mail Auto Attendant Profiles when you assign them to extensions.
To configure, click the link Voice Mail Auto Attendant Profile. See “Voice Mail Auto Attendant Profile”.
3. To allow callers to reach the SLT extension using Dial By Name option offered by the VMS Auto Attendant,
abbreviate the extension user’s name to three letters and type it in the Abbreviated Name field.
When the VMS Auto Attendant is used, callers can be prompted to Dial by Name of the desired party
(extension user) instead the extension number.
4. Select the Call Transfer Type for the Day and Call Transfer Type for Night from the following options:
• Transfer when extension answers: When the caller dials the extension number, the VMS Auto
Attendant will transfer the call when the extension answers (goes OFF-Hook).
If the extension does not answer23, the VMS Auto Attendant will transfer the call to the mailbox of the
extension, if assigned, or take the caller back to the home node.
• Take caller to Mailbox: When the caller dials the extension number, the VMS Auto Attendant will
check if the extension number has a mailbox assigned and transfer the call to the mailbox of the
extension.
• Transfer immediately: When the caller dials the extension number, the VMS Auto Attendant will
transfer the call on the extension without checking whether it is busy or free.
23. The VMS will wait for the duration of the Wait for Answer Timer (default: 15 seconds; the timer is configurable). If the call is not
answered before this timer expires, it is treated as No Reply.
If the extension is busy the VMS Auto Attendant will transfer the call to the mailbox of the extension, if
assigned, or take the caller back to the home node.
• Transfer when extension permits: The VMS Auto Attendant prompts the caller to record his/her
name. It puts the caller on hold and places the call on the desired extension. If the extension is free and
answers the call, the VMS announces the caller’s name to the extension user and prompts the
extension user to choose whether or not to speak to the caller. If the extension user chooses to talk, the
VMS transfers the call.
If the extension user chooses not to talk, the VMS transfers the call to the mailbox of the extension
user, if assigned, and asks the caller to leave a message.
5. Enable Announce Name, if you want the VMS Auto Attendant to announce the extension user’s name to
the caller when transferring the call to the extension. Default: Disabled.
When you enable Announce Name, make sure you record the extension user’s name on the VMS. For
instructions, see “Recording Station Names”.
DDI Routing
2. In the field DDI Number for SIP Trunk 1, enter the DDI number you want to assign to this SLT extension
from among the DDI numbers provided by the ITSP with whom you have registered SIP Trunk 1.
3. In the field DDI Number for SIP Trunk 2, enter the DDI number you want to assign to this SLT extension
from among the DDI numbers provided by the ITSP with whom you have registered SIP Trunk 2.
4. In the field DDI Number for SIP Trunk 3, enter the DDI number you want to assign to this SLT extension
from among the DDI numbers provided by the ITSP with whom you have registered SIP Trunk 3.
5. In the field DDI Number for SIP Trunk 4, enter the DDI number you want to assign to this SLT extension
from among the DDI numbers provided by the ITSP with whom you have registered SIP Trunk 4.
6. To allow the system to land incoming calls with the same DDI number on this SLT extension, keep Allow
Incoming DDI Calls enabled. Default: Enabled.
7. To allow the system to send the DDI number as CLI when outgoing calls are made from this SLT
extension, keep Send DDI Number as CLI enabled. Default: Enabled.
If you clear the check box, the system will send the MSN number as CLI of outgoing calls made from
this extension.
SMDR Storage
The ETERNITY offers Station Message Detail Recording (SMDR) that enables you to record the details of Internal,
Incoming (IC) and Outgoing (OG) calls made from/to all its extensions. To obtain SMDR as a report, you must
enable SMDR Storage, and set filters. See “Station Message Detail Recording (SMDR)” to know more.
1. Click [+] SMDR Storage to expand options.
2. Select the type of internal calls to be stored from the combo box Store Internal Calls. You can select from
the following options:
• Made by/made to this extension (the system will store all calls made to and from this extension).
• If made by this extension (the system will store outgoing calls made from this extension)
• If made to this extension (the system will store only incoming calls from other extensions)
• Never (the system will not store internal calls)
3. To store details of incoming calls from external numbers, select Store Incoming Calls. Default: selected.
4. To store details of outgoing calls made by the extension user to external numbers, select Store Outgoing
Calls. Default: selected.
Call Budget
The Call Budget feature allot a 'budget' limit for outgoing calls made by the extension. See “Call Budget” for more
information. If you want to enable this feature on this extension,
This parameter is related to the CLIP feature. It allows you to choose whether the system should display the CLI of
the ‘Held Party’ or the CLI of the ‘Transferring Party’ to the transfer destination extension while the call is being
transferred.
See the feature description for “Calling Line Identification and Presentation (CLIP)” to know more.
• Display Number of Party kept on Hold when call is transferred on this extension.
Walk-In/Walk-Out
This parameter is related to the feature Walk-In Class of Service. ETERNITY offers two types of Walk-In: i) One-
Call per Walk-In, where the extension user is automatically logged out after a call. ii) Walk-In until Logout, where
the extension user remains logged on until s/he manually walks out or a second user walks into the same
extension. To know more about this feature, see “Walk-In Class of Service”.
2. Select the radio button of the type of Walk-Out mode you want to assign to the extension:
• Walk-Out on completion of call automatically: Select this option, if you want to assign One-Call per
Walk-In to the extension.
Call Forward
2. Set the Call Forward No Reply Timer (sec) to the desired value, if required. The range of this timer is .
Default: 030 seconds.
Call Forward No Reply Timer signifies the duration for which the system will wait for an extension to
answer an incoming call, before forwarding the call it to the programmed destination phone number as Call
Forward-No Reply. By default the Timer is set to 30 seconds. Refer the feature description for “Call
Forward” to know more.
3. Select the type of calls to be forwarded If Call is forwarded on external number. You may select from the
following options:
• Forward only Internal calls
• Forward only Trunk calls
• Forward all calls (internal as well as trunk calls)
Default: Forward only Trunk calls.
This parameter is relevant for the features “Call Forward” and “Mobility Extension”.
More Features
1. Click More... link to expand.
3. Now, configure the Call Taping Profile number you selected for the extension. To do this,
• Click the link Call Taping. The page opens in a new window, displaying the parameters of the profile
you selected for the extension.
• Enable Call Taping for Outgoing Calls. Type the external numbers that you want the system to
tape. You may type as many as 99 numbers.
• Select the check box Tape Calls without CLI? if you want incoming calls without CLI to be taped.
Default: Disabled.
• Select the check box Call Taping for Internal Calls, if you want internal calls made and received
by the extension to be taped.
• Close the pop-up window by clicking [x] on the top-right corner of the window.
2. Select the type of Call Duration Control to be applied from the options:
• OFF
• Apply as per profile 1
• Apply as per profile 2
• Apply as per profile 3
• Apply as per profile 4
• Click the settings button. The page opens in a new window. The page displays the parameters of the
profile you selected for the extension.
• Enable Apply CDC to Internal Calls, if CDC is to be applied on internal calls. Default: Disabled.
• Enable Apply CDC for Incoming Calls received from trunks, if CDC is to be applied on incoming
external calls. Default: Disabled.
• Enable Apply CDC for Outgoing Calls, if CDC is to be applied to outgoing external calls. Default:
Disabled.
• If required, change the CDC Timer to the desired duration. The range of the timer is 0001 to 9999
seconds. Default: 160 seconds.
• Enable Disconnect Call after CDC Timer check box if you want calls to be disconnected on the expiry
of the CDC Timer. Default: Disabled.
• In the Apply CDC for calls matching with numbers column, type the external numbers on which you
want to apply CDC. You can enter as many as 99 numbers.
• In the Do Not Apply CDC for calls matching with numbers column, enter the numbers which you
want to be exempt from CDC.
Priority
1. Select a Priority from 1 to 9 for the SLT extension. Default: 5.
Each extension of the ETERNITY is assigned a Priority Level starting from 1, 2, 3... to 9, with '1' being
lowest Priority and '9' being highest Priority. Whenever an extension (phone) with higher priority calls an
extension with lower priority, a triple ring is placed on the called extension. To know more, read the feature
description “Priority”.
Personal Directory
1. Select a Personal Directory number from 01 to 50 that you want to assign to the SLT extension. Default:
None.
A Personal Directory is a list of 25 frequently dialed numbers, each of which are stored by Index number
(location code), Name and Trunk Access Codes (). The Personal Directory is necessary for using the
features “Abbreviated Dialing” and “Dial By Name”.
The Personal Directory can be configured also by the extension user. Refer the topic “Abbreviated Dialing” for
instructions on configuring the Personal Directory.
If you have not configured Time Table, you may do so now, by clicking the Time Table link. Define the
working days, and the start and end time of the working hours for each working day.
2. Select the type of alarm notification the system should use When User answers Alarm Call. Your options
are:
• Play Voice Message: The extension user is played a message recorded in the Voice Module on
answering the alarm call.
• Play Music-on-Hold: The extension user is played music-on-hold on answering the alarm call.
• Play VMS Alarm Greeting: The extension user is greeted by the VMS on answering the alarm call.
See
• Route to Operator: The alarm call is routed to the Operator extension, so that the Operator can the
serve the alarm request to the extension user.
Help Desk
Configure this parameter if you want to define the extension as a “Help Desk”.
2. Select the check box Help Desk Auto set ACB on calling extension, when this extension is busy.
Default: Disabled.
2. Select the check box Do not allow Outgoing Calls without Account Code (Force Account Code).
Now that you have finished configuring the SLT extension. You may configure the next SLT extension.
The Copy dialog box opens. The dialog box displays the number and name of the extension to be copied
in the Copy from: field.
The dialog box also lists the numbers and names of the extensions to which you can copy the values
under Copy To:
2. Select the check boxes of the extension numbers to which you want to Copy To.
The ETERNITY NE supports two Digital Key Phone (DKP) extension ports.
If you have enabled 'On-Site Configuration', only the DKP port on which a DKP/DSS Console is connected
displayed on this page for configuration.
On this page,
More: Click this button to view all parameter links on the page.
Less: Click this button to view only the essential parameter links on the page.
Expand: Click to expand a link to display all parameters under the link.
Collapse: Click to collapse a link. Hides all parameters under the link.
To configure the second DKP extension, click the DKP Extension number (name) tab.
To copy the same DKP parameter values to the other DKP, SLT and SIP extensions, use Copy button.
Class of Service
Define the Class of Service for the DKP extension for Day time and Night time.
Toll Control
1. Click [+] to expand Toll Control options. Set the desired Toll Control for the DKP extension for the Day
and Night.
2. Select the type of Calls Allowed during Day: All Calls, No Calls, Local Calls, Regional Calls, National
Calls, and Limited Calls 1,2, 3. Default: All Calls.
3. Select the type of Calls Allowed during Night: All Calls, No Calls, Local Calls, Regional Calls, National
Calls, and Limited Calls 1,2, 3. Default: All Calls.
The Toll Control levels on this page are based on the allowed and denied number lists of Local, Regional,
National, International, and Limited Call numbers you configured on the “Number Patterns” page.
For each Toll Control Level from 0 to 3, you must assign 'Call Privilege’24. For each Call Privilege, you need to
configure the corresponding number strings to be allowed and number strings to be denied. See “Dynamic Lock” to
know more about this feature.
2. Select the call privilege for Calls allowed for Lock Level 1: No Calls, Local Calls, Regional Calls, National
Calls, All Calls.
3. Select the call privilege for Calls allowed for Lock Level 2: No Calls, Local Calls, Regional Calls, National
Calls, All Calls.
4. Select the call privilege for Calls allowed for Lock Level 3: No Calls, Local Calls, Regional Calls, National
Calls, All Calls.
The Lock Levels on this page are based on the allowed and denied number lists of Local, Regional,
National, International, and Limited Call numbers you configured on the “Number Patterns” page.
5. If you have not configured the allowed and denied number list for the Calls allowed/denied for the selected
Lock Level, or if you want to add to the existing lists, you may do it now.
• Click the settings icon.
• The Number Patterns page will open in a new window.
• Configure the Allowed and Denied Numbers for Local, Regional, National, International, and Limited
Calls.
• Click Submit.
• Close the window after you have configured the list.
24. The Call Privilege types are: No Calls, Local Calls, Regional Calls, National Calls, International Calls and Limited Calls.
Outgoing calls (to external numbers) are made by dialing Trunk Access Codes (TAC).
• TAC for users worldwide are: 0, 5, 61, 62, 63, 64.
• TAC for users in USA are: 9, 5, 81, 82, 83, 84.
For each TAC, you need to select the Outgoing Trunks. All external calls made by dialing a particular TAC will be
routed through the outgoing trunks you selected for that TAC.
You can also apply Least Cost Routing logic on the selected trunks, so that ETERNITY routes the outgoing call
through the trunk that costs the lowest for the call.
2. Select Trunks allowed for ‘0’ dialing. The outgoing call will be routed through the selected trunks when
the extension user dials TAC ‘0’.
On the left, the trunks appear with their names (if configured in “Naming Trunks”) and port numbers in a
sequence, starting with CO trunks, followed by Mobile trunks and SIP trunks.
If you have not assigned any names to the trunks, they will appear with their default names (CO, MOB,
SIP) and port numbers.
If you have enabled On-Site Configuration, only those trunks that are connected will appear in the box.
• To select a trunk, place your cursor on the desired trunk, and click the Select>> button.
Or
• Press the ctrl key and click the left mouse button to select multiple trunks.
• You may change the sequence of the trunks you selected, if required, using the Up and Down arrow
buttons on the right display box.
• You can also delete trunks from the ones you have selected.
• You may enable Rotation, if you have selected more than one trunk. Default: Disabled.
When you enable Rotation, each new outgoing call is routed through the subsequent trunk in the
group25. This ensures equal distribution of outgoing call traffic on all trunks.
25. The first call through the first trunk, the second through the second, the third through the third trunk, and so forth. Thus each new
call is routed through the trunk next to the one that routed the previous outgoing call.
All the trunks appear in the field Trunks allowed for ‘0’ dialing, in the sequence you selected, separated
by commas. For example, BSNL (CO-1), BSNL (CO-3), Reliance (CO-4) and Pulver (SIP1).
3. To apply Least Cost Routing on the Trunks allowed for 0 dialing, select the desired LCR method from the
combo box:
• Number Based: Choose this option if the service providers of the trunks you selected offer different
tariffs according to area or distance, or phone numbers dialed.
• Time Based: Choose this option if the service providers of the selected trunks offer a different tariff
according to the time of the day.
• Time + Number Based: Choose option if the service providers of the selected trunks offer different
tariffs according to the time of the day as well as area/distance/phone number.
• Service Provider Based: Choose this option if the same service providers of the selected trunks offer
different rates for calls made to numbers within their own network and for calls made to numbers of
another Service Provider's network.
• Configure LCR method that you selected for the trunk group.
4. To configure Least Cost Routing method you selected, click the link LCR settings. A new window opens.
• Go to the LCR method you selected for Trunks allowed for 0 dialing.
• Configure the LCR method. See Least Cost Routing under Advanced Settings for instructions.
• Click Submit.
• Close the window.
5. Select Trunks allowed for ‘5’ dialing. The outgoing call will be routed through the selected trunks when
the extension user dials TAC ‘5’. Follow the same steps as described above.
• Double-click the field. A multiple selection box opens.
• Select trunks, placing your cursor on the desired trunk, and clicking the Select>> button.
• Change the sequence of the trunks you selected, if required, using the Up and Down arrow buttons on
the right display box. Delete trunks from the ones you have selected, if required.
• Enable Rotation, if you have selected more than one trunk. Default: Disabled.
• Click OK.
All the trunks appear in the field Trunks allowed for ‘5’ dialing, in the sequence you selected, separated
by commas.
• Choose a Least Cost Routing method, if you want to apply it on the trunks. Default: OFF.
• Configure the Least Cost Routing method you selected by clicking LCR Settings link.
6. Select Trunks allowed for ‘61’ dialing. The outgoing call will be routed through the selected trunks when
the extension user dials TAC ‘61’.
7. Select Trunks allowed for ‘62’ dialing. The outgoing call will be routed through the selected trunks when
the extension user dials TAC ‘62’.
• Follow the same steps as described above to select the trunks, enable Rotation, and apply Least Cost
Routing.
8. Select Trunks allowed for ‘63’ dialing. The outgoing call will be routed through the selected trunks when
the extension user dials TAC ‘63’.
• Follow the same steps as described above to select the trunks, enable Rotation, and apply Least Cost
Routing.
9. Select Trunks allowed for ‘64’ dialing. The outgoing call will be routed through the selected trunks when
the extension user dials TAC ‘64’.
• Follow the same steps as described above to select the trunks, enable Rotation, and apply Least Cost
Routing.
Phone Settings
2. Set the number of Call Appearances for the DKP extension. Default: 2
Call Appearances (also referred to as 'call loops') define the capacity of a DKP extension to handle
multiple calls simultaneously. A Call Appearance allows an extension user to attend to more than one
calling party at a time.
A minimum of two Call Appearances must be assigned to a DKP extension - Operator extension or
Executive extension - so that the extension user can put one party on hold while talking to another. A third
Call Appearance allows the extension user to put two calls on hold, make/attend a third call and toggle
between three calls.
DKP extensions for Executives are usually assigned 2 Call Appearances, while the Operator extension is
assigned 6 Call Appearances to handle 6 calls simultaneously.
3. Select the CO CLIP Pattern for the DKP. This is the type of Calling Line Presentation on the DKP for
incoming calls from trunks. You can select any of these options:
• Name Only: only the name of the caller will be displayed.
• Number Only: only the number of the caller will be displayed.
• Number + Name: both the name and the number of the caller will be displayed.
Default: Number + Name.
ETERNITY NE has language support for English, French, German, Spanish, Portuguese, and Italian.
When you select any of these languages, all the prompts and command strings will appear in the selected
language.
5. Select a Ringer Mode for the DKP from the four options:
• Ring immediate (it rings immediately as a fresh calls lands on the DKP).
• Ring if idle (rings only if the DKP is idle).
• Ring after a delay (if the call is still not answered).
• Ring OFF (silent mode).
6. If you selected Ring after a delay as Ringer Mode, set the Ring Delay Timer (sec), if required, to the
desired value.
The Ring Delay Timer is the time in seconds the ETERNITY NE waits on receiving a call to ring on the
DKP. The range of this timer is 0 to 99 seconds. Default: 10 seconds.
7. If you want to enable Ringer Auto Acknowledge mode, set the Acknowledge Timer (sec) to the desired
value.
The Ringer Auto Acknowledge mode determines when to stop the ring on the DKP. There are two options
for Ringer Auto Acknowledge:
• Stop only when the call is answered.
• Stop after a delay.
To stop the ring on the DKP after a delay, the Acknowledge Timer must be configured. The range of this
timer is 01 to 99 seconds. Default: 00 seconds.
To stop the ring only when the Call is answered or manually acknowledged, the Acknowledge Timer must
be set to '00'. By default, Ring Auto Acknowledge is turned OFF.
8. To assign the Ring Destination for the DKP extension, select the desired destination for Play Ring on. You
may choose
• Speakerphone: The ring will be played on the Speakerphone.
• Headset: The ring will be played on the Headset.
When you select the Headset as the destination, make sure that you enable Connect Headset to the DKP.
The speech path of both, the Headset and the Handset is common. If the Headset is not connected and
you have selected the Headset as the ring destination, the ring will be played on the speaker of the
Handset.
9. Select the desired Ring Tune according to your/DKP user’s preference. Default: 1.
10. Set the Ringer Volume to the desired level according to your preference. Default: 5.
11. To increase/decrease the volume of outgoing speech (Transmit Gain) on the handset of the DKP, set the
Handset Transmit Volume to the desired level, from 0 to 9. Default: 5.
12. To increase/decrease the volume of incoming speech (Receive Gain) on the handset of the DKP, set the
Handset Receive Volume to the desired level, from 0 to 9. Default: 5.
13. To increase/decrease the volume of outgoing speech (Transmit Gain) on the headset port of the DKP, set
the Headset Transmit Volume to the desired level, from 0 to 9. Default: 5.
14. To increase/decrease the volume of outgoing speech (Receive Gain) on the headset port of the DKP, set
the Headset Receive Volume to the desired level, from 0 to 9. Default: 5.
15. To change the Transmit Gain of the Speakerphone MIC Volume, set Hands-free Transmit Volume to the
desired level, from 0 to 9. Default: 5.
16. To change the Receive Gain of the Speakerphone MIC Volume, set Hands-free Receive Volume to the
desired level, from 0 to 9. Default: 5.
17. To change the key DTMF Side Tone of the DKP, set the Key Click Volume to the desired level, from 0 to
9. Default: 5
Key Click Volume is the tone you hear as you press the dial pad keys of EON.
18. To enable DTMF dialing on the DKP, keep the DTMF Generation flag enabled. Default: Enabled.
19. Select the desired DTMF Transmit Level from 0 to 9 for DTMF generation from the DKP. Default: 5.
20. To use a Headset with the DKP, set Headset Connected? to Yes. Default: No.
Make sure that you connect a Headset to the DKP, if you select Yes.
21. Select the check box of Auto Answer to enable this feature on the DKP. Default: Disabled.
When you set the “Auto Answer” feature on the DKP, the DKP goes OFF-Hook automatically after a preset
period of time, without the extension user having to pick up the handset or press the speaker or headset
key. When you enable Auto Answer, you must configure the Auto Answer Timer.
22. If you enabled Auto Answer on the DKP, set the Auto Answer Timer (sec) to the desired value.
If you set the Auto Answer Timer to 0, Auto Answer feature will be disabled.
23. Change, if required, the Backlight brightness of the DKP LCD display, by setting the LCD Backlight Level
to the desired value.
The intensity of the backlight brightness increases from 0 to 4, where '0' will cause the backlight to be
turned OFF. '1' signifies minimum intensity, '4' signifies maximum intensity. Select a level from 1 to 4 from
the list in the combo box. Default: 3.
24. Set the Back Light Off Timer (sec) to the desired value, if required.
The backlight of the DKP LCD display can be kept switched ON continuously, or can be set to switch Off
automatically after a predefined period of time, by setting the Backlight Off Timer. The range of the
Backlight Off Timer is from 000 to 999 seconds. Default: 60 seconds.
25. Set the LCD Contrast Level to a level from 1 to 4 that is comfortable to you.
The DKP has 4-level contrast control for its LCD display. Level 1 signifies minimum and level 4 is the
maximum. Default: 3.
In addition to the above listed Phone Settings configured by the Installer, DKP extension users can also change
their phone settings, referred to as ‘DKP Personal Settings’ to match their preferences and requirement. The DKP
Personal Settings include:
• Ringer Volume
• Ringer Tune
• Ringer Mode
• Ringer Acknowledge Mode
• Speech Volume (Transmit/Receive)
• Time Zone (Working Hours/Non-working Hours)
• User Status (Present/Absent)
• Keypad Security (Lock/Open)
• Call Answer Type - Manual/Auto
• Headset/Handset Connectivity option
To be able change the DKP personal settings, the DKP extension user must access and navigate the phone menu.
See “Digital Key Phone-Operation” to know more.
2. Now, let us attempt to configure the feature Trunk to Trunk Transfer on the key that has presently CO3
configured on it.
3. Click the CO3 key. A dialog box opens, with the options for the Functions to be Performed by the key
4. In the dialog box, in the Select Function Type list box, select the function to be performed by the key.
5. In the Select Offset drop down list, all the features that can be assigned to DSS keys are listed.
6. Select Trunk to Trunk Call Transfer from the list of features in the Select Offset box.
12. Follow the same instructions to assign features to other DSS keys. Selecting the appropriate Function
Type and the Offset for each feature/function.
To assign direct access to DKP2, select DKP as Function Type and 2 as Offset.
To assign direct access to Mobile Trunk 1, select MOBILE as Function Type and 1 as Offset.
To assign direct access to SIP Trunk 1, select SIP as Function Type and 1 as Offset.
Repeat the same to assign direct access to Mobile Trunk 2 and SIP Trunk 2.
• Click OK, each time you select a Function Type and Offset in the dialog box.
• You can reinstate default key assignment any time, by clicking the Default button at the bottom of the
window.
The default key map of the DSS64 connected to the DKP port appears on your screen.
2. Click the DSS key. A dialog box opens, with the options for the Functions to be Performed by the key
3. In the Select Function Type list box, select the function to be performed by the key. For instance, if you
want to use this key to call a Department Group, select the option Dept. Group as Function Type.
You can reinstate default key assignment any time, by clicking the Default button at the bottom of the
window.
7. When you finish configuring the DSS keys, close the DSS Keys Settings window.
3. Keep the option System should ask Password while Accessing Mail Box enabled, if you want
password protection for the mailbox.
By default, access to the mailbox is password protected. The “User Password” is required to access the
mailbox. Whenever the mailbox owner accesses the mailbox, the VMS will ask for the (user) password.
Since a Mailbox can be accessed using the default User Password, 1111, extension users who are
assigned a mailbox are recommended to change their User Password to a unique 4 digit number to
prevent unauthorized access to their mailbox.
4. Define the Mail Box Size (minutes) for the extension. You may change the mailbox size to any desired
value from 001 to 999 minutes. Default: 5 minutes.
5. Define the Maximum Message Length (sec) for the extension. This is the length of each message that
callers are allowed to record in the mailbox of the extension. You can change the message length to any
desired value from 001 to 999 seconds. Default: 15 seconds.
The VMS stops recording the message of the callers if it exceeds the maximum message length, and
stores only that part of the message that was recorded within the maximum message length limit.
6. Select the option for delivery of new messages When Mail Box is Full. You may select any of the
following message delivery options:
• Deliver New Message in General mailbox: The VMS will not record the message in the General
mailbox. A General mailbox is a shared mailbox between extension users.
Only extension users who have General Mailbox in their “Class of Service (COS)” are allowed to
access it.
When you select this option, make sure that General Mailbox is enabled in the Class of Service of
the DKP extension. Refer “Class of Service (COS)” for instructions.
• Overwrite Old messages: The VMS will overwrite the old messages to record the new message in
the mailbox. The VMS will overwrite the oldest message first.
7. To play to the DKP extension user (mailbox owner), message details such as Date and Time (when the
message was recorded by the caller), the caller’s number26, the extension number dialed by the caller27,
select Play Message Detail after Delivery of Message.
You may select from the following options for Play message details:
• Never: The VMS will not play message details to the extension user after playing the message.
• Always: The VMS will play message details to the extension user after playing each message.
• On Demand: The VMS will play message details to the extension user only when the user requests
it. On completion of each message, the VMS will prompt the extension user to press a digit for date
and time stamp. When the mailbox owner presses the digit, the VMS will play the message details.
Default: On Demand.
8. Select the Voice Mail/Message Wait Notification Type for the extension. The VMS will notify the
extension user of the new messages in the mailbox and the message wait set by another extension user.
• Stuttered Dial Tone/Voice Message: When the extension user goes OFF-Hook, s/he will hear a
voice message, if a pre-recorded Voice Module has been assigned for Message Wait Notification. If
no voice module is recorded and assigned, the extension user will hear a stuttered dial tone
instead.
If you want voice message to be played as message wait notification, record and assign a Voice
Module. See “Voice Message Applications” for instructions.
ETERNITY can play only 4 Voice Modules simultaneously. The Voice Module for Message Wait Notification
will not be played if there are already 4 being played simultaneously. In which case, Stuttered Dial Tone will
be played for Message Wait Notification, when the extension user goes OFF-Hook.
• LED Lamp: If the DKP has a 'Message Wait' lamp, you may select this option as Message Wait
Indication. The lamp will blink continuously and will be turned off when the extension user has
retrieved all the waiting messages.
26. The number of person who left the message in the mailbox.
27. The number of the extension user for whom the message is intended.
When the extension user answers the call, the VMS informs the user of the new message and
allows the extension user to access it.
9. To notify the extension user via e-mail about new messages in the mailbox, select the check box Voice
Mail Notification via E-mail.
• To mail the new voice message to the extension user as attachment, select With Attachment (.wav
file).
• To mail the notification only to the extension user, select Without Attachment.
• Type the E-mail Address of the extension user to which the voice mail notification is to be sent. The e-
mail ID may consist of 64 characters (maximum). Default: blank.
10. To assign Department Group mailbox to the DKP extension, select the check box Assign Mail Box of
Department Group. Default: Disabled.
You can assign the Mailbox of a Department Group to DKP extensions, even to those DKP extensions that
are not included in the Department Group. See “Department Call”.
• Select the Department Group whose mailbox you want to assign to the extension from the combo box
Specify Department Group.
2. Select a Profile from 1 to 4 as Voice Mail Auto Attendant Profile for the extension. Default: 1.
The Voice Mail Auto profile determines the welcome message to be played to mailbox owner (extension
user) when they reach the home node. It also determines whether or not the user should be taken to the
root node directly.
3. To allow callers to reach the DKP extension using Dial By Name option offered by the VMS Auto
Attendant, abbreviate the DKP extension user’s name to three letters and type it in the Abbreviated Name
field.
When the VMS Auto Attendant is used, callers can be prompted to Dial by Name of the desired party
(extension user) instead the extension number.
4. Select the Call Transfer Type for the Day and Call Transfer Type for Night from the following options:
• Transfer when extension answers: When the caller dials the extension number, the VMS Auto
Attendant will transfer the call when the extension answers (goes OFF-Hook).
If the extension does not answer28, the VMS Auto Attendant will transfer the call to the mailbox of the
extension, if assigned, or take the caller back to the home node.
• Take caller to Mailbox: When the caller dials the extension number, the VMS Auto Attendant will
check if the extension number has a mailbox assigned and transfer the call to the mailbox of the
extension.
• Transfer immediately: When the caller dials the extension number, the VMS Auto Attendant will
transfer the call on the extension without checking whether it is busy or free.
• Transfer when extension rings: When the caller dials the extension number, the VMS Auto Attendant
will wait for the extension to start ringing and then transfer the call.
If the extension is busy the VMS Auto Attendant will transfer the call to the mailbox of the extension, if
assigned, or take the caller back to the home node.
• Transfer when extension permits: The VMS Auto Attendant prompts the caller to record his/her
name. It puts the caller on hold and places the call on the desired extension. If the extension is free and
answers the call, the VMS announces the caller’s name to the extension user and prompts the
extension user to choose whether or not to speak to the caller. If the extension user chooses to talk, the
VMS transfers the call.
If the extension user chooses not to talk, the VMS transfers the call to the mailbox of the extension
user, if assigned, and asks the caller to leave a message.
5. Enable Announce Name, if you want the VMS Auto Attendant to announce the extension user’s name to
the caller when transferring the call to the extension. Default: Disabled.
When you enable Announce Name, make sure you record the extension user’s name on the VMS. For
instructions, see “Recording Station Names”.
28. The VMS will wait for the duration of the Wait for Answer Timer (default: 15 seconds; the timer is configurable). If the call is not
answered before this timer expires, it is treated as No Reply.
2. In the field DDI Number for SIP Trunk 1, enter the DDI number you want to assign to this DKP extension
from among the DDI numbers provided by the ITSP with whom you have registered SIP Trunk 1.
3. In the field DDI Number for SIP Trunk 2, enter the DDI number you want to assign to this DKP extension
from among the DDI numbers provided by the ITSP with whom you have registered SIP Trunk 2.
4. In the field DDI Number for SIP Trunk 3, enter the DDI number you want to assign to this DKP extension
from among the DDI numbers provided by the ITSP with whom you have registered SIP Trunk 3.
5. In the field DDI Number for SIP Trunk 4, enter the DDI number you want to assign to this DKP extension
from among the DDI numbers provided by the ITSP with whom you have registered SIP Trunk 4.
6. To allow the system to land incoming calls with the same DDI number on this DKP extension, keep Allow
Incoming DDI Calls enabled. Default: Enabled.
7. To allow the system to send the DDI number as CLI when outgoing calls are made from this DKP
extension, keep Send DDI Number as CLI enabled. Default: Enabled.
If you clear the check box, the system will send the MSN number as CLI of outgoing calls made from this
extension.
SMDR Storage
The Station Message Detail Recording (SMDR) feature of ETERNITY enables you to record the details of Internal,
Incoming (IC) and Outgoing (OG) calls made from/to all its extensions. To obtain SMDR as a report, you must
enable SMDR Storage, and set filters. See “Station Message Detail Recording (SMDR)” to know more.
2. Select the type of internal calls to be stored from the combo box Store Internal Calls. You can select from
the following options:
• Made by/made to this extension (the system will store all calls made to and from this extension).
• If made by this extension (the system will store outgoing calls made from this extension)
• If made to this extension (the system will store only incoming calls from other extensions)
• Never (the system will not store internal calls)
3. To store details of incoming calls from external numbers, select Store Incoming Calls. Default: selected.
4. To store details of outgoing calls made by the extension user to external numbers, select Store Outgoing
Calls. Default: selected.
Call Budget
The Call Budget feature allot a 'budget' limit for outgoing calls made by the extension. See “Call Budget” for more
information. If you want to enable this feature on this extension,
3. Define the calling permission for the extension when it has used up the call budget allotted to it. Select the
desired option for Call Privilege when Call Budget is consumed:
• No Calls
• Local Calls
• Regional Calls
• National Calls
• All Calls
Default: No Calls.
This parameter is related to the CLIP feature. It allows you to choose whether the system should display the CLI of
the ‘Held Party’ or the CLI of the ‘Transferring Party’ to the transfer destination extension while the call is being
transferred.
See the feature description for “Calling Line Identification and Presentation (CLIP)” to know more.
• Display Number of Party kept on Hold when call is transferred on this extension.
Walk-In/Walk-Out
This parameter is related to the feature Walk-In Class of Service. ETERNITY offers two types of Walk-In: i) One-
Call per Walk-In, where the extension user is automatically logged out after a call. ii) Walk-In until Logout, where
the extension user remains logged on until s/he manually walks out or a second user walks into the same
extension. To know more about this feature, see “Walk-In Class of Service”.
2. Select the radio button of the type of Walk-Out mode you want to assign to the extension:
• Walk-Out on completion of call automatically: Select this option, if you want to assign One-Call per
Walk-In to the extension.
• Walk-Out when user gives command to Walk-Out: Select this option, if you want to assign Walk-In
until Logout to the extension.
Call Forward
2. Set the Call Forward No Reply Timer (sec) to the desired value, if required. The range of this timer is.
Default: 030 seconds.
Call Forward No Reply Timer signifies the duration for which the system will wait for an extension to
answer an incoming call, before forwarding the call it to the programmed destination phone number as Call
Forward-No Reply. By default the Timer is set to 30 seconds. Refer the feature description for “Call
Forward” to know more.
This parameter is relevant for the features “Call Forward” and “Mobility Extension”.
More Features
Call Taping
To use the “Call Taping” feature on the extension,
3. Now, configure the Call Taping Profile number you selected for the extension. To do this,
• Click the link Call Taping. The page opens in a new window, displaying the parameters of the profile
you selected for the extension.
• Select the check box Tape Calls without CLI? if you want incoming calls without CLI to be taped.
Default: Disabled.
• Select the check box Call Taping for Internal Calls, if you want internal calls made and received
by the extension to be taped.
• Close the window by clicking [x] on the top-right corner of the window.
2. Select the type of Call Duration Control to be applied from the options:
• OFF
• Apply as per profile 1
• Apply as per profile 2
• Apply as per profile 3
• Apply as per profile 4
3. Configure the Call Duration Control Profile number you selected for the extension. To do this,
• Click the settings button. The page opens in a new window. The page displays the parameters of the
profile you selected for the extension.
• Enable Apply CDC to Internal Calls, if CDC is to be applied on internal calls. Default: Disabled.
• Enable Apply CDC for Incoming Calls received from trunks, if CDC is to be applied on incoming
external calls. Default: Disabled.
• Enable Apply CDC for Outgoing Calls, if CDC is to be applied to outgoing external calls. Default:
Disabled.
• If required, change the CDC Timer to the desired duration. The range of the timer is 0001 to 9999
seconds. Default: 160 seconds.
• Enable Disconnect Call after CDC Timer check box if you want calls to be disconnected on the expiry
of the CDC Timer. Default: Disabled.
• In the Apply CDC for calls matching with numbers column, type the external numbers on which you
want to apply CDC. You can enter as many as 99 numbers.
Priority
Each extension of the ETERNITY is assigned a Priority Level starting from 1, 2, 3... to 9, with '1' being lowest
Priority and '9' being highest Priority. Whenever an extension (phone) with higher priority calls an extension with
lower priority, a triple ring is placed on the called extension. To know more, read the feature description “Priority”.
1. Select a Priority from 1 to 9 for the DKP extension. Default: 5.
If this DKP extension is assigned to Operator, you may want to set a higher priority for this extension.
Personal Directory
1. Select a Personal Directory number from 01 to 50 that you want to assign to the DKP extension. Default:
None.
A Personal Directory is a list of 25 frequently dialed numbers, each of which are stored by Index number
(location code), Name and Trunk Access Codes (). The Personal Directory is necessary for using the
features “Abbreviated Dialing” and “Dial By Name”.
When a Personal Directory is assigned to an DKP, it must also be configured. The Personal Directory can be
configured also by the extension user. Refer the topic “Abbreviated Dialing” for instructions on configuring the
Personal Directory.
Time Table
2. Select a Time Table for the DKP extension. Default: System Time Table.
2. Select the type of alarm notification the system should use When User answers Alarm Call. Your options
are:
• Play Voice Message: The extension user is played a message recorded in the Voice Module on
answering the alarm call.
• Play Music-on-Hold: The extension user is played music-on-hold on answering the alarm call.
• Play VMS Alarm Greeting: The extension user is greeted by the VMS on answering the alarm call.
See
Help Desk
Configure this parameter if you want to define the extension as a “Help Desk”.
2. Select the check box Help Desk Auto set ACB on calling extension, when this extension is busy.
Default: Disabled.
2. Select the check box Do not allow Outgoing Calls without Account Code (Force Account Code).
Now that you have finished configuring the DKP extension. You may configure the next DKP extension.
The Copy dialog box opens. The dialog box displays the name of the extension and number to be copied
in the Copy from: field.
The dialog box also lists the names and numbers of the extensions to which you can copy the values
under Copy To:
3. Click OK.
The parameter values of the extension you Copy From will be applied on the extension(s) you Copy To.
SIP Extensions function like any normal DKP/SLT extension of the ETERNITY, allowing you to make and receive
calls to any extension user of the ETERNITY and to external numbers over PSTN, GSM, and VoIP lines, depending
on the “Logical Partition” configured in the system.
SIP Extensions are a licensed feature. To know more, refer the topic “License Management”.
• The Extended IP Phone for ETERNITY NE supplied by Matrix. The Matrix Extended IP Phone functions
just like a DKP.
• Any standard SIP phone or SIP enabled device, such as an IP phone, a Soft phone, an Analog phone
adapter.
You can register SIP-enabled devices at three different locations as a single SIP Extension for Call
Forking.
The number of SIP extensions you have specified on the “Pre-requisites” page will be displayed on this page.
On this page,
Less: Click this button to view only the essential parameter links on the page.
Expand: Click to expand a link to display all parameters under the link.
Collapse: Click to collapse a link. Hides all parameters under the link.
To configure another SIP extension, click the SIP Extension number (name) tab.
To copy the same SIP extension parameter values to the other SIP or DKP and SLT extensions, use Copy
button.
The SIP extension numbers and the names (either default or the names you configured) appear on the tabs on this
page, starting with SIP extension 303 to 318.
If this flag is disabled, you will not be able to use this extension. You may clear the check box, when you
want to deactivate the extension.
SIP ID is necessary for registering the SIP Extension with the Registrar of the VoIP module. It is the
number with which you can call the SIP Extension. Any extension user of the ETERNITY NE can call a SIP
Extension by dialing the SIP ID assigned to the SIP extension. SIP ID of each SIP Extension must be a
unique number string of a maximum of 6 digits. Any combination of digits from 0 to 9 and the characters *
and # are allowed.
By default, SIP Extensions are assigned the following SIP ID (Access Codes):
01 303
02 304
: :
16 318
Authentication Password is the password to be used by the VoIP module’s Registrar Server to
authenticate the SIP messages received from this SIP Extension. You can enter a maximum of 24 digits as
password. The valid digits for the password are 0 to 9, * and #.
Authentication ID is applicable only if any of the SIP message 'Authentication' options, REGISTER or
INVITE or SUBSCRIBE or PUBLISH, is enabled. Else, Authentication ID will not be used.
Class of Service
Define the Class of Service for the SIP extension for Day time and Night time.
Toll Control
2. Select the type of Calls Allowed during Day: All Calls, No Calls, Local Calls, Regional Calls, National
Calls, and Limited Calls 1, 2, 3. Default: All Calls.
3. Select the type of Calls Allowed during Night: All Calls, No Calls, Local Calls, Regional Calls, National
Calls, and Limited Calls 1, 2, 3. Default: All Calls.
The Toll Control levels on this page are based on the allowed and denied number lists of Local, Regional,
National, International, and Limited Call numbers you configured on the “Number Patterns” page.
4. If you have not configured the allowed and denied number list for the Type of Calls you selected as Toll
Control, or if you want to add to the existing list, you may do it now.
• Click the settings icon.
• The Number Patterns page will open in a new window.
• Configure the Allowed and Denied Numbers.
• Click Submit.
• Close the window after you have configured the list.
For each Toll Control Level from 0 to 3, you must assign 'Call Privilege’29. For each Call Privilege, you need to
configure the corresponding number strings to be allowed and number strings to be denied. See “Dynamic Lock” to
know more about this feature.
2. Select the call privilege for Calls allowed for Lock Level 1: No Calls, Local Calls, Regional Calls, National
Calls, All Calls.
3. Select the call privilege for Calls allowed for Lock Level 2: No Calls, Local Calls, Regional Calls, National
Calls, All Calls.
4. Select the call privilege for Calls allowed for Lock Level 3: No Calls, Local Calls, Regional Calls, National
Calls, All Calls.
The Lock Levels on this page are based on the allowed and denied number lists of Local, Regional,
National, International, and Limited Call numbers you configured on the Number Patterns page.
5. If you have not configured the allowed and denied number list for the Calls allowed/denied for the selected
Lock Level, or if you want to add to the existing lists, you may do it now.
• Click the settings icon.
• The Number Patterns page will open in a new window.
29. The Call Privilege types are: No Calls, Local Calls, Regional Calls, National Calls, International Calls and Limited Calls.
Outgoing calls (to external numbers) are made by dialing Trunk Access Codes (TAC).
• TAC for users worldwide are: 0, 5, 61, 62, 63, 64.
• TAC for users in USA are: 9, 5, 81, 82, 83, 84.
For each TAC, you need to select the Outgoing Trunks. All external calls made by dialing a particular TAC will be
routed through the outgoing trunks you selected for that TAC.
You can also apply Least Cost Routing logic on the selected trunks, so that ETERNITY routes the outgoing call
through the trunk that costs the lowest for the call.
2. Select Trunks allowed for ‘0’ dialing. The outgoing call will be routed through the selected trunks when
the extension user dials TAC ‘0’.
If you have not assigned any names to the trunks, they will appear with their default names (CO, MOB,
SIP) and port numbers.
If you have enabled On-Site Configuration, only those trunks that are connected will appear in the box.
• To select a trunk, place your cursor on the desired trunk, and click the Select>> button.
Or
• Press the ctrl key and click the left mouse button to select multiple trunks.
• You may change the sequence of the trunks you selected, if required, using the Up and Down arrow
buttons on the right display box.
• You can also delete trunks from the ones you have selected.
• You may enable Rotation, if you have selected more than one trunk. Default: Disabled.
When Rotation is OFF, calls are routed through the first trunk in the group. If this trunk is busy, the call
is routed to the next trunk in the group.
All the trunks appear in the field Trunks allowed for ‘0’ dialing, in the sequence you selected, separated
by commas. For example, BSNL (CO-1), BSNL (CO-3), Reliance (CO-4) and Pulver (SIP1).
3. To apply Least Cost Routing on the Trunks allowed for 0 dialing, select the desired LCR method from the
combo box:
• Number Based: Choose this option if the service providers of the trunks you selected offer different
tariffs according to area or distance, or phone numbers dialed.
• Time Based: Choose this option if the service providers of the selected trunks offer a different tariff
according to the time of the day.
• Time + Number Based: Choose option if the service providers of the selected trunks offer different
tariffs according to the time of the day as well as area/distance/phone number.
• Service Provider Based: Choose this option if the same service providers of the selected trunks offer
different rates for calls made to numbers within their own network and for calls made to numbers of
another Service Provider's network.
• Configure LCR method that you selected for the trunk group.
4. To configure Least Cost Routing method you selected, click the link LCR settings. A new window opens.
• Go to the LCR method you selected for Trunks allowed for 0 dialing.
• Configure the LCR method. See Least Cost Routing under Advanced Settings for instructions.
• Click Submit.
• Close the window.
5. Select Trunks allowed for ‘5’ dialing. The outgoing call will be routed through the selected trunks when
the extension user dials TAC ‘5’. Follow the same steps as described above.
• Double-click the field. A multiple selection box opens.
• Select trunks, placing your cursor on the desired trunk, and clicking the Select>> button.
• Change the sequence of the trunks you selected, if required, using the Up and Down arrow buttons on
the right display box. Delete trunks from the ones you have selected, if required.
• Enable Rotation, if you have selected more than one trunk. Default: Disabled.
• Click OK.
All the trunks appear in the field Trunks allowed for ‘5’ dialing, in the sequence you selected, separated
by commas.
• Choose a Least Cost Routing method, if you want to apply it on the trunks. Default: OFF.
• Configure the Least Cost Routing method you selected by clicking LCR Settings link.
30. The first call through the first trunk, the second through the second, the third through the third trunk, and so forth. Thus each new
call is routed through the trunk next to the one that routed the previous outgoing call.
7. Select Trunks allowed for ‘62’ dialing. The outgoing call will be routed through the selected trunks when
the extension user dials TAC ‘62’.
• Follow the same steps as described above to select the trunks, enable Rotation, and apply Least Cost
Routing.
8. Select Trunks allowed for ‘63’ dialing. The outgoing call will be routed through the selected trunks when
the extension user dials TAC ‘63’.
• Follow the same steps as described above to select the trunks, enable Rotation, and apply Least Cost
Routing.
9. Select Trunks allowed for ‘64’ dialing. The outgoing call will be routed through the selected trunks when
the extension user dials TAC ‘64’.
• Follow the same steps as described above to select the trunks, enable Rotation, and apply Least Cost
Routing.
3. Keep the option System should ask Password while Accessing Mail Box enabled, if you want
password protection for the mailbox.
By default, access to the mailbox is password protected. The “User Password” is required to access the
mailbox. Whenever the mailbox owner accesses the mailbox, the VMS will ask for the (user) password.
4. Define the Mail Box Size (minutes) for the extension. You may change the mailbox size to any desired
value from 001 to 999 minutes. Default: 5 minutes.
5. Define the Maximum Message Length (sec) for the extension. This is the length of each message that
callers are allowed to record in the mailbox of the extension. You can change the message length to any
desired value from 001 to 999 seconds. Default: 15 seconds.
The VMS stops recording the message of the callers if it exceeds the maximum message length, and
stores only that part of the message that was recorded within the maximum message length limit.
6. Select the option for delivery of new messages When Mail Box is Full. You may select any of the
following message delivery options:
• Do not offer to Record a Message: The VMS will not offer the caller the option of Leaving a Message.
• Deliver New Message in General mailbox: The VMS will not record the message in the General
mailbox. A General mailbox is a shared mailbox between extension users.
Only extension users who have General Mailbox in their “Class of Service (COS)” are allowed to
access it.
When you select this option, make sure that General Mailbox is enabled in the Class of Service of the
SIP extension. Refer “Class of Service (COS)” for instructions.
• Overwrite Old messages: The VMS will overwrite the old messages to record the new message in the
mailbox. The VMS will overwrite the oldest message first.
7. To play to the SIP extension user (mailbox owner), message details such as Date and Time (when the
message was recorded by the caller), the caller’s number31, the extension number dialed by the caller32,
select Play Message Detail after Delivery of Message.
You may select from the following options for Play message details:
• Never: The VMS will not play message details to the extension user after playing the message.
• Always: The VMS will play message details to the extension user after playing each message.
• On Demand: The VMS will play message details to the extension user only when the user requests
it. On completion of each message, the VMS will prompt the extension user to press a digit for date
and time stamp. When the mailbox owner presses the digit, the VMS will play the message details.
Default: On Demand.
8. Select the Voice Mail/Message Wait Notification Type for the extension. The VMS will notify the
extension user of the new messages in the mailbox and the message wait set by another extension user.
31. The number of person who left the message in the mailbox.
32. The number of the extension user for whom the message is intended.
If you want voice message to be played as message wait notification, record and assign a Voice
Module. See “Voice Message Applications” for instructions.
ETERNITY can play only 4 Voice Modules simultaneously. The Voice Module for Message Wait Notification
will not be played if there are already 4 being played simultaneously. In which case, Stuttered Dial Tone will
be played for Message Wait Notification, when the extension user goes OFF-Hook.
• LED Lamp: If the SIP has a 'Message Wait' lamp, you may select this option as Message Wait
Indication. The lamp will blink continuously and will be turned off when the extension user has retrieved
all the waiting messages.
• Ring: The extension will ring for the duration of the Message Wait Ring Timer (configurable; default: 30
seconds), for as many times as the Message Wait Ring Count (configurable; default: 10 times), at the
interval set as the Message Wait Ring Timer Interval (configurable; default: 30 minutes).
When the extension user answers the call, the VMS informs the user of the new message and allows
the extension user to access it.
9. To notify the extension user via e-mail about new messages in the mailbox, select the check box Voice
Mail Notification via E-mail.
• To mail the new voice message to the extension user as attachment, select With Attachment (.wav
file).
• To mail the notification only to the extension user, select Without Attachment.
• Type the E-mail Address of the extension user to which the voice mail notification is to be sent. The e-
mail ID may consist of 64 characters (maximum). Default: blank.
10. To assign Department Group mailbox to the SIP extension, select the check box Assign Mail Box of
Department Group. Default: Disabled.
You can assign the Mailbox of a Department Group to SIP extensions, even to those SIP extensions that
are not included in the Department Group. See “Department Call”.
• Select the Department Group whose mailbox you want to assign to the extension from the combo box
Specify Department Group.
2. Keep the check box Authenticate REGISTER enabled for Authentication of REGISTER Request in SIP
Messages. Default: Enabled.
3. Keep the check box Authenticate INVITE enabled for Authentication of INVITE Request in SIP
Messages. Default: Enabled.
4. Keep the check box Authenticate SUBSCRIBE enabled for Authentication of SUBSCRIBE Request in
SIP Messages. Default: Enabled.
Make sure that the Authentication ID for the SIP Extension has been programmed, when the above SIP
Message Options - REGISTER, INVITE, SUBSCRIBE - are enabled.
5. Select the check box Allow Busy Lamp Field (BLF)33 Subscription to enable the SIP extension to
monitor the status of another extension, device or number. Default: Disabled.
6. Select the check box Allow PUBLISH to allow the SIP extension user to choose whether or not to show
their Presence Status to other SIP Extensions. Default: Disabled.
33. BLF, a typical feature supported by PCM/TDM PBX and Key Telephone Systems, is also supported on SIP Extensions.
In PCM/TDM PBX and Key Telephone Systems, this feature is typically used by the Operator to monitor the status of another
extension, i.e. whether it is available, ringing or busy. The status of the other extensions is indicated on the special function keys
configured on the Operator's console. This helps the Operator decide whether to place the call, or transfer the call to that exten-
sion, or pick up the call ringing on that extension.
With BLF Subscription enabled on the SIP Extension, the user can monitor the status of another extension, device or number
within the same Proxy Domain.
7. Keep the check box Authenticate PUBLISH enabled, if you enabled Allow Publish. Default: Enabled.
8. Select the check box Allow Presence Subscription to enable the Presence Subscription on the SIP
Extension. Default: Disabled.
When Presence Subscription is enabled on a SIP Extension, the extension user can view the status of
other SIP-enabled Terminals, i.e. whether or not they are available.
The SIP Extension for which you have enabled Presence Subscription will be able to view Presence of
only those SIP Extensions which have PUBLISH enabled.
9. Select Vocoders34 in the order of preference from the drop down box.
The Vocoders supported by ETERNITY in the order of preference, i.e. 1st to 7th, by default are:
• G.723
• G.729 AB
• GSM FR
• iLBC - 30 ms
• iLBC - 20 ms
• G.711 Law
• G.711 A - Law
If you do not want to select any Vocoder, you can select the option 'None'. However, if all Vocoder
Preferences from 1 to 7 are set to 'None', incoming and outgoing calls will be blocked.
10. If you selected G.723 codec as a Preferred Vocoder, select G.723 Bit Rate as: Bit Rate: 5.3 Kbps or 6.3
Kbps. Default: 6.3kpbs.
When G.723 is negotiated, the selected Bit Rate will be applied only when sending the RTP packets.
When receiving RTP packets from the remote end, both Bit Rates of G.723 will be accepted.
11. To suppress ‘Silence’ packets and allow only Voice packets through, select the check box Silence
Suppression. Default: Disabled.
• ETERNITY NE supports Silence Suppression for all Vocoders except GSM FR.
• Silence Suppression must be disabled if you have selected 'Pass Through' as the "Fax Type".
12. Select the appropriate DTMF Option from the three DTMF types supported by ETERNITY: RTP (RFC
2833) or SIP Info, or InBand. Default: RTP (RFC2833).
The DTMF type determines how the DTMF digits will be sent over the IP Network, when a DTMF digit is
pressed.
13. To apply Echo Cancellation for SIP to CO trunk calls, SIP to Digital Trunks (Mobile, SIP) and Extensions
(SIP, SIP).
34. Vocoders are the various voice codecs used to compress the data in RTP packets for optimum use of bandwidth and for ensuring
voice quality. You can set 7 Vocoder options in the order of preference.
• Select Echo Cancellation Tail Length (msec) for Extensions and Digital Trunks. It may be 32, 64, or
128 milliseconds. Default: 32 milliseconds.
14. Configure Jitter Buffer35 to cut down on packet delays and improve voice quality.
• Select the Type of Jitter Buffer you want to use: Static or Dynamic. Default: Dynamic.
Static Jitter Buffer's internal delay is static, whereas, the Dynamic Jitter Buffer's internal delay adapts
itself to the jitter in the network.
• If you selected Static Jitter Buffer, configure Minimum Delay36, from 10 to 280 milliseconds. Default 10
milliseconds.
The value configured in the Minimum Delay determines the size of the Static Jitter buffer.
• If you selected Dynamic Jitter Buffer, configure Minimum Delay37 and Optimization Factor, from 1 to
13. Default: 10. By default Minimum Delay is set to 10 milliseconds.
The Optimization Factor determines the rate of adaptation of the Dynamic Jitter Buffer to the jitter in the
network. The minimum size of the Dynamic Jitter buffer depends on the 'Minimum Delay' configured.
In networks with higher jitter, a higher value should be configured as Optimization Factor.
The actual size of the Dynamic Jitter Buffer will be determined by the DSP on the basis of the
Optimization Factor configured and actual network condition. Dynamic Jitter buffer can go up to
maximum 300 milliseconds.
15. If you are using Fax over IP (FoIP)38, select the protocol as Fax Type. You may select:
• T.38 (UDPTL)
• T.38 (RTP)
• Pass-through.
Default: T.38 (UDPTL)
• 'Pass Through' and 'T.38' will work only if the peer devices also support the same option.
• If you select 'Pass through' as Fax type, you must disable 'Silence Suppression'.
• If the fax sent using T.38 is rejected, ETERNITY NE will use Pass Through for sending the Fax.
16. If you selected T.38 as Fax Type, configure T.38 Fax Parameters.
35. The speed at which voice packets travel through a network depends on the condition of the network. All voice packets may not
come at the same speed. This variation in the delay in receiving packets, known as Jitter, affects voice quality. Jitter Buffer helps
overcome the delay in receiving voice packets and improves voice quality. Jitter Buffer receives voice packets, stores them and
sends it to the DSP to process it at evenly spaced intervals, thus improving voice quality.
36. This parameter is to be configured for both Static and Dynamic Jitter Buffer. The Minimum Delay determines the size of the Static
Jitter Buffer and when Jitter Buffer type is Static, the Minimum Delay defines the size of the Static Jitter Buffer. 'Minimum Delay'
can be from 10 to 280 milliseconds. By default Minimum Delay is set to 10 milliseconds. The Static Jitter Buffer will store each
received voice packets for the configured time and then it will send it to DSP for voice processing.
37. When Jitter Buffer type is Dynamic, the Minimum Delay specifies the minimum time for which the Dynamic Jitter Buffer will store
the received voice packet before sending it to the DSP for voice processing.
38. You can send/receive Fax, by connecting a fax machine to the SLT port of ETERNITY NE.
• Set the Max Rate (kbps) to: 2.4, 4.8, 7.2, 9.6,12, or 14.4.
This parameter controls the Fax image transfer speed. As EQM is inversely proportional to Fax Max
Rate, if you receive poor quality fax, the Fax Max Rate should be reduced. Default: 14.4 kbps.
• Set the Packet Period (msec) to: 5, 10, 15, 20, 25, 30, 35, or 40. Default: 40.
This parameter sets the sampling rate of TDM signal. If you cannot improve fax quality by lowering Fax
Max Rate, you may reduce the Fax Packet Period.
When you reduce the Fax Packet Period, the fax image will be sent at lower speed. EQM is inversely
proportional to Fax Packet Period.
The Image Redundancy Level is redundancy level for output Image, which can be from 0 to 3.
Fax Image transfer speed is inversely proportional to this parameter. If this parameter is low then fax is
transferred faster. EQM is directly proportional to this parameter. If this parameter is high, you will
receive good quality fax.
You may increase the Fax Image Level from 1 to 3 if the quality of fax does not improve with Fax Max
Rate and Fax Packet Period.
This is a redundancy level for T.30 control data. Fax Data Level can be set from 0 to 7. Level 0 means
no redundancy. Redundancy level increases from 1 towards 7. The higher the level set, the slower
would be the fax transmission.
EQM is directly proportional to this parameter. The higher the Fax Data Redundancy Level, the better
the EQM.
17. Adjust Gain Settings, if required, to increase or decrease the incoming and outgoing volume level of calls
from SIP Digital Trunks/Extension voice calls.
• Set Rx Gain for SIP to Digital Trunk/DKP Voice call (dB) from -31 to 31 dB, if required, to increase
incoming speech volume level of calls from SIP trunks to Digital Trunks/Extension. Default: 0.
• Set Tx Gain for SIP to Digital Trunk/DKP Voice call (dB) from -31 to 31 dB, if required, to increase
outgoing speech volume level of calls from SIP trunks to Digital Trunks/Extension. Default: 0.
• You can also set the Rx and Tx Gains for SIP to Analog Trunks and Extensions (CO trunks and SLT).
• To increase Rx and Tx Gain for SIP to CO trunks, go to "“CO Trunks”".
• To increase Rx and Tx Gain for SIP to SLT extensions, go to “SLT Extensions”.
• Select dB level for Data Gain, from -31 to 31. Default: -11 dB.
• Select dB level for Bypass Gain, from -31 to 31. Default: -9 dB.
Pass Through Fax packets are transported using RTP protocol. Normally, fax calls require less gain
compared to voice calls. However, to improve fax reception, ETERNITY allows you to configure gain
settings for fax (Data Gain and Bypass Gain).
ETERNITY supports Fax Receive Gain for SIP to Digital Trunk calls as well as SIP to SLT Calls.
19. Configure Pass Through FAX Rx Gain (SIP-SLT Call), if you want to improve quality of Pass Through
Fax received on a fax machine connected to an SLT port. This parameter has relevance only when you
select ‘Pass Through’ as Fax Type.
• Select dB level for Data Gain, from -31 to 31. Default: -11 dB.
• Select dB level for Bypass Gain, from -31 to 31. Default: -9 dB.
2. Select a Profile from 1 to 4 as Voice Mail Auto Attendant Profile for the extension. Default: 1.
The profile determines the welcome message to be played to mailbox owner (extension user) when they
reach the home node. It also determines whether or not the user should be taken to the root node directly.
Make sure you also configure the Voice Mail Auto Attendant Profiles when you assign them to extensions.
To configure, click the link Voice Mail Auto Attendant Profile. See “Voice Mail Auto Attendant Profile”.
3. To allow callers to reach the SIP extension using Dial By Name option offered by the VMS Auto Attendant,
abbreviate the SIP extension user’s name to three letters and type it in the Abbreviated Name field.
When the VMS Auto Attendant is used, callers can be prompted to Dial by Name of the desired party
(extension user) instead the extension number.
• Transfer when extension answers: When the caller dials the extension number, the VMS Auto
Attendant will transfer the call when the extension answers (goes OFF-Hook).
If the extension does not answer39, the VMS Auto Attendant will transfer the call to the mailbox of the
extension, if assigned, or take the caller back to the home node.
• Take caller to Mailbox: When the caller dials the extension number, the VMS Auto Attendant will
check if the extension number has a mailbox assigned and transfer the call to the mailbox of the
extension.
• Transfer immediately: When the caller dials the extension number, the VMS Auto Attendant will
transfer the call on the extension without checking whether it is busy or free.
• Transfer when extension rings: When the caller dials the extension number, the VMS Auto Attendant
will wait for the extension to start ringing and then transfer the call.
If the extension is busy the VMS Auto Attendant will transfer the call to the mailbox of the extension, if
assigned, or take the caller back to the home node.
• Transfer when extension permits: The VMS Auto Attendant prompts the caller to record his/her
name. It puts the caller on hold and places the call on the desired extension. If the extension is free and
answers the call, the VMS announces the caller’s name to the extension user and prompts the
extension user to choose whether or not to speak to the caller. If the extension user chooses to talk, the
VMS transfers the call.
If the extension user chooses not to talk, the VMS transfers the call to the mailbox of the extension
user, if assigned, and asks the caller to leave a message.
5. Enable Announce Name, if you want the VMS Auto Attendant to announce the extension user’s name to
the caller when transferring the call to the extension. Default: Disabled.
When you enable Announce Name, make sure you record the extension user’s name on the VMS. For
instructions, see “Recording Station Names”.
DDI Routing
39. The VMS will wait for the duration of the Wait for Answer Timer (default: 15 seconds; the timer is configurable). If the call is not
answered before this timer expires, it is treated as No Reply.
2. In the field DDI Number for SIP Trunk 1, enter the DDI number you want to assign to this SIP extension
from among the DDI numbers provided by the ITSP with whom you have registered SIP Trunk 1.
3. In the field DDI Number for SIP Trunk 2, enter the DDI number you want to assign to this SIP extension
from among the DDI numbers provided by the ITSP with whom you have registered SIP Trunk 2.
4. In the field DDI Number for SIP Trunk 3, enter the DDI number you want to assign to this SIP extension
from among the DDI numbers provided by the ITSP with whom you have registered SIP Trunk 3.
5. In the field DDI Number for SIP Trunk 4, enter the DDI number you want to assign to this SIP extension
from among the DDI numbers provided by the ITSP with whom you have registered SIP Trunk 4.
6. To allow the system to land incoming calls with the same DDI number on this SIP extension, keep Allow
Incoming DDI Calls enabled. Default: Enabled.
7. To allow the system to send the DDI number as CLI when outgoing calls are made from this SIP extension,
keep Send DDI Number as CLI enabled. Default: Enabled.
If you clear the check box, the system will send the MSN number as CLI of outgoing calls made from this
extension.
SMDR Storage
The Station Message Detail Recording (SMDR) feature of ETERNITY enables you to record the details of Internal,
Incoming (IC) and Outgoing (OG) calls made from/to all its extensions. To obtain SMDR as a report, you must
enable SMDR Storage, and set filters. See “Station Message Detail Recording (SMDR)” to know more.
1. Click [+] SMDR Storage to expand options.
2. Select the type of internal calls to be stored from the combo box Store Internal Calls. You can select from
the following options:
• Made by/made to this extension (the system will store all calls made to and from this extension).
• If made by this extension (the system will store outgoing calls made from this extension)
• If made to this extension (the system will store only incoming calls from other extensions)
• Never (the system will not store internal calls)
3. To store details of incoming calls from external numbers, select Store Incoming Calls. Default: selected.
4. To store details of outgoing calls made by the extension user to external numbers, select Store Outgoing
Calls. Default: selected.
The Call Budget feature allot a 'budget' limit for outgoing calls made by the extension. See “Call Budget” for more
information. If you want to enable this feature on this extension,
3. Define the calling permission for the extension when it has used up the call budget allotted to it. Select the
desired option for Call Privilege when Call Budget is consumed:
• No Calls
• Local Calls
• Regional Calls
• National Calls
• All Calls
Default: No Calls.
This parameter is related to the CLIP feature. It allows you to choose whether the system should display the CLI of
the ‘Held Party’ or the CLI of the ‘Transferring Party’ to the transfer destination extension while the call is being
transferred.
See the feature description for “Calling Line Identification and Presentation (CLIP)” to know more.
• Display Number of Party kept on Hold when call is transferred on this extension.
This parameter is related to the feature Walk-In Class of Service. ETERNITY offers two types of Walk-In: i) One-
Call per Walk-In, where the extension user is automatically logged out after a call. ii) Walk-In until Logout, where
the extension user remains logged on until s/he manually walks out or a second user walks into the same
extension. To know more about this feature, see “Walk-In Class of Service”.
2. Select the radio button of the type of Walk-Out mode you want to assign to the extension:
• Walk-Out on completion of call automatically: Select this option, if you want to assign One-Call per
Walk-In to the extension.
• Walk-Out when user gives command to Walk-Out: Select this option, if you want to assign Walk-In
until Logout to the extension.
Call Forward
2. Set the Call Forward No Reply Timer (sec) to the desired value, if required. The range of this timer is .
Default: 030 seconds.
Call Forward No Reply Timer signifies the duration for which the system will wait for an extension to
answer an incoming call, before forwarding the call it to the programmed destination phone number as Call
Forward-No Reply. By default the Timer is set to 30 seconds. Refer the feature description for “Call
Forward” to know more.
3. Select the type of calls to be forwarded If Call is forwarded on external number. You may select from the
following options:
• Forward only Internal calls
• Forward only Trunk calls
• Forward all calls (internal as well as trunk calls)
Default: Forward only Trunk calls.
This parameter is relevant for the features “Call Forward” and “Mobility Extension”.
Configure the settings for the Matrix Extended IP Phones you have registered as SIP Extensions. You can register
three Matrix IP Phones at three different locations as a single SIP Extension.
If you have connected more than one Matrix IP Phone as a SIP Extension, configure their settings as Location 1,
Location 2 and Location 3.
2. Select the check box Enable Matrix IP Phone Mode. Default: Disabled.
3. Enter the Location Name for the phone to identify the phone. Location name may be the place where the
phone is located (e.g.: Head office, branch, residence). The Location Name may consist of 18 characters
(maximum). Default: Blank.
4. Enter the MAC Address40 of the Matrix Extended IP Phone connected at this location in hexadecimal
format, e.g. 00:50:C2:55:B0:10. Default: blank.
ETERNITY validates the IP-Phone on the basis of the MAC Address, and provides configuration on
validation.
As ETERNITY allows registration of the SIP Extension from three different locations, it identifies the SIP
Extension in each location by the programmed MAC address.
40. MAC address is the address of the electronic hardware devices such as a computer, which is hard-coded into the device during
manufacture and cannot be modified. No two devices can have similar MAC address and thus it uniquely identifies your phone.
MAC address is assigned as per the IANA standard. The MAC Address of the phone will be used as source MAC address on all
Ethernet frames.
• If the Extended IP phone is in the same network (LAN) as ETERNITY NE, select Use Ethernet IP
Address as Registrar Server IP Address.
• If the Extended IP phone is in the Global Network and ETERNITY NE is connected to Internet over
Mobile WAN, select Use Mobile Port IP Address as Registrar Server IP Address.
• If the Extended IP Phone is connected to the Global Network and ETERNITY NE is located behind a
Router, select Use Router/STUN's IP Address as Registrar Server IP Address.
Make sure the Router’s Public IP Address is configured in the Network Parameters.
• If the Extended IP Phone is connected in the Global Network and ETERNITY NE is located behind a
NAT Router, and STUN is programmed, select Use Router/STUN's IP Address as Registrar Server IP
Address.
• If Dynamic DNS is configured in the Network Parameters, select Use Dynamic DNS Host Name as
Registrar Server IP Address.
6. To set the call progress tone generation standards of the country where the Matrix IP Phone is installed,
select the Call Progress Tone - Region. Default: Region 1.
7. Select the desired Dial Tone Type from the combo box: Type 1 or Type 2. Default: Type 1.
8. Select the region/country whose Date and Time settings the IP Phone should follow. Default: India.
9. If you want to enable Daylight Saving Time (DST) on the phone, set Apply DST? to Yes. Default: No.
The Daylight Saving Time convention followed in the country/region you selected will be automatically
applied. The IP phone will change its date and time settings according to the DST convention of the
selected country/region.
10. Select the CO CLIP Pattern for the IP phone. This is the type of Calling Line Presentation on the phone for
incoming calls from trunks. You can select any of these options:
• Name Only (only the name of the caller will be displayed).
• Number Only (only the number of the caller will be displayed).
• Number + Name (both the name and the number of the caller will be displayed).
ETERNITY NE provides language support for English, French, German, Spanish, Portuguese, and Italian
on the Matrix IP Phone. When you select any of these languages, all the prompts and command strings
will appear in the selected language.
13. If you selected Ring after a delay as Ringer Mode, set the Ring Delay Timer (sec), if required, to the
desired value.
The Ring Delay Timer is the time in seconds the system waits on receiving a call before ringing on the IP
phone. The range of this timer is 0 to 99 seconds. Default: 10 seconds.
14. If you want to enable Ringer Auto Acknowledge mode, set the Acknowledge Timer (sec) to the desired
value.
The Ringer Auto Acknowledge mode determines when to stop the ring on the IP phone. There are two
options for Ringer Auto Acknowledge:
• Stop only when the call is answered.
• Stop after a delay.
To stop the ring on the IP phone after a delay, the Acknowledge Timer must be configured. The range of
this timer is 01 to 99 seconds. Default: 00 seconds.
To stop the ring only when the Call is answered or manually acknowledged, the Acknowledge Timer must
be set to '00'. By default, Ring Auto Acknowledge is turned OFF.
15. To assign the Ring Destination for the IP phone, select the desired destination for Play Ring on. You may
choose
• Speakerphone: The ring will be played on the Speakerphone.
• Headset: The ring will be played on the Headset.
Default: Speakerphone.
When you select the Headset as the destination, make sure that you set the flag ‘Headset Connected?’ to
Yes, connect a Headset to the IP Phone.
16. Select the desired Ring Tune according to your/IP phone user’s preference. Default: 1.
17. Set the Ringer Volume to the desired level, from 0 to 7, according to your preference. Default: 5.
18. To increase/decrease the volume of outgoing speech (Transmit Gain) on the handset of the IP phone, set
the Handset Transmit Volume to the desired level, from 0 to 7. Default: 4.
19. To increase/decrease the volume of incoming speech (Receive Gain) on the handset of the IP phone, set
the Handset Receive Volume to the desired level, from 0 to 7. Default: 4.
20. To increase/decrease the volume of outgoing speech (Transmit Gain) on the headset of the IP phone, set
the Headset Transmit Volume to the desired level, from 0 to 7. Default: 4.
21. To increase/decrease the volume of outgoing speech (Receive Gain) on the headset of the IP phone, set
the Headset Receive Volume to the desired level, from 0 to 7. Default: 4.
23. To change the Receive Gain of the Speakerphone MIC Volume, set Speaker Receive Volume to the
desired level, from 0 to 7. Default: 4.
24. To use a Headset with the IP phone, set Headset Connected? to Yes. Default: No.
Make sure that you connect a Headset to the IP phone, if you select Yes.
25. Select the Auto Answer check box to enable this feature on the IP phone. Default: Disabled.
When you set the “Auto Answer” feature on the IP phone, the phone goes OFF-Hook automatically after a
preset period of time, without the extension user having to pick up the handset or press the speaker or
headset key. When you enable Auto Answer, you must configure the Auto Answer Timer.
26. If you enabled Auto Answer on the phone, set the Auto Answer Timer (sec) to the desired value.
This timer defines the time in seconds that the IP phone should wait before going OFF-Hook to auto
answer a call. The range of this timer is 1 to 9 seconds. Default: 1 second.
27. Adjust the Backlight brightness of the phone’s LCD display, by setting the LCD Backlight Level to the
desired value, from 1 to 4. Default: 3.
28. Set the Back Light Off Timer (sec) to the desired value, if required, from 000 to 999 seconds. Default: 10
seconds.
29. Set the LCD Contrast Level to a level from 1 to 4 that is comfortable to you. Default: 3.
• SIP Listening Port: This is the port on which the IP phone listens for SIP messages over TCP. This
port is also used as the source port for sending SIP messages to the remote peer. The valid range for
this port is 1024-65534. Default: 5060.
• RTP Listening Port: This is the port on which the IP phone listens for SIP messages over TCP. This
port is also used as the source port for sending RTP packets. This port is also used as the source port
for sending RTP packets to the remote peer. The valid range for this port is 1025-65278. Default: 8000.
31. Set the SIP Quality of Service (QoS) for SIP signaling as:
32. If the IP phone is connected behind a NAT router, configure NAT Keep Alive.
• Select the check box Enable NAT Keep Alive to send Keep Alive messages periodically to refresh the
binding in the NAT router. Default: Disabled.
• Define as Interval (sec), the time period, from 001 to 999 seconds, after which the phone should send
Keep Alive message. Default: 120 seconds.
33. Set the following Timers to the desired value, where required:
• SIP INVITE Timer (sec): This is the time in seconds that the phone waits for a response from the
called party after ending INVITE message. This timer starts after sending INVITE message to the
called party and stops on receipt of the provisional response or the final response or when the user
disconnects the call. On expiry of the timer, the phone terminates the call process and gives an error
tone to the user. The range of the SIP INVITE TIMER is 10-180 seconds. Default: 30 seconds.
• SIP Provisional Timer (sec): This is the time in seconds that the phone waits for final response after
receiving the provisional response from the called party. This timer starts on the receipt of the
provisional response from the called party and stops on receipt of the final response from the called
party or when the user disconnects the call. On expiry of the timer, the IP phone terminates the call
process and gives error tone to the user. The range of SIP Provisional Timer is 10-180 seconds.
Default: 60 seconds.
• General Request Timer (sec): This is the time in seconds for which the phone waits for response of a
transaction request. This timer starts on initiating a transaction. This timer stops on receipt of a
response for the request. On expiry of the timer, the phone clears the transaction. This timer is used for
Registration request, etc. The range of the General Request Timer is 10-60 seconds. Default: 20
seconds.
34. To debug using Syslog Client supported by the phone, configure Debug parameters:
When the Debug flag is enabled, the phone will send the debug messages to the Syslog Server IP
address. Debug report can be viewed on the Syslog Server or any other application which can capture
the Syslog messages/debug sent by the phone.
• Enter the IP Address of the remote Syslog Server as Syslog Server Address.
• Enter the Server Port, i.e. the address of the Listening Port of the Syslog Server from 1024-65535.
Default: 514.
Syslog uses the UDP as transport protocol and listens on the port 514 (the default listening port).
• Select Debug Levels from these options, by selecting the respective check box:
• SIP
• System
• Hardware
• Call
• Menu
• User Interface
The phone supports multiple debug levels. You may select any or all of these debug levels. The Syslog
Client will send only the debug messages for the selected level to the remote server on the IP network.
For example, if the debug log of 'Call's is required, you can select this option, and disable all others.
The default key map of the Extended IP Phone appears on your screen.
3. In the Select Function Type list box, select the function to be performed by the key. For example, you
want to use the key to call the Operator.
The Operator function is a Feature, so select the option FEATURE from the Select Function Type list
box.
In the Select Offset drop down list, all the features that can be assigned to keys are listed.
• In the Select Function Type list box, select the option SACommand, as Remote DND is a System
Administrator (SA) Command.
• Click OK. The box closes. Remote DND feature will appear in abbreviated form as R-DND on the key
label.
Follow the same instructions to assign features to other DSS keys. Selecting the appropriate Function Type
and the Offset for each feature/function.
If you want assign a feature, select FEATURE as function type, and select the desired feature as Offset.
If you want to use the key to call a DKP or a SIP extension, select DKP or SIP Extension as Function Type
and select the number of the extension as Offset.
To assign direct access to a mobile trunk, select MOBILE as Function Type and the desired port number 1 or
2 as Offset.
To assign direct access to a SIP Trunk, select SIP as Function Type and the desired trunk number from 1 to 4
as Offset.
• Click OK, each time you select a Function Type and Offset in the dialog box.
• You can reinstate default key assignment any time, by clicking the Default button at the bottom of the
window.
If you have completed the configuration of the Matrix IP Phone Settings at location 1, follow the same steps as
described above to configure the Matrix Extended IP Phone at the other two locations.
More Features
Click More... link to expand.
Call Appearance
1. Set the number of Call Appearances for the SIP extension. Default: 2
A Call Appearance allows an extension user to attend to more than one calling party at a time. A minimum
of two Call Appearances must be assigned to a SIP extension - Operator extension or Executive extension
- so that the extension user can put one party on hold while talking to another. A third Call Appearance
allows the extension user to put two calls on hold, make/attend a third call and toggle between three calls.
The more the number of Call Appearances assigned to an extension, the more the number of calls the
extension user can handle simultaneously.
SIP extensions for Executives are usually assigned 2 Call Appearances, while the Operator extension is
assigned 6 Call Appearances to handle 6 calls simultaneously.
Call Taping
To use the “Call Taping” feature on the extension,
3. Now, configure the Call Taping Profile number you selected for the extension. To do this,
• Click the link Call Taping. The page opens in a new window, displaying the parameters of the profile
you selected for the extension.
• Enable Call Taping for Outgoing Calls. Type the external numbers that you want the system to
tape. You may type as many as 99 numbers.
• Select the check box Tape Calls without CLI? if you want incoming calls without CLI to be taped.
Default: Disabled.
• Select the check box Call Taping for Internal Calls, if you want internal calls made and received
by the extension to be taped.
• Close the window by clicking [x] on the top-right corner of the window.
2. Select the type of Call Duration Control to be applied from the options:
• OFF
• Apply as per profile 1
• Apply as per profile 2
• Apply as per profile 3
• Apply as per profile 4
3. Configure the Call Duration Control Profile number you selected for the extension. To do this,
• Click the settings button. The page opens in a new window. The page displays the parameters of the
profile you selected for the extension.
• Enable Apply CDC for Incoming Calls received from trunks, if CDC is to be applied on incoming
external calls. Default: Disabled.
• Enable Apply CDC for Outgoing Calls, if CDC is to be applied to outgoing external calls. Default:
Disabled.
• If required, change the CDC Timer to the desired duration. The range of the timer is 0001 to 9999
seconds. Default: 160 seconds.
• Enable Disconnect Call after CDC Timer check box if you want calls to be disconnected on the expiry
of the CDC Timer. Default: Disabled.
• In the Apply CDC for calls matching with numbers column, type the external numbers on which you
want to apply CDC. You can enter as many as 99 numbers.
• In the Do Not Apply CDC for calls matching with numbers column, enter the numbers which you
want to be exempt from CDC.
Priority
Each extension of the ETERNITY is assigned a Priority Level starting from 1, 2, 3... to 9, with '1' being lowest
Priority and '9' being highest Priority. Whenever an extension (phone) with higher priority calls an extension with
lower priority, a triple ring is placed on the called extension. To know more, read the feature description “Priority”.
1. Select a Priority from 1 to 9 for the SIP extension. Default: 5.
If this SIP extension is assigned to Operator, you may want to set a higher priority for this extension.
Personal Directory
1. Select a Personal Directory number from 01 to 50 that you want to assign to the SIP extension. Default:
None.
A Personal Directory is a list of 25 frequently dialed numbers, each of which are stored by Index number
(location code), Name and Trunk Access Codes (). The Personal Directory is necessary for using the
features “Abbreviated Dialing” and “Dial By Name”.
When a Personal Directory is assigned to an SIP, it must also be configured. The Personal Directory can
be configured also by the extension user. Refer the topic “Abbreviated Dialing” for instructions on
configuring the Personal Directory.
Time Table
1. Select a Time Table for the SIP extension. Default: System Time Table.
If you have not configured Time Table, you may do so now, by clicking the Time Table link. Define the
working days, and the start and end time of the working hours for each working day.
2. Select the type of alarm notification the system should use When User answers Alarm Call. Your options
are:
• Play Voice Message: The extension user is played a message recorded in the Voice Module on
answering the alarm call.
• Play Music-on-Hold: The extension user is played music-on-hold on answering the alarm call.
• Play VMS Alarm Greeting: The extension user is greeted by the VMS on answering the alarm call.
See
• Route to Operator: The alarm call is routed to the Operator extension, so that the Operator can the
serve the alarm request to the extension user.
Help Desk
Configure this parameter if you want to define the extension as a “Help Desk”.
2. Select the Assign Help Desk function to this Extension check box. Default: Disabled.
2. Select the check box Do not allow Outgoing Calls without Account Code (Force Account Code).
Now that you have finished configuring the SIP extension. You may configure the next SIP extension.
3. Use the Copy button to apply the same SIP extension settings you configured on this extension.
The Copy dialog box opens. The dialog box displays the name of the extension and number to be copied
in the Copy from: field.
The dialog box also lists the names and numbers of the extensions to which you can copy the values
under Copy To:
3. Click OK.
The parameter values of the extension you Copy From will be applied on the extension(s) you Copy To.
ETERNITY NE supports Door Phone. See “Door Phone” to know more about this feature and how it works.
1. Select a Time Table for the SLT extension. Default: System Time Table.
If you have not configured Time Table, you may do so now, by clicking the Time Table hyper link. Define
the working days, and the start and end time of the working hours for each working day.
2. Select the mode of routing for calls landing on the Door Phone from the combo box Route Door Phone
Calls.
• Select At wish if you want the flexibility to have calls routed to a group of extensions or to an external
number, as and when you desire.
• Select As per Schedule if you want the system to route the Door Phone Calls automatically to a
landing destination phone (extension or external number) according to the time of the day: Day and
Night.
Default: At wish.
• Select the extension numbers as landing destination. These may be SLT, DKP or SIP extensions.
• Enable Continuous Ring41 for each extension, if you want the extension to ring till the incoming call is
answered. Default: Disabled.
• Set the Ring Timer for each extension, if required. This timer defines the time for which the extension,
on which the call lands, should ring. Default: 015 seconds.
• Enable Rotation42, if you have selected more than one extension. Default: Disabled.
41. When Continuous Ring is selected, the first extension in the Door Phone extension group you have created will continue to ring,
even as the system hunts for other extensions in the group to land the call. If the call still remains unanswered, the system will
return the call to the first extension once again. This flag has no relevance, if there is only one destination extension in the group.
42. When you enable Rotation, each new call lands on the next extension to the one that received the last call. This ensures equal dis-
tribution of incoming calls to all the destination extensions. Rotation will have no relevance, if there is only one destination exten-
sion.
5. To configure external number, enter the desired destination number in the External Number field.
6. If you selected Route Door Phone Calls – As per Schedule, select the destination you want to Route
Door Phone Calls during Day to. The destination may be Extensions or External Number. Default:
Extensions.
• If you selected Extensions as the landing destination for door phone calls during the day, double click
the field Extensions for Day.
• If you selected External Number, as the landing destination for door phone calls during the night, enter
the destination number in the External Number field.
7. Select the destination you want to Route Door Phone Calls during Night to. You may select Extension
or External Number, as desired. Default: Extensions.
• If you selected Extensions as the landing destination for door phone calls at night, double click the
field Extensions for Night.
• If you selected External Number, as the landing destination for door phone calls at night, enter the
destination number in the External Number field.
The External Number is the same for Day time and Night Time, which means, you can have calls routed to
one external number only.
8. For the External number you have entered as landing destination for (At Wish and Scheduled) door phone
call route, define the outgoing trunk in the field To dial External No., Use trunk.
A multiple selection box opens. On the left, the trunks appear with their names (if configured in “Naming
Trunks”) and port numbers in a sequence, starting with CO trunks, followed by Mobile trunks and SIP
trunks43.
• To select a trunk, place your cursor on the desired trunk, and click the Select>> button.
Or
• Press the ctrl key and click the left mouse button to select multiple trunks.
To change the sequence of the trunks you selected, use the Up and Down arrow buttons on the right
display box.
To delete trunks from the ones you have selected use Delete key on your keyboard.
• Enable Rotation if you have selected more than one trunk. Default: Disabled.
43. If you have not assigned any names to the trunks, they will appear with their default names (CO MOB, SIP) and port numbers. If
you have enabled On-Site Configuration, only those trunks that are connected will appear in the box.
When Rotation is OFF, calls are routed through the first trunk in the group. If this trunk is busy, the call
is routed to the next trunk in the group.
All the trunks you selected appear in the To dial External No. Use Trunk field, in the sequence you
selected, separated by commas.
9. To apply Least Call Routing (LCR) on the trunks, select the desired LCR method from the combo box:
• Number Based: Choose this option if the service providers of the trunks you selected offer different
tariffs according to area or distance, or phone numbers dialed.
• Time Based: Choose this option if the service providers of the selected trunks offer a different tariff
according to the time of the day.
• Time + Number Based: Choose option if the service providers of the selected trunks offer different
tariffs according to the time of the day as well as area/distance.
• Service Provider Based: Choose this option if the same service providers of the selected trunks offer
different rates for calls made to numbers within their own network and for calls made to numbers of
another Service Provider's network.
• To configure or to modify the Least Cost Routing method you selected, click LCR Settings link.
A new window opens.
• Go to the LCR method you selected for Trunks.
• Configure the LCR method. See “Least Cost Routing (LCR)”for instructions.
• Click Submit. Close the window.
10. Define how long the Door Phone should ring on the destination extensions by setting Ring Door Phone
Call for to the desired value. Default: 30 seconds.
11. If you are using a Door Lock in conjunction with the Door Phone,
• Select the check box Use Door Lock. Default: Disabled.
• Set the time for which the Door Lock should Open Door for, from 01 to 99 seconds. Default: 5
seconds.
44. The first call through the first trunk, the second through the second, the third through the third trunk, and so forth. Thus each new
call is routed through the trunk next to the one that routed the previous outgoing call.
The Call Pickup (CPU) feature of ETERNITY NE allows extension users to answer (internal and trunk) calls ringing
on other extensions from their own extension; without physically going to the ringing extensions. For this extensions
must be assigned to CPU Groups.
You can create as many as 8 CPU Groups, and assign the extensions to these groups.
2. For the desired Call Pickup Group Number, Double click to Select Extensions field.
A multiple selection box opens. The extensions appear on the left, with names and numbers, as
configured. If you have not configured names and numbers, the extensions will appear with their default
access codes and extension numbers.
If you have enabled On-Site Configuration, only those extensions that are connected will appear.
3. Select the extensions which are to be assigned to the CPU Group. These may be SLT, DKP or SIP
extensions.
Place your cursor on the desired extension and click the Select>> button.
Or
Press the ctrl key and click the left mouse button to select multiple extensions.
You can also delete extensions from the ones you have selected using the Delete key of your keyboard.
4. Click OK.
All the extension numbers you selected will appear in the field Select Extensions for the CPU Group
number. The extension numbers appear in the sequence you selected, separated by commas.
5. To create another CPU group, Double click to select field against the desired group number (02 to 08).
ETERNITY NE supports a maximum of 6 Analog Trunks. The number of trunks available for configuration vary
according to your configuration.
The number of CO Trunks you have specified on the “Pre-requisites” page will be displayed on this page.
If you enabled On-site Configuration on the System Prerequisites page, CO ports that are connected to the
network will appear on this page.
On this page,
More: Click this button to view all parameter links on the page.
Less: Click this button to view only the essential parameter links on the page.
Expand: Click to expand a link to display all parameters under the link.
Collapse: Click to collapse a link. Hides all parameters under the link.
To copy the same CO parameter values to the other CO trunks, use the Copy button.
The CO Trunk numbers and the names (either default or the names you configured) appear on the tabs on this
page, starting with CO Trunks CO-1 to CO-6.
If you assigned names to the CO trunks on the “Naming Trunks” page, the same names will appear on the tabs.
To change the name of the CO trunk, you must go back to the “Naming Trunks” page.
2. Define the Calling Line Identification Format for the CO line. Selecting the format supported by your
service provider from the below options. You may consult the service provider this trunk line.
• None
• Any ETSI DTMF format
• Any FSK V.23 format
• Any FSK Bellcore format
• 1st Ring, ETSI DTMF, 2nd Ring Polarity Reversal, ETSI DTMF, 1st Ring
• 1st Ring, FSK, 2nd Ring
• DT-AS, FSK, 1st Ring
• RP-AS, FSK, 1st Ring
• Polarity Reversal, DT-AS, FSK, 1st Ring
• Any DTMF Format (without Start/Stop Code)
• If you select Extensions as the landing destination, select the landing destination extensions in the
corresponding field.
• Select the extension numbers as landing destination. These may be SLT, DKP or SIP extensions.
• Set the Ring Timer for the extensions. This timer defines the time for which the extension, on which
the call lands, should ring. Default: 015 seconds.
• Set Continuous Ring45 for the extensions, if you want the extensions to ring till the incoming call is
answered. Default: Disabled.
• Enable Rotation46, if you have selected more than one extension. Default: Disabled.
• Click OK. All the extensions you selected will appear in the Extension field, in the sequence you
selected, separated by commas.
• To route the incoming call to an auto attendant, if the selected extensions do not answer the call,
45. When Continuous Ring is selected, the first extension in the group you have created will continue to ring, even as the system hunts
for other extensions in the group to land the call. If the call still remains unanswered, the system will return the call to the first exten-
sion once again. This flag has no relevance, if you select only one landing extension.
46. When Rotation is enabled, each new call lands on the extension next to the one that received the call last. This ensures equal dis-
tribution of incoming calls to all the destination extensions in this group.
• Select the destination for the unanswered call in the field If not answered, route to. You may
select the Built-In Auto Attendant or the Voice Mail Auto Attendant. Default: Built-In Auto
Attendant.
• If you select Built-in Auto Attendant as the landing destination for calls during the day,
• the Voice Modules 02 to 13 will be played as “DID Greeting Messages” and “DID Guidance
Messages”.
• If you want to play a different message, make sure you record the desired message(s) in the Voice
Modules after completing the installation with Basic Settings. Refer the topic “Voice Message
Applications” to know more.
3. To enable “Direct Inward System Access (DISA)” on the CO trunk, select the check box Activate Direct
Inward System Access. Default: Disabled.
• PIN Authentication–Multiple calls: Select this option if you want to enable DISA with PIN
Authentication and allow multiple calls during the DISA login session.
• CLI Authentication–Multiple calls: Select this option if you want to enable DISA login with CLI
Authentication and allow multiple calls to be made during the DISA login session.
• CLI Authentication–One call Answer Signaling: Select this option if you want to enable DISA
session with CLI Authentication, and allow only a single call to be made during the DISA login
session.
This form of DISA may be used when ETERNITY NE is installed in a Gateway application. See
“Gateway Application–Answer Signaling”
If you select Multiple Calls or CLI Authentication One Call as CLI Authentication, you must configure the
DISA CLI Authentication Table under the Advanced Settings link. See “Direct Inward System Access
(DISA)”
4. To enable “Trunk Auto Answer”47 feature on the trunk, select the type of Trunk Auto Answer you want:
• For all Calls: the system will answer all incoming calls landing on the trunk line.
• When Busy: the system will answer incoming calls on the trunk, if the landing destination is busy.
• Select the Trunk Auto Answer Greeting Message you want the system to play when greeting the
callers, from the following options:
• Play Music on Hold:
• Play Greeting Message1:
• Play Greeting Message2:
• Play Greeting Message3:
• Play Greeting Message4:
Default: Do Not Play.
When you select a Greeting Message, you must record a Voice Module with the desired Greeting
Message and assign the Voice Module to Trunk Auto Answer Greeting application.
You can record 4 different Greetings Messages for Trunk Auto Answer and assign a different Greeting
message for the Day and Night. See the topic “Voice Message Applications” under Advanced Settings link
for instructions on recording and assigning voice modules to greeting messages.
• Select the Trunk Auto Answer RBT Message you want the system to play to callers from the
following options:
• Play Music on Hold
• Play RBT Message1
• Play RBT Message2
• Play RBT Message3
• Play RBT Message4
• If you do not want RBT message to be played, select Do Not Play. The system will play Ring Back
Tone to the caller.
When you select an RBT Message (1–4), you must record a Voice Module with the desired RBT Message
and assign the Voice Module to Trunk Auto Answer RBT Message application
You can record 4 different RBT Messages for Trunk Auto Answer and assign a different RBT message for
the Day and Night. See “Voice Message Applications” under Advanced Settings link for instructions on
recording and assigning voice modules for RBT messages.
• Select the Trunk Auto Answer Busy Bye Message you want the system to play to callers when the
landing destination extension is busy, from the following options
• Play Music on Hold
• Play Bye Message1
• Play Bye Message2
• Play Bye Message3
• Play Bye Message4
If you do not want Busy Bye message to be played, select Do Not Play. The system will play Busy
Tone to the caller.
47. Trunk Auto Answer enables calls landing on a trunk to be answered automatically by greeting the caller with a voice message
before the call is actually handled.
When you select a Bye Message (1–4), you must record a Voice Module with the desired Bye Message
and assign the Voice Module to Trunk Auto Answer Bye Message application
You can record 4 different Bye Messages for Trunk Auto Answer and assign a different Bye message for
the Day and Night. See “Voice Message Applications” under Advanced Settings link for instructions on
recording and assigning voice modules for Bye messages.
5. Select the landing destination to Route Calls during Night to: Operator, Extensions, Built-In Auto
Attendant, Voice Mail Auto Attendant.
Follow the same instructions as described in the previous step for selecting landing destination for the Day.
If you have selected Voice Mail Auto Attendant as the destination for Routing calls during the day or during night,
2. Select the desired Voice Mail Auto Attendant Profile number for the trunk, from 1 to 4. Default: 1.
The Voice Mail System of ETERNITY answers calls and processes them according to the Voice Mail Auto
Attendant Profile assigned to the trunk. So, you must configure Voice Mail Auto Attendant Profile number
you have selected for the trunk.
3. To configure the Voice Mail Auto Attendant Profile, click the settings icon.
A new window opens.
Configure the Voice Mail Auto Attendant Profile. For detailed instructions, see “Voice Mail Auto Attendant
Profile”. Close the window.
The feature, Automatic Number Translation of ETERNITY modifies dialed numbers or part thereof to match the
specific route numbering plan understood by the destination network—CO, GSM, VoIP—by adding or stripping
country and area codes.
For example, you can configure Automatic Number Translation such that when an extension user dials a local
landline number, the ETERNITY prefixes the number with the appropriate country-area code when it routes the call
2. Select the desired ANT Table number, table 1 to 8, which you want to apply on the trunk. Default: OFF.
The table has two columns: one for Dialed Number Strings and the other for the corresponding Substitute
Number strings. The number strings are stored against an Index number, from 1 to 32. You can to enter as
many as 32 number strings.
6. Enter the Dialed Number strings and their corresponding Substitute Number strings in the respective
columns. Enter the Dialed Numbers and their corresponding Substitute Numbers against each Index
number. For example, if '95' is entered as the Dialed Number at Index 01 and its Substitute Number '91'
must be entered at Index 01.
SMDR Storage
The Station Message Detail Recording (SMDR) feature of ETERNITY enables you to record the details of Internal,
Incoming (IC) and Outgoing (OG) calls made from/to all its trunks and extensions. To obtain SMDR as a report, you
must enable SMDR Storage, and set filters. See “Station Message Detail Recording (SMDR)” and “Station
Message Detail Recording–Storage” to know more.
1. Click [+] SMDR Storage to expand options.
2. To store details of incoming calls landing on this trunk, select the check box Store Incoming Calls.
Default: selected.
3. To store details of outgoing calls made from this trunk, select the check box Store Outgoing Calls.
Default: selected.
4. Click Submit.
This parameter determines what the system should do when an external called party in speech on this trunk is put
on Hold by an extension user by pressing a DSS key to dial another port.
For example, the DKP extension user on DKP1 port is in the middle of speech with an external party on a CO2.
If the extension user of DKP1 presses a DSS key to call another extension port, DKP2, two situations are possible,
depending on the DSS Key Interface you configure:
i. The external caller on CO2 will be played music-on-hold, and the user on DKP1 will hear Ring Back
Tone. The call will be placed on DKP2.
ii. The external caller on CO2 will be disconnected. DKP1 user will hear Ring Back Tone. The call will be
placed on DKP2.
2. Select the radio button of the desired DSS Key Interface option:
• Disconnect current call when DKP user presses DSS key of other trunk, while in speech using
this trunk.
• Put current call on Hold, when DKP user presses DSS key of other trunk, while in speech using
this trunk. (Default).
3. Click Submit.
Cost Factor
Cost Factor is used for grading the cost of routing calls from a trunk for the purpose of least cost routing. See “Least
Cost Routing (LCR)”
2. Select a Cost Factor for this trunk from the drop down list, from 01 to 99. Default: 01.
Each Call Cost Calculation Pulse Rate option contains a Pulse Rate Type, which you must configure.
Some service providers offer discounted rates for holidays. In which case, you can configure the rates for
normal days in the Normal Pulse Rate Table and the rates for holidays in the Holiday Pulse Rate Table.
See “Call Cost Calculation (CCC)”
Configure the Pulse Rate Type according to the billing pattern of the service provider of this trunk.
Trunks following the same billing pattern should be assigned the same Pulse Rate Option.
2. Define the Call Cost Calculation Schedule for the Pulse Rate option you selected.
The Pulse Rates offered by service providers may vary according to the time of the day. In such cases, you
must divide the day into Time Zones, from 1 to 4, as required, to match the time of the pulse rates offered
by your service provider.
Specify the Start Time and the End time (in 24 hours, HH:MM format) for the Time Zone Index in which
the particular Pulse Rate will be applied.
The default Time Schedule (start and end time) for each Time Zone Index is as follows:
T1 00:00 23:59
T2 00:00 23:59
T3 00:00 23:59
T4 00:00 23:59
If your service provider offers the same Pulse Rate for calling a number for the entire day, there is no need
to change the default Time Schedule. The system will follow Time Zone 1.
Call Budget
Configure this parameter if you want to control the cost of phone calls made from this trunk. See “Call Budget on
Trunk” to know more.
1. To enable call budget on the trunk, set Use Call Budget? to Yes. Default: No.
2. Define the call budget in terms of number of calls in the field Restrict outgoing calls if total number of
calls exceeds. Default: 9999 (in local currency)
3. Define the call budget in terms of amount in the field Restrict outgoing calls if total cost of the calls
exceeds. Default: 999999 (local currency).
4. Define the call budget in terms of duration in the field Restrict outgoing calls if total call duration
exceeds. Default: 999999 minutes
5. To have the system reset the Number of Calls/Amount/Minutes automatically on a particular date, select
the desired date from the Reset Call Budget on combo box as 1st to 31st of the every month.
The consumed Call Budget Amount/Minutes/Number of Calls can be reset from SE and SA Mode, referred
to as Manual Reset. Refer the feature description “Call Budget on Trunk”.
Call Back
Configure this parameter if you want to enable the ‘Call Back on Trunk Port’ feature on the trunk. See “Call Back on
Trunk Ports” to know more.
1. Select a Call Back Profile, 1 to 4, for the trunk port from the combo box.
Make sure you also configure the Call Back profile you select here.
Hardware Features
2. Select your Trunk Type. Three types of CO trunks can be interfaced to a CO port of the ETERNITY NE:
• Normal Dial type: This is the conventional CO available from the PSTN.
• Hotline type: The CO connecting two destinations immediately on grabbing the trunk.
• Hotline Dial type: A special CO available from the PSTN, which works as a normal dial type for some
time and works as a hotline thereafter.
Default: Normal.
3. Select AC Termination Impedance. The AC Termination Impedance of the CO port must match with the
AC Termination Impedance supported by the PSTN network. By default, the AC Termination Impedance is
set as per the “Region” you have selected; however, you may select the impedance supported by your
PSTN network from the following options.
• 600
• 900
• 270+(750 || 150 nF) and 275+(780 || 150 nF)
• 220+(820 || 120 nF) and 220+(820 || 115 nF)
• 370+(620 || 310 nF)
• 320+(1050 || 230 nF)
Default: 600 ohms (for default Region ‘India’)
You can configure the Dialing method as Pulse or Tone (with configurable Pulse Ratio and DTMF ON-OFF
period) according to the Dialing method supported by the CO network to which the CO port is connected.
5. If you selected Pulse as Dial Type, select the appropriate Pulse Dial Ratio from the following options
according to the type of Pulse Dialing Ration supported by your CO Network.
• 10PPS, 1:2
• 10PPS, 2:3
• 10PPS, 1:1
• 20PPS, 1:2
• 20PPS,2:3
• 20PPS, 1:1
Default: 10 PPS 1:2.
6. If required, you may You can increase or decrease the level of Incoming Speech (Receive Gain) on the
Trunk by changing the Rx Gain (dB) to the desired level.
7. To increase or decrease the level of Outgoing Speech (Transmit Gain) on the Trunk, change the Tx Gain
(dB) to the desired level.
8. Select the type of Answer Supervision48 supported by your Network from the following options. Default:
Pseudo Answer.
• Pseudo Answer: It is used when no signaling is available from the PSTN. If this option is selected, the
call will be considered as matured on the expiry of the 'Pseudo Answer Supervision Timer'
(configurable; default 20 seconds), irrespective of whether or not the call actually gets matured. After
this, the Call Duration Timer starts. Finally, the system starts detecting the "Disconnect Supervision"
signal configured for the CO port.
• Polarity Reversal: It is used as maturity signal when the answer signaling is given in the form of
Battery Reversal. If the battery polarity of the line is -ve for TIP and +ve for RING, when the called party
has answered the call, the CO network will reverse the battery polarity, i.e. TIP becomes +ve and Ring
-ve. After this, the Call Duration Timer is started. Finally system starts detecting the Disconnect
Supervision signal configured for the CO port.
• 12 KHz/16 KHz Pulse: When called party answers the call, the CO network generates a 12/16 KHz
metering pulse to indicate call maturity. Finally, the system will start detecting Disconnect Supervision
signal configured for the CO port.
48. It is a signal from the CO network to indicate to call maturity. Whenever you make an outgoing call from CO Trunk, the CO network
will give answer signaling when the called party answers the call.
This feature is particularly useful if you want to use “Call Cost Calculation (CCC)” to enable accurate billing. When the signal is
received, the billing will start and in the absence of this signal, the call will not be billed, ensuring that unanswered and unsuccess-
ful call attempts are not billed.
9. If you selected Pseudo Answer as Answer Supervision signal, configure the Pseudo Answer
Supervision Timer.
This is the time period for which the system will wait before treating a call as matured (regardless of
whether or not it was answered). The range of this Timer is from 001 to 255 seconds. Default: 20 seconds.
When Pseudo Answer is selected as Answer Supervision signal, the call duration measured by the system
will not accurately reflect the actual call duration because the Pseudo Answer Supervision Timer is not
related to the actual call maturity. For example, if the Pseudo Answer Supervision Timer is set to 15
seconds, the call will be considered as matured after 15 seconds, even if it is answered after 20 seconds.
Similarly, if this Timer is set to 80 seconds, but the call was answered after 20 seconds and disconnected
after 40 seconds, this call will never be considered as matured as it ends before 80 seconds.
10. Select the Disconnect Supervision49 signal type supported by your network from the following options.
Default: None.
• None: When there this no signaling supported. Select this option only if there is no Disconnect
Supervision signal supported.
• Polarity Reversal: Call disconnection is signaled as Polarity Reversal when the call is
disconnected by the remote user. For example, if the battery polarity of the CO port is '+ve' for TIP
and '-ve' for RING in speech condition then on disconnection by the remote user, TIP will become '-
ve' and Ring '+ve'. The user gets an Error tone and the CO port is released.
• Open Loop Disconnect: Call Disconnection is signaled in the form of Open Loop, whereby the
Battery voltage on the CO port is removed for a short duration. Voltage is restored after this short
duration. However, the Polarity of Battery Voltage on the CO port is not changed.
This option is to be selected when call disconnection is signaled in the form of Open Loop
Disconnect pulse by the CO network. System will check Open Loop Disconnect signal for the time
duration set in Open Loop Disconnect Timer for each CO port. If the time of the Open Loop signal
detected is less than the Open Loop Disconnect Timer you have set, it will not be considered as
49. It is a signal from the CO network to indicate call disconnection. Whenever a call (incoming or outgoing) made from the CO Trunk
is disconnected by the remote party, the CO network will send Disconnect signal to the CO port. ETERNITY will detect this signal
and release the CO port.
Disconnect Supervision signal is important when a PCO machine is connected to the (SLT Port) ETERNITY and or when ETER-
NITY NE is deployed in a Gateway application.
In such application scenarios, it is desirable that calls that are disconnected by either end - calling party or called party - is termi-
nated by the system and the port is released. If the called (remote) party has disconnected the call but the calling party (extension
that made the outgoing call from ETERNITY) has not disconnected the call, the call remains live within the system.
So, Disconnect Supervision signal is important, particularly when calls are routed from CO-to-CO ports, to indicate to the system
that it needs to disconnect the call and release the port.
• Select the same Answer Supervision and Disconnect Supervision signal type supported by your CO
network for the CO ports. Consider the following case:
• The CO network supports Polarity Reversal signal as Answer and Disconnect Supervision.
• But you have selected 'Pseudo Answer' as Answer Supervision signal and 'Polarity Reversal' as
Disconnect Supervision signal for the CO ports in the system.
• In this case, when a call is made through the CO port, the call will be considered as matured after
the Pseudo Answer Supervision Timer.
• Now, when the called party answers the call, the CO generates 'Polarity Reversal' as answer
supervision signal on the CO port.
• But as 'Polarity Reversal' is also selected as the Disconnect Supervision for the port, the system will
interpret this (Answer Signaling) signal as Disconnect Supervision signal and disconnect the call.
11. If you selected Open Look Disconnect as Disconnect Supervision type, configure the Open Loop
Disconnect Timer. The range of this timer is from 001 to 999 milliseconds. Default: 200 ms.
12. If Call Disconnection is signalled by your CO Network in the form of Disconnect Tone, configure
Disconnect Tone Detection.
To enable the system to detect the Disconnect Tone accurately, you must configure the Cadence (ON-
OFF time) and Frequency of the Disconnect Tone, as supported by the CO network.
• Operator: This parameter has 3 options: No operator, Modulation (*), Addition (+)
If Modulation is selected, frequency 1and frequency 2 will be used as modulation i.e. F1* F2
If Addition is selected, frequency 1 and frequency 2 will be used as addition i.e. F1 + F2.
• Frequency 2 (Hz): Frequency 2 is from 20 to 1400 Hz. Select Frequency 2 if the Disconnect Tone
supported by the CO network consists of Dual Frequency.
• First Cycle Cadence - ON Time 1 (msec), OFF Time 1 (msec): Cadence is configurable from 0040-4000
msec for ON Time 1 and OFF Time 1.
• Subsequent Cycles: ON Time 2 (msec), OFF Time 2 (msec): Cadence is configurable from 0040-4000
msec for ON Time 2 and OFF Time 2.
When disconnect tone is detected on port matches the Frequency and Cadences you have configured, the
call will be disconnected and the CO port will be released.
13. Configure DTMF Out Dial parameters. While dialing out the DTMF digits from the CO port, the following
attributes of DTMF signal50 are critical.
• DTMF Signal ON Time (msec): It is the width of DTMF digit to be dialed out by the CO port. Set the
ON time to the desired value. Default: 102 milliseconds.
The 'level' of each DTMF digit is fixed, at -6.0 dB, but you may configure these parameters to match the
CO network requirement.
14. Configure the following DTMF Detection parameters, if necessary. The default settings of DTMF
Detection serve the requirements of most of the applications. However, you may fine tune the following
parameters if you face any problems in DTMF detection.
• Minimum Level (dB): This is the minimum level (dB) of the DTMF digit to be considered as valid.
Default: -4.5dB.
• Minimum ON Time (msec): This is the minimum time period for which the DTMF signal should be
present in order to be detected. The valid range of this time is 17 to 204 milliseconds. By default: 34
milliseconds.
• Minimum OFF Time (msec): This is the minimum time period between successive DTMF digits. The
valid range of this time is 17 to 204 milliseconds. Default: 68.
• Speech Delay Timer: It is the time after which the system gives dial tone to the extension, when the
extension user grabs the CO Trunk.
To understand the significance of this timer, let us consider a situation. Extension 201 does not have
calling permission for long distance numbers. The user of extension 201 grabs a CO Trunk, and dials a
number 1022-6305555. The system dials out this number, as it starts with '1', but since the actual dial tone
from the CO comes after some time, the CO interprets this number as 022-6305555 and establishes
speech. This way an extension user who does not have permission for long distance calling, can dial out a
long distance number. This can be prevented by configuring the Speech Delay Timer to an appropriate
value.
The range of this timer is from 000 to 255 seconds. Default: 1 second.
• Pause Timer: It is the time for which the system waits after grabbing the CO Trunk to dial out an external
number.
When the system automatically dials out the number51, if the CO network is of the old type, it is possible
that the system may dial out the number even before it gets the dial tone. This may result in a wrong
number being dialed out. The Pause Timer helps avoid this.
When features such as Redial or Auto-Redial or Abbreviated Dialing are used, the ETERNITY NE grabs
the trunk line, waits for the Pause Timer before dialing out the number.
The range of this timer is from 0500 to 3000 milliseconds. Default: 1000 milliseconds.
50. These DTMF Out Dial attributes would be used when the features Redial, Auto Redial and Abbreviated Dialing are used to dial out
the numbers from the CO port. These attributes are also applicable when you make a call from a DKP that has DTMF generation
disabled.
51. When features such as Redial or Auto-Redial or Abbreviated Dialing are used, the ETERNITY grabs the trunk line and dials the
number on its own.
To get correct indication, the ETERNITY NE supports the Ring Cadence OFF timer on the CO port so that
ring can continue, even for incoming calls with long Ring OFF period.
The range of the Ring Cadence OFF timer is from 1 to 6 seconds. Default: 6 seconds.
• Flash Time (msec): This parameter is relevant for dialing out Flash on the CO Trunk to access some of
the features of the CO Network. Configure the desired time of Flash to be generated on the CO Trunk.
Default: 600 milliseconds.
• ON Hook Speed: This parameter allows you to set the duration of time for the line-side device to go on-
hook.
The ON-Hook speed you set here is measured from the time the ON-Hook bit is cleared until the loop
current equals zero. Select the desired ON-Hook Speed from the following options:
• <0.5msec
• 3 msec
• 26 msec
Default: <0.5msec.
• OFF Hook Speed: This parameter defines the time to settle the line transients after which transmission or
reception can occur. Select the desired OFF-Hook Speed from the following options:
• 512 ms
• 128 ms
• 64 ms
• 8 ms
Default: 8 ms.
16. To limit Loop Current, enable Current Limiting. Default: Disabled. When this flag is enabled, the Loop
Current is limited to a maximum of 60mA.
17. Set the Minimum Loop Current at which DAA module of the CO Trunk port can operate. Select the
minimum operational loop current from the following options as per your requirement:
• 10
• 12
• 14
• 16
Default: 10 mA.
18. If required, you may adjust the Tip Ring Voltage on the line side.
In countries where Low voltage is required, you should set lower TIP/RING voltage in ETERNITY. Adjust
the values of the Tip Ring Voltage to match your country requirements from the following options:
• 3.1
• 3.2
• 3.35
• 3.5
Default: 3.5.
'High' signifies 20Mohm Ringer Impedance. This is the default Ringer Impedance provided on the line side
by the DAA module of the CO port. The DAA Module can provide higher impedance when 'Synthesized'
impedance is selected.
Some countries like Poland, South Africa and Slovenia require higher ring impedance which is achieved by
the DAA module, when Ringer Impedance is set to 'Synthesized' impedance.
20. Set the Ringer Threshold (Vrms) level. This parameter defines the level below which the CO port would
not validate the Ring signal, and the level above which it would validate the Ring signal. Set the Ringer
Threshold to the desired value from the following options:
• 13.5 - 16.5
• 19.35 - 23.65
• 40.5 - 49.5
Default: 13.5 - 16.5 Vrms.
If the CO trunk port you are configuring is located in a “Behind the PBX Application”, configure PPDC. 'Pre-
PSTN Digit Count' or PPDC is the number of digits (dialed by an extension) to be ignored by the system
before toll control check is begun. It is the same as the number.
In Behind the PBX Applications, another PBX may be connected to the ETERNITY NE, with some of its
CO Trunks terminating into the Extension ports of the other PBX, and other trunks being directly
connected to the PSTN.
If the CO Trunk port is directly connected to the PSTN, PPDC must be set to '0'.
For Trunk ports connected to extensions of another PBX, PPDC must be configured as per the number of
digits in the Trunk Access Code defined for that PBX.
If the Trunk Access Code is a single digit, select '1'. If the Trunk Access Code is double or triple digit
number, accordingly select '2' or '3' as PPDC.
To know more about this feature, refer “Behind the PBX Application”.
21. To increase the incoming speech volume level of calls from SIP trunks to this CO trunk, set Rx Gain at SIP
Trunk (dB). Default: 0dB.
22. To increase the outgoing speech volume level of calls from SIP trunks to CO trunks, set Tx Gain at SIP
Trunk (dB). Default: 6dB.
23. To improve the quality of Fax over IP52, configure Pass Through Fax–Data Gain at SIP Trunk, if you
selected Pass Through Fax as Type of Fax over IP on SIP trunks, and if Pass Through Fax is to be
received on a CO trunk. Select the dB Level for Data Gain. Default: -11 dB.
24. Select the dB Level for Pass Through Fax–Bypass Gain at SIP Trunk. Default: -9 dB.
52. Normally, fax calls require less gain compared to voice calls. However, to improve fax reception, ETERNITY allows the configuring
of gain settings for fax. Fax gain settings consist of Data Gain and Bypass Gain. ETERNITY supports Fax Receive Gain for SIP to
CO Trunks, SIP to Digital Trunk calls as well as SIP to SLT Calls.
More Features
Click More... link to expand.
Priority
Each trunk of the ETERNITY NE can be assigned a Priority Level starting from 1, 2, 3... to 9, with '1' being lowest
Priority and '9' being highest Priority. Whenever there are incoming calls on multiple trunks, the call on the trunk
with higher priority will be answered by the Operator extension first. To know more, see “Priority”.
Time Table
1. Select a Time Table for the CO trunk. Default: System Time Table.
If you have not configured Time Table, you may do so now, by clicking the Time Table link. Define the
working days, and the start and end time of the working hours for each working day.
1. Select the check box Do not allow Outgoing Calls without Account Code to enable Forced Account
code on the trunk. Default: Disabled.
When you enable this flag, the system will prompt extension users to dial the Account Code whenever they
grab this trunk to dial out a number. The system will allow extension users to dial out numbers only after
they have dialed the Account Code or Name.
Account Codes feature must also be enabled in the Class of Service of extension users who are to be
allowed this feature.
The Copy dialog box opens. The dialog box displays the number and name of the trunk to be copied in the
Copy from: field.
The dialog box also lists the numbers and names of the trunks to which you can copy the values under
Copy To:
3. Click OK.
The parameter values of the trunk you Copy From will be applied on the trunk(s) you Copy To.
• On this page, the number of Mobile Trunks you have specified on the “Pre-requisites” page will be
displayed on this page.
• If you enabled On-site Configuration on the Prerequisites page, the Mobile port that has a SIM Card
present will appear on this page.
On this page,
More: Click this button to view all parameter links on the page.
Less: Click this button to view only the essential parameter links on the page.
Expand: Click to expand a link to display all parameters under the link.
Collapse: Click to collapse a link. Hides all parameters under the link.
To copy the same mobile trunk parameter values to the other mobile trunks, use the Copy button.
Click the desired Mobile trunk number/name tab to configure the parameters.
If you assigned names to the mobile trunks on the “Naming Trunks” page, the same names will appear on the tabs.
For example, if you have named Mobile Port 1 as Vodafone, Mobile Port 2 as Airtel, these names will appear in
place of Mobile 1 and Mobile 2.
If you want to change the name of the Mobile port, you must go back to the “Naming Trunks” page.
SIM PIN
1. You may change the SIM PIN for the SIM Card inserted in the mobile port.
Recall that you have changed the SIM PIN of the SIM Card using a Handset before installing it in the
system to the default '1234'. Now, you may change it to the desired SIM PIN.
The SIM PIN may be up to 8 digits long. Only the digits from 0 to 9 are allowed.
Incoming Calls
1. Select the route for incoming calls from the following options:
• Route calls as programmed for Day and Night time. Select this option if you want incoming calls to
be routed to specific destinations during the day and night. If you select this option, you must also
configure the landing destinations for incoming calls during the day and the night.
• Ignore all incoming calls53.
• Reject all incoming calls.
53. In this case, the caller will continue to get ring back tone until the call is timed out by the network. No outgoing call can be made
from the mobile port until the incoming call times out and the port is released.
• Select the landing destination to Route Calls during Day to. You may select:
• Operator
• Extensions
• Built-in Auto Attendant
• Voice Mail Auto Attendant (if available).
• If you select Extensions as the landing destination, select the landing destination extensions in the
corresponding field.
• Select the extension numbers as landing destination. These may be SLT, DKP or SIP extensions.
• Set the Ring Timer for the extensions. This timer defines the time for which the extension, on which
the call lands, should ring. Default: 015 seconds.
• Set Continuous Ring54 for the extensions, if you want the extensions to ring till the incoming call is
answered. Default: Disabled.
• Click OK. All the extensions you selected will appear in the Extension field, in the sequence you
selected, separated by commas.
• To route the incoming call to an auto attendant, if the selected extensions do not answer the call,
• Select the check box Ring Extension/s and set the ring duration in the for___seconds field.
Default: 10 seconds. This is the time for which the system will ring on the destination extensions
you selected, and wait for an extension to answer. If the call remains unanswered on the expiry
of this timer, ETERNITY will route the call to the auto attendant.
• Select the destination for the unanswered call in the field If not answered, route to. You may
select the Built-In Auto Attendant or the Voice Mail Auto Attendant. Default: Built-In Auto
Attendant.
• If you select Built-in Auto Attendant as the landing destination for calls during the day, the
Voice Modules 02 to 13 will be played to callers as “DID Greeting Messages” and “DID
Guidance Messages”.
• If you want to play a different message, make sure you record the desired message(s) in the
Voice Modules after completing the installation with Basic Settings. Refer the topic “Voice
Message Applications” to know more.
3. To enable “Direct Inward System Access (DISA)” on the mobile trunk, select the check box Activate
Direct Inward System Access. Default: Disabled.
• PIN Authentication–Multiple calls: Select this option if you want to enable DISA with PIN
Authentication and allow multiple calls during the DISA login session.
• CLI Authentication–Multiple calls: Select this option if you want to enable DISA login with CLI
Authentication and allow multiple calls to be made during the DISA login session.
• CLI Authentication–One call: Select this option if you want to enable DISA session with CLI
Authentication, and allow only a single call to be made during the DISA login session.
If you select Multiple Calls or CLI Authentication One Call as CLI Authentication, you must configure the
DISA CLI Authentication Table under Advanced Settings. See “Direct Inward System Access (DISA)”
4. To enable “Trunk Auto Answer”56 feature on the trunk, select the type of Trunk Auto Answer you want:
54. When Continuous Ring is selected, the first extension in the group you have created will continue to ring, even as the system hunts
for other extensions in the group to land the call. If the call still remains unanswered, the system will return the call to the first exten-
sion once again. This flag has no relevance, if you select only one landing extension.
55. When Rotation is enabled, each new call lands on the extension next to the one that received the call last. This ensures equal dis-
tribution of incoming calls to all the destination extensions in this group.
56. Trunk Auto Answer enables calls landing on a trunk to be answered automatically by greeting the caller with a voice message
before the call is actually handled.
• When Busy: the system will answer incoming calls on the trunk, if the landing destination is busy.
• Select the Trunk Auto Answer Greeting Message you want the system to play when greeting the
callers, from the following options:
• Play Music on Hold:
• Play Greeting Message1:
• Play Greeting Message2:
• Play Greeting Message3:
• Play Greeting Message4:
When you select a Greeting Message, you must record a Voice Module with the desired Greeting
Message and assign the Voice Module to Trunk Auto Answer Greeting application.
You can record 4 different Greetings Messages for Trunk Auto Answer and assign a different Greeting
message for the Day and Night. See the topic “Voice Message Applications” under Advanced Settings link
for instructions on recording and assigning voice modules to greeting messages.
• Select the Trunk Auto Answer RBT Message you want the system to play to callers from the
following options:
• Play Music on Hold
• Play RBT Message1
• Play RBT Message2
• Play RBT Message3
• Play RBT Message4
• If you do not want RBT message to be played, select Do Not Play. The system will play Ring Back
Tone to the caller.
When you select an RBT Message (1–4), you must record a Voice Module with the desired RBT Message
and assign the Voice Module to Trunk Auto Answer RBT Message application.
You can record 4 different RBT Messages for Trunk Auto Answer and assign a different RBT message for
the Day and Night. See “Voice Message Applications” under Advanced Settings link for instructions on
recording and assigning voice modules for RBT messages.
• Select the Trunk Auto Answer Busy Bye Message you want the system to play to callers when the
landing destination extension is busy, from the following options
• Play Music on Hold
• Play Bye Message1
• Play Bye Message2
• Play Bye Message3
If you do not want Busy Bye message to be played, select Do Not Play. The system will play Busy
Tone to the caller.
When you select a Bye Message (1–4), you must record a Voice Module with the desired Bye Message
and assign the Voice Module to Trunk Auto Answer Bye Message application.
You can record 4 different Bye Messages for Trunk Auto Answer and assign a different Bye message for
the Day and Night. See “Voice Message Applications” under Advanced Settings link for instructions on
recording and assigning voice modules for Bye messages.
5. Select the landing destination to Route Calls during Night to: Operator, Extensions, Built-in Auto
Attendant, Voice Mail Auto Attendant.
Follow the same instructions as described in the previous step for selecting landing destination for the Day.
1. To apply the RCOC (Return Call to Original Caller) feature on the mobile trunk, select the check box Set
RCOC for Calling Extension, when Called Number is busy/ switched-off / not responding.
Default: Disabled.
When RCOC is enabled on the Mobile trunk port, the system routes calls returned by remote parties back
to the extensions that originally made the call from this port (i.e. the original callers' extensions). To know
more, refer the feature description for “RCOC (Return Call to Original Caller)”.
1. If you have selected Voice Mail Auto Attendant as the destination for Routing calls during the day or during
night,
• Select the desired Voice Mail Auto Attendant Profile number for the trunk, from 1 to 4. Default: 1.
The Voice Mail System of ETERNITY answers calls and processes them according to the Multi-level
DID Profile assigned to the trunk. So, you must configure the VMS Auto Attendant Profile number you
have selected for the trunk.
• To configure the Voice Mail Auto Attendant Profile, click the settings icon.
• Configure the Voice Mail Auto Attendant Profile. For detailed instructions, see “Voice Mail Auto
Attendant Profile”. Close the window.
The feature, Automatic Number Translation of ETERNITY modifies dialed numbers or part thereof to match the
specific route numbering plan understood by the destination network—CO, GSM, VoIP—by adding or stripping
country and area codes.
For example, you can configure Automatic Number Translation such that when an extension user dials a local
landline number, the ETERNITY prefixes the number with the appropriate country-area code when it routes the call
through the GSM network. To know more about this feature, see the description for “Automatic Number
Translation”
1. If you want to apply the Automatic Number Translation57 feature on the mobile trunk,
• Select the desired ANT Table number, table 1 to 8, which you want to apply on the trunk. Default: OFF.
57. Automatic Number Translation modifies dialed numbers or part thereof to match the specific route numbering plan understood by
the destination network—CO, GSM, VoIP—by adding or stripping country and area codes.For example, you can configure Auto-
matic Number Translation such that when an extension user dials a local landline number, the ETERNITY prefixes the number with
the appropriate country-area code when it routes the call through the GSM network.To know more about this feature, see the
description for “Automatic Number Translation”.
The table has two columns: one for Dialed Number Strings and the other for the corresponding
Substitute Number strings. The number strings are stored against an Index number, from 1 to 32. You
can to enter as many as 32 number strings.
• Enter the Dialed Number strings and their corresponding Substitute Number strings in the respective
columns. A dialed number and its corresponding substitute number must be stored at the same index.
For example, if '95' is entered as the Dialed Number at Index 01, its Substitute Number '91' must be
entered at Index 01.
SMDR Storage
1. To record details and generate reports of Internal, Incoming and Outgoing calls made from/to the mobile
trunk, enable SMDR Storage. See “Station Message Detail Recording (SMDR)” and “Station Message
Detail Recording–Storage” to know more.
• Select the check box Store Incoming Calls to store details of incoming calls landing on this trunk.
Default: selected.
• Select the check box Store Outgoing Calls to store details of outgoing calls made from this trunk.
Default: selected.
• Set the Network Registration Retry Count from 1 to 255. This is the number of attempts the mobile
port will make to register with the network of the service provider whose SIM you have inserted in the
port. Default: 50 attempts.
The mobile port is configured to automatically locate and register with the Network that supports the
SIM card installed on it. Also, at each power ON, the mobile port (i.e. SIM) will automatically register
with the Network that supports the SIM on it.
However, if the Mobile port fails to register, it will restart the process of network registration on the
expiry of the Network Registration Retry Timer58. On the expiry of this timer, the system will retry
registration for the Count (i.e. number of times) configured and with each re-try attempt, the count will
be decremented by one.
• Select the Frequency Band supported by the service provider network whose SIM card is installed in
the mobile port. Default: 900+1800MHz.
• Frequency Band selection not required if the Mobile Card has SIMCOM 3G module.
• When you change the Frequency Band, the change will be effected after the next system restart or the
next Mobile Port restart.
• Set the Network Selection Mode for the mobile port as Automatic or Manual. Default: Automatic.
58. The Network Registration Retry Timer defines the time for which the Mobile port, which has failed to register with the network,
should wait before attempting to re-register with the network. Network registration retry timer is 2 minutes and is non-configurable.
Manual Network Selection mode may be enabled when there is more than one network operator in
available in the area59, so as to prevent the SIM card from registering with another available network
and resulting in 'Roaming' charges.
• Select the Network Preferred Mode for the mobile port as Dual, GSM or UMTS. Default: Dual Mode.
• If you have enabled Manual network selection mode, configure the table Priority for Manual Network
Selection.
In this table, enter the Network Operator Codes (MCC-MNC)60 in the order of priority for the Mobile
Port. The codes must not exceed 8 digits. You can store up to 9 Network Operator Codes in the order
of priority.
So, whenever you register with the network manually, the system will select the Network Operator that
matches in order of priority. If the Mobile port fails to register, it will restart the process of network
selection on the expiry of the Network Registration Retry Timer.
If no match is found, the Mobile port (SIM) will not get registered with any of the available network
operators and no calls can be made or received on this port.
• When you change the Network Selection Mode to ‘Manual’ and the Network Operator Code manually,
the change you made will not come into effect until you have restarted the system.
This parameter determines what the system should do when an external called party in speech on this
trunk is put on Hold by a DKP extension user by pressing a DSS key to dial another port.
For example, the DKP extension user on DKP1 port is in the middle of speech with an external party on a
Mobile 1. If the extension user of DKP1 presses a DSS key to call another extension port, DKP2, two
situations are possible, depending on the DSS Key Interface you configure:
The MNC is usually a 2/3-digit code. The MCC-MNC combination uniquely identifies the home network of the mobile terminal or
the mobile user. For example, AirTel, a GSM network operator in India, has different MNC assigned to its networks in various
states. The MNC for AirTel in the state of Maharashtra is 90, while the same for the state of Gujarat is 98.
b. The external caller on Mobile 1 will be disconnected. DKP1 user will hear Ring Back Tone. The call will
be placed on DKP2.
• Select the radio button of the desired DSS Key Interface option:
• Disconnect current call when DKP user presses DSS key of other trunk, while in speech
using this trunk.
• Put current call on Hold, when DKP user presses DSS key of other trunk, while in speech
using this trunk. (Default).
Cost Factor
Cost Factor is used for grading the cost of routing calls from a trunk for the purpose of least cost routing. See “Least
Cost Routing (LCR)”
2. Select a Cost Factor for this trunk from the drop down list, from 01 to 99. Default: 01.
Configure the Pulse Rate Type according to the billing pattern of the service provider of this trunk.
Trunks following the same billing pattern should be assigned the same Pulse Rate Option.
2. Define the Call Cost Calculation Schedule for the Pulse Rate option you selected.
The Pulse Rates offered by service providers may vary according to the time of the day. In such cases, you
must divide the day into Time Zones, from 1 to 4, as required, to match the time of the pulse rates offered
by your service provider.
Specify the Start Time and the End time (in 24 hours, HH:MM format) for the Time Zone Index in which
the particular Pulse Rate will be applied.
The default Time Schedule (start and end time) for each Time Zone Index is as follows:
T1 00:00 23:59
T2 00:00 23:59
T3 00:00 23:59
T4 00:00 23:59
If your service provider offers the same Pulse Rate for calling a number for the entire day, there is no need
to change the default Time Schedule. The system will follow Time Zone 1.
Call Budget
1. To enable call budget on the mobile trunk, set Use Call Budget? to Yes. Default: No.
• Define the call budget in terms of number of calls in the field Restrict outgoing calls if total number
of calls exceeds. Default: 9999 (in local currency)
• Define the call budget in terms of amount in the field Restrict outgoing calls if total cost of the calls
exceeds. Default: 999999 (local currency).
• To have the system reset the Number of Calls/Amount/Minutes automatically on a particular date,
select the desired date from the Reset Call Budget on combo box as 1st to 31st of the every month.
The consumed Call Budget Amount/Minutes/Number of Calls can be reset from SE and SA Mode, referred
to as Manual Reset. Refer the feature description “Call Budget on Trunk”.
Call Back
1. Click [+] Call Back to enable the ‘Call Back on Trunk Port’ feature on the mobile trunk. See “Call Back on
Trunk Ports” to know more about this feature.
• Select a Call Back Profile, 1 to 4, for the trunk port from the combo box.
Make sure you also configure the Call Back profile you select here.
Gain Settings
• To increase or decrease the Transmit volume level of the mobile port, set the Tx Gain. Default: 3.
• To increase or decrease the Receive volume level of the mobile port, set the Rx Gain. Default: 3.
If you change the Tx or Rx Gain during an active call, the change you made will not apply on the current
call. It will be applied on the next call.
• Set the Pause Time between successive DTMF digits. Default: 1 second.
The Pause Time provides the delay in number dialing from the mobile port. The Pause Time will be
applicable when the digit 'P' is configured in the DTMF number string which is to be out dialed as DTMF
digits on the Mobile port.
For example, if PPP2 is to be out dialed and Pause timer is configured as 3 seconds, the ETERNITY
will out dial the digit 2 after 9 seconds, i.e. after a delay of individual P i.e. 3+3+3 =9. The range of this
time is from 1 to 9. The Pause Time is used in the “Multi-Stage Dialing” feature.
More Features
Click More... link to expand.
Priority
1. Select a Priority level from 1 to 9 for the trunk. Default: 9.
Each trunk of the ETERNITY NE can be assigned a Priority Level starting from 1, 2, 3... to 9, with '1' being
lowest Priority and '9' being highest Priority. Whenever there are incoming calls on multiple trunks, the call
on the trunk with higher priority will be answered by the Operator extension first. To know more, see
“Priority”.
Time Table
1. Select a Time Table for the Mobile trunk. Default: System Time Table.
Send CLI
1. To hide the subscriber number of the mobile port from being displayed to the called party, select the check
box Do not send CLI for all Outgoing Calls.
This feature will work only if subscribed/supported by your mobile service provider.
Anonymous Calls
1. To allow incoming calls without CLI to land on the mobile port, select the check box Accept Incoming
Calls without CLI.
Debug
1. To initiate debug for the mobile port, select the check box Enable Debug. Default: Disabled.
When you enable this flag, the system will prompt extension users to dial the Account Code whenever
they grab this trunk to dial out a number. The system will allow extension users to dial out numbers only
after they have dialed the Account Code or Name.
Account Codes feature must also be enabled in the Class of Service of extension users who are to be
allowed this feature. See “Account Codes” to know more about this feature.
Now that you have finished configuring the Mobile trunk. You may configure the next Mobile trunk.
The Copy dialog box opens. The dialog box displays the number and name of the trunk to be copied in the
Copy from: field.
The dialog box also lists the numbers and names of the trunks to which you can copy the values under
Copy To:
2. Select the check boxes of the trunk numbers to which you want to Copy To.
• If you have enabled On-site Configuration, this page will appear if your system has detected the
presence of the VoIP module.
• If On-site Configuration is disabled, you will reach this page if you have set the flag 'Is VoIP used?’ to
‘Yes’ on the “Pre-requisites” page.
On this page,
More: Click this button to view all parameter links on the page.
Less: Click this button to view only the essential parameter links on the page.
Expand: Click to expand a link to display all parameters under the link.
Collapse: Click to collapse a link. Hides all parameters under the link.
VoIP Network
Quality of Service
Quality of Service (QoS) refers to priority of IP packets on network layer. QoS is configured for both signaling (SIP)
and media (RTP). ETERNITY sends all the SIP signaling messages and the RTP packets according to the QoS
settings. To configure QoS for SIP (signaling) and RTP (media):
1. Define the priority bits for SIP messages in the field SIP Diffserve/ToS. Valid range is 00 to 63. Default:
26.
2. Define the priority bits for RTP packets in the field RTP DiffServe/ToS. Valid range is 00 to 63. Default: 46.
You can have any matured incoming or outgoing call to be disconnected automatically, if silence (No RTP Packets)
is detected for more than a specified duration of time. To use this feature,
2. Define Detection Time (sec). This Timer defines the duration for which silence must be detected
continuously, for the system to consider it as silence detection and disconnect the call. The valid range of
this Timer is from 001 to 999 seconds. By default, it is set to 999 seconds.
3. Define the number of voice channels you would like to reserve for SIP trunk calls in the field Channel
Reservation for SIP Trunks. Default: 0.
ETERNITY NE supports 8 voice channels, which can be used by SIP Extensions and SIP trunks.
It may happen that SIP Extension users use up most of the voice channels, leaving too few or none for
making/receiving SIP Trunk calls.
This can be avoided by reserving some voice channels exclusively for SIP trunk calls.
This parameter is to be configured if you want to support reliable transmission of (SIP) provisional responses and to
use PRACK (Provisional Acknowledgement).
1. To enable SIP 100rel, select the check box Use 100rel. Default: disabled.
ETERNITY NE supports transporting of SIP messages over User Datagram Protocol (UDP) as well as Transfer
Control Protocol (TCP) connection. Despite the advantages that SIP over TCP offers, it is more common to use
UDP to transport SIP messages. To receive SIP messages over TCP,
1. Select the check box Enable SIP over TCP. Default: Enabled.
To be able to send SIP messages over TCP, you must configure 'TCP' or 'TCP (Fallback to UDP)'.
SIP/RTP Ports
This port defines the port on which the ETERNITY listens for SIP messages transported over UDP. This
port is also used as the source port for sending SIP messages to the remote peer. The valid range for this
port is 1024-65535. Default: 05060.
3. Enter the RTP Listening Port address. This port defines the port on which the VoIP Port of ETERNITY
listens for RTP Packets. This port is also used as the source port for sending RTP packets to the remote
peer. The valid range for this port is 1024-65278. The default RTP Listening Port is 08000.
This parameter is to be configured when the ETERNITY (Ethernet-WAN) is connected behind a NAT router61 and
SIP messages are transported over UDP. UDP NAT Keep Alive messages must be sent to refresh the UDP binding
in the NAT router.
1. Select the check box Enable UDP NAT Keep Alive to send UDP NAT Keep Alive messages periodically to
refresh the binding in the NAT router. Default: disabled.
2. Define the Interval (sec) for sending the UDP NAT Keep Alive messages. This time period should be less
than the UDP Binding Timer of the router. The valid range is 001-999 seconds. Default:120 seconds.
3. Select the Type of Message to be sent when UDP NAT Keep Alive is enabled. Select either REGISTER or
NOTIFY. Default: NOTIFY.
This parameter is to configured when ETERNITY (Ethernet-WAN) is connected behind a NAT router and SIP
messages are transported over TCP. TCP NAT Keep Alive messages must be sent to refresh the TCP binding in
the NAT router.
1. Select the check box Enable TCP NAT Keep Alive to send TCP NAT Keep Alive messages periodically to
refresh the binding in the NAT router. Default: disabled.
2. Define the Interval for sending TCP NAT Keep Alive messages. This time period should be less than the
Binding Timer of the router. The valid range is 001-999 seconds. By default it is set to 120 seconds.
61. Network Address Traversal (NAT) allows multiple hosts in the network to share the single public routable IP address. Means all the
hosts in the private network shall be identified by single public IP address in the global IP cloud.
1. Set the SIP Invite Timer. This is the time in seconds that ETERNITY waits for a response from the called
party after ending the INVITE message. This timer starts after sending INVITE message to the called party
and stops on receipt of the provisional response or the final response or when the user disconnects the
call. On expiry of the timer, the call process is terminated by the ETERNITY and an error tone is played to
the user. The range of the SIP Invite Timer is 10-180 seconds. Default: 30 seconds.
2. Set the SIP Provisional Timer. This is the time in seconds that the ETERNITY waits for the final response
after receiving the provisional response from the called party. This timer starts on the receipt of the
provisional response from the called party and stops on receipt of the final response from the called party
or when the user disconnects the call. On expiry of the timer, ETERNITY terminates the call process and
plays error tone to the user. The range of SIP provisional Timer is 10-180 seconds. Default: 60 seconds.
3. Set the General Request Timer to the desired duration. This is the time in seconds for which ETERNITY
waits for response for a transaction request. This timer starts on the initiation of a transaction. This timer
stops on the receipt of a response for the request. On expiry of the timer, ETERNITY clears the
transaction. This timer is used for Registration request. The range of the General Request Timer is 10-60
seconds. Default: 20 seconds.
If you are connecting SIP Extensions to ETERNITY NE, configure the following general parameters for SIP
extensions:
1. Use SIP Port fetched using STUN: If ETERNITY is located behind the NAT router and you have not
configured Port Forwarding on the Router for SIP, you must enable this.
2. Select the Source Port IP Address. This parameter specifies the NAT Traversal mechanism for SIP
messages. You can select any of these options:
• Use VoIP Ethernet Port IP Address: Select this parameter if your ETERNITY is directly connected to
the public IP network (not behind a NAT Router).
• Use IP Address fetched using STUN: Select this parameter if your ETERNITY is located behind a
NAT Router, and you have configured 'Use IP Address fetched using STUN' in the “Network
Parameters”.
3. Set the Maximum Registration Timer (sec) to the desired value. This is the maximum expiry timer, which
ETERNITY will accept in the REGISTER request received. If the system receives more than this value, the
configured value will be sent in the SIP message. The same Timer is used to handle SUBSCRIBE
requests. The valid range of this timer is from10 to 99999 seconds. Default: 180 seconds.
4. Set the Minimum Registration Timer (sec) to the desired value. This is the minimum expiry timer, which
the User Agent should send in its REGISTER request. If the expiry value in the REGISTER message is
less than this value, the request will be rejected. The valid range of this timer is from 10 to 99999 seconds.
Default: 180 seconds.
5. Enter the Private Key. It may consist of a maximum of 24 characters (all ASCII characters allowed).
Default, the field is blank.
Private key is a security mechanism used by ETERNITY NE to authenticate SIP messages. It is the MD5
authentication key used by the system to encrypt/decrypt the SIP messages.
The ETERNITY NE supports 6 SIP Trunks, depending on the variant. On this page, the number of SIP Trunks you
have specified on the “Pre-requisites” page will be displayed.
• You will reach this page only if you have specified the Number of SIP Trunks Used on the 'System Pre-
requisites' page.
• SIP Trunks are to be configured only if you are using Internet Telephony Service Providers for VoIP
calls.
On this page,
More: Click this button to view all parameter links on the page.
Less: Click this button to view only the essential parameter links on the page.
Expand: Click to expand a link to display all parameters under the link.
Collapse: Click to collapse a link. Hides all parameters under the link.
To configure another SIP trunk, click the Trunk number (name) tab.
To copy the same SIP trunk parameter values to the other SIP trunks, use the Copy button.
If you assigned names to the SIP trunks on the “Naming Trunks” page, the same names will appear on the tabs.
For example, if you have named SIP 1 as GlobalTalk, SIP 2 as Pulver, these names will appear in place of SIP
Trunk 1 and SIP Trunk 2.
If you want to change the name of the SIP trunk, you must go back to the “Naming Trunks” page.
Click the desired SIP trunk number/name tab to configure the parameters.
SIP
1. Select the check box Enable SIP Trunk to activate the SIP trunk.
2. Set the SIP Trunk Mode as Proxy or Peer-to-Peer, according to your requirement.
If you are using the services of an Internet Telephony Service Provider (ITSP), select Proxy to register this
SIP trunk with the ITSP.
If you are not using this service, select Peer-to-Peer, and configure the Peer-to-Peer Table. To do this,
• Click the arrow icon.
• The Peer-to-Peer Table page will open in a new window.
• Configure this table.
• Click Submit to save your entries.
• Close the window.
This is the ID which remote parties will use to call this SIP Trunk. The SIP ID may be a number or text
consisting of a maximum of 40 characters.
If you have defined the trunk mode as Proxy, enter the User ID provided by your ITSP. For example, if SIP
URI provided by the ITSP is [email protected], enter 12345 in this field.
If you have defined the trunk mode as Peer-to-Peer, enter the desired User ID.
4. If you selected Peer-to-Peer as the SIP trunk mode, you may configure the trunk to Treat Incoming Peer-
to-Peer call as: Trunk or Station.
If you select Trunk, the Route Incoming call routing configured for the SIP trunk will be applied.
If you select Station, the incoming call on the trunk will be routed to the Extension Number received in the
“To:” field in the INVITE message.
If a Trunk Access Code and External Number are received in the ‘To:” field of the INVITE message, the
call will be dialed out using the outgoing trunks you configure in Select Trunks for Outgoing Calls.
If you select Station, you may also configure the parameters Class of Service, Caller ID on Call
Transfer, Toll Control and Select Trunks for Outgoing Calls.
Proxy/Registrar Parameters
If you have defined the SIP trunk mode as Proxy, configure the Proxy/Registrar Server parameters.
1. Enter the Proxy/Registrar Server Address and the Registrar Server Port provided by your ITSP.
The Registrar Server Listening Port ranges from 1024 to 65535. Default: 05060.
2. Enter the authentication ID (user ID) provided by your ITSP in the field Authentication ID.
3. Enter the password provided by your ITSP for the Authentication ID in the field Authentication Password.
Outbound Proxy
If your ITSP (for this SIP trunk) has a SIP outbound server to handle voice calls, configure outbound proxy settings.
2. Enter the IP address or domain name of the outbound proxy server and the server port address in the field
Outbound Proxy Server Address: Port. The default server port address is 05060.
3. Set the Re-registration Timer. This is the time period after which the ETERNITY NE will send registration
request to maintain registration binding with the Registrar server62.
62. The Registrar Server deletes an entry of its client from its database on expiry of a fixed timer. This timer is set by the Registrar
Server. The Re-registration Timer of ETERNITY enables it to send a registration request before the Timer of the Registrar Server
expires to remain registered on the Server.
4. Define the Registration Retry Time. This Timer stands for the period between retries for registration with
the Registrar Server. If the registration attempt fails, ETERNITY sends the registration request on the
expiry of this Timer again. The system continues to send the registration request till it gets registered with
the Registrar Server. The valid range of this timer is from 00001- 65535. Default: 10 seconds.
2. Select the landing destination to Route Calls during Day to. You may select:
• Operator
• Extensions
• Built-in Auto Attendant
• Voice Mail Auto Attendant (if available).
• If you select Extensions as the landing destination, select the landing destination extensions in the
corresponding field.
• Select the extension numbers as landing destination. These may be SLT, DKP or SIP extensions.
• Set the Ring Timer for the extensions. This timer defines the time for which the extension, on which
the call lands, should ring. Default: 015 seconds.
• Set Continuous Ring63 for the extensions, if you want the extensions to ring till the incoming call is
answered. Default: Disabled.
• Enable Rotation64, if you have selected more than one extension. Default: Disabled.
• Click OK. All the extensions you selected will appear in the Extension field, in the sequence you
selected, separated by commas.
• To route the incoming call to an auto attendant, if the selected extensions do not answer the call,
63. When Continuous Ring is selected, the first extension in the group you have created will continue to ring, even as the system hunts
for other extensions in the group to land the call. If the call still remains unanswered, the system will return the call to the first exten-
sion once again. This flag has no relevance, if you select only one landing extension.
64. When Rotation is enabled, each new call lands on the extension next to the one that received the call last. This ensures equal dis-
tribution of incoming calls to all the destination extensions in this group.
• Select the destination for the unanswered call in the field If not answered, route to. You may
select the Built-In Auto Attendant or the Voice Mail Auto Attendant. Default: Built-In Auto
Attendant.
• If you select Built-in Auto Attendant as the landing destination for calls during the day,
• the Voice Modules 02 to 13 will be played as “DID Greeting Messages” and “DID Guidance
Messages”.
• If you want to play a different message, make sure you record the desired message(s) in the Voice
Modules after completing the installation with Basic Settings. Refer the topic “Voice Message
Applications” to know more.
3. To enable “Direct Inward System Access (DISA)” on the SIP trunk, select the check box Activate Direct
Inward System Access. Default: Disabled.
• PIN Authentication–Multiple calls: Select this option if you want to enable DISA with PIN
Authentication and allow multiple calls during the DISA login session.
• CLI Authentication–Multiple calls: Select this option if you want to enable DISA login with CLI
Authentication and allow multiple calls to be made during the DISA login session.
• CLI Authentication–One call: Select this option if you want to enable DISA session with CLI
Authentication, and allow only a single call to be made during the DISA login session.
If you select Multiple Calls or CLI Authentication One Call as CLI Authentication, you must configure the
DISA CLI-Authentication Table under Advanced Settings link. See “Direct Inward System Access (DISA)”
4. To enable “Trunk Auto Answer”65 feature on the trunk, select the type of Trunk Auto Answer you want:
• For all Calls: the system will answer all incoming calls landing on the trunk line.
• When Busy: the system will answer incoming calls on the trunk, if the landing destination is busy.
• Select the Trunk Auto Answer Greeting Message you want the system to play when greeting the
callers, from the following options:
• Play Music on Hold:
• Play Greeting Message1:
65. Trunk Auto Answer enables calls landing on a trunk to be answered automatically by greeting the caller with a voice message
before the call is actually handled.
When you select a Greeting Message, you must record a Voice Module with the desired Greeting
Message and assign the Voice Module to Trunk Auto Answer Greeting application.
You can record 4 different Greetings Messages for Trunk Auto Answer and assign a different Greeting
message for the Day and Night. See the topic “Voice Message Applications” under Advanced Settings link
for instructions on recording and assigning voice modules to greeting messages.
• Select the Trunk Auto Answer RBT Message you want the system to play to callers from the
following options:
• Play Music on Hold
• Play RBT Message1
• Play RBT Message2
• Play RBT Message3
• Play RBT Message4
• If you do not want RBT message to be played, select Do Not Play. The system will play Ring Back
Tone to the caller.
When you select an RBT Message (1–4), you must record a Voice Module with the desired RBT Message
and assign the Voice Module to Trunk Auto Answer RBT Message application
You can record 4 different RBT Messages for Trunk Auto Answer and assign a different RBT message for
the Day and Night. See “Voice Message Applications” under Advanced Settings link for instructions on
recording and assigning voice modules for RBT messages.
• Select the Trunk Auto Answer Busy Bye Message you want the system to play to callers when the
landing destination extension is busy, from the following options:
• Play Music on Hold
• Play Bye Message1
• Play Bye Message2
• Play Bye Message3
• Play Bye Message4
If you do not want Busy Bye message to be played, select Do Not Play. The system will play Busy
Tone to the caller.
When you select a Bye Message (1–4), you must record a Voice Module with the desired Bye Message
and assign the Voice Module to Trunk Auto Answer Bye Message application.
You can record 4 different Bye Messages for Trunk Auto Answer and assign a different Bye message for
the Day and Night. See “Voice Message Applications” under Advanced Settings link for instructions on
recording and assigning voice modules for Bye messages.
Follow the same instructions as described in the previous step for selecting landing destination for the Day.
When RCOC is applied on the SIP trunk, the system routes calls returned by remote parties back to the extensions
that originally made the call from this port (i.e. the original callers' extensions). To know more, refer the feature
description for “RCOC (Return Call to Original Caller)”.
1. To apply the RCOC feature on the SIP trunk, select the check box Set RCOC for Calling Extension, when
Called Number is busy/ switched-off / not responding. Default: Disabled.
If you have selected Voice Mail Auto Attendant as the destination for Routing calls during the day or during night,
2. Select the desired Voice Mail Auto Attendant Profile number for the trunk, from 1 to 4. Default: 1.
The Voice Mail System of ETERNITY NE answers calls and processes them according to the Voice Mail
Auto Attendant Profile assigned to the trunk. So, you must configure the Voice Mail Auto Attendant Profile
number you have selected for the trunk.
3. To configure the Voice Mail Auto Attendant Profile, click settings icon.
A new window opens.
Configure the Voice Mail Auto Attendant Profile. For detailed instructions, see “Voice Mail Auto Attendant
Profile”. Close the window.
DDI Routing
1. Enter the DDI Pilot Number provided by the ITSP with whom you have registered this SIP Trunk.
The Automatic Number Translation feature of ETERNITY modifies dialed numbers or part thereof to match the
specific route numbering plan understood by the destination network—PSTN, GSM, VoIP—by adding or stripping
country and area codes.
For example, you can configure Automatic Number Translation such that when an extension user dials a local
landline number, the ETERNITY prefixes the number with the appropriate country-area code when it routes the call
through the GSM network. To know more about this feature, see the description for “Automatic Number
Translation”.
2. Select the desired ANT Table number, table 1 to 8, which you want to apply on the trunk. Default: OFF.
The table has two columns: one for Dialed Number Strings and the other for the corresponding Substitute
Number strings. The number strings are stored against an Index number, from 1 to 32. You can to enter as
many as 32 number strings.
6. Enter the Dialed Number strings and their corresponding Substitute Number strings in the respective
columns, against the same Index numbers. For example, if '95' is entered as the Dialed Number at Index
01 and its Substitute Number '91' must be entered at Index 01.
SMDR Storage
The Station Message Detail Recording (SMDR) feature of ETERNITY enables you to record the details of Internal,
Incoming (IC) and Outgoing (OG) calls made from/to all its trunks and extensions. To obtain SMDR as a report, you
must enable SMDR Storage, and set filters. See “Station Message Detail Recording (SMDR)” and “Station
Message Detail Recording–Storage” to know more.
2. To store details of incoming calls landing on this SIP trunk, select the check box Store Incoming Calls.
Default: selected.
3. To store details of outgoing calls made from this SIP trunk, select the check box Store Outgoing Calls.
Default: selected.
4. Click Submit.
This parameter determines what the system should do when an external called party in speech on this trunk is put
on Hold by an extension user by pressing a DSS key to dial another port.
For example, the DKP extension user on DKP1 port is in the middle of speech with an external party on a CO2.
If the extension user of DKP1 presses a DSS key to call another extension port, DKP2, two situations are possible,
depending on the DSS Key Interface you configure:
i. The external caller on CO2 will be played music-on-hold, and the user on DKP1 will hear Ring Back
Tone. The call will be placed on DKP2.
ii. The external caller on CO2 will be disconnected. DKP1 user will hear Ring Back Tone. The call will be
placed on DKP2.
2. Select the radio button of the desired DSS Key Interface option:
• Disconnect current call when DKP user presses DSS key of other trunk, while in speech using
this trunk.
• Put current call on Hold, when DKP user presses DSS key of other trunk, while in speech using
this trunk. (Default).
3. Click Submit.
Cost Factor
Cost Factor is used for grading the cost of routing calls from a trunk for the purpose of least cost routing. See “Least
Cost Routing (LCR)”
2. Select a Cost Factor for this trunk from the drop down list, from 01 to 99. Default: 01.
Each Call Cost Calculation Pulse Rate option contains a Pulse Rate Type, which you must configure.
Some service providers offer discounted rates for holidays. In which case, you can configure the rates for
normal days in the Normal Pulse Rate Table and the rates for holidays in the Holiday Pulse Rate Table.
See “Call Cost Calculation (CCC)”
Configure the Pulse Rate Type according to the billing pattern of the service provider of this trunk.
Trunks following the same billing pattern should be assigned the same Pulse Rate Option.
2. Define the Call Cost Calculation Schedule for the Pulse Rate option you selected.
The Pulse Rates offered by service providers may vary according to the time of the day. In such cases, you
must divide the day into Time Zones, from 1 to 4, as required, to match the time of the pulse rates offered
by your service provider.
Specify the Start Time and the End time (in 24 hours, HH:MM format) for the Time Zone Index in which
the particular Pulse Rate will be applied.
T1 00:00 23:59
T2 00:00 23:59
T3 00:00 23:59
T4 00:00 23:59
If your service provider offers the same Pulse Rate for calling a number for the entire day, there is no need
to change the default Time Schedule. The system will follow Time Zone 1.
Call Budget
Configure this parameter if you want to control the cost of phone calls made from this trunk. See “Call Budget on
Trunk” to know more.
1. To enable call budget on the trunk, set Use Call Budget? to Yes. Default: No.
2. Define the call budget in terms of number of calls in the field Restrict outgoing calls if total number of
calls exceeds. Default: 9999 (in local currency)
3. Define the call budget in terms of amount in the field Restrict outgoing calls if total cost of the calls
exceeds. Default: 999999 (local currency).
4. Define the call budget in terms of duration in the field Restrict outgoing calls if total call duration
exceeds. Default: 999999 minutes
5. To have the system reset the Number of Calls/Amount/Minutes automatically on a particular date, select
the desired date from the Reset Call Budget on combo box as 1st to 31st of the every month.
The consumed Call Budget Amount/Minutes/Number of Calls can be reset from SE and SA Mode, referred
to as Manual Reset. Refer the feature description “Call Budget on Trunk”.
Configure this parameter if you want to enable the ‘Call Back on Trunk Port’ feature on the trunk. See “Call Back on
Trunk Ports” to know more.
1. Select a Call Back Profile, 1 to 4, for the trunk port from the combo box.
Make sure you also configure the Call Back profile you select here.
1. Set the time for which the DTMF digit should remain ON while being dialed out by the ETERNITY NE as
DTMF ON Time.
The range of this timer is from 051 to 255 milliseconds. Default: 102 msec.
2. Set the time for which ETERNITY should wait before dialing the successive DTMF digits as Inter Digit
Pause Time. The range of this timer is from 051 to 255 milliseconds. Default: 102 msec.
3. Set the time for ----- as Pause Time (sec). The range of this timer is from 1 to 9 seconds. Default: 1
second.
• Pause Timer (sec): This parameter is used for the “Multi-Stage Dialing” feature. This Timer is required for
inserting delay while digits of a number string are out dialed from the SIP trunk. The Pause Timer will be
applicable when the letter 'P' is configured in the DTMF number string which is to be out dialed as DTMF
digits on the SIP trunk. The range of this timer is from 1 to 9 seconds. By default the Timer is set to 3
seconds.
For example, if 'PPP7' is to be out dialed and Pause timer is programmed as 3 seconds, the ETERNITY
will out dial the digit 7 after 9 seconds, i.e. after a delay of individual P (i.e. 3+3+3 =9). The range of this
Timer is from 1 to 9.
1. Define the CLI of the SIP Trunk to be sent to the remote party on outgoing calls made using this SIP trunk.
You may select any of the following Send CLI options as desired:
• CLIR: Select this option if you do not want the CLI to be sent.
• SIP ID: You may select this option if you want the SIP ID programmed on the SIP Trunk to be sent as
CLI.
• Calling Party-wise: Select this option if you want to send the Calling Extension Number (i.e. the
number of the extension making the outgoing call through the SIP trunk) as CLI.
The DDI number assigned to the calling extension will be sent, instead of its extension number.
If the calling extension has disabled the parameter ‘Send DDI as CLI’, then its Pilot number configured
in the DDI Routing on this SIP Trunk will be sent as CLI.
If calling extension has enabled CLIR, no CLI will be sent by this SIP Trunk.
• Fixed Number: Select this option if you want a specific number to be sent as CLI. When you select this
option, you must also define the number to be sent as CLI.
You may select this option you want to send any of your trunk line numbers as CLI on the SIP Trunk so
as to enable the called party to call back the calling party using this CLI.
Since it is not possible to call back a SIP ID, Fixed Number offers you a solution, using which you can
send a trunk line number as CLI on the SIP Trunk. Using this CLI, the called party can call back the
calling party.
The Fixed Number may consist of a maximum of 40 characters, including all ASCII characters.
By default, ‘SIP ID is set as the Send CLI option for all SIP Trunks.
When extension number of the calling extension is blank, and the ‘Send CLI Option’ programmed for the
the SIP Trunk is other than "SIP ID", then also SIP ID will be sent as CLI.
Vocoders
2. Select Vocoders66 in the order of preference from the drop down box.
The Vocoders supported by ETERNITY in the order of preference, i.e. 1st to 7th, by default are:
• G.723
• G.729 AB
• GSM FR
• iLBC - 30 ms
• iLBC - 20 ms
• G.711 Law
• G.711 A - Law
If you do not want to select any Vocoder, you can select the option 'None'. However, if all Vocoder
Preferences from 1 to 7 are set to 'None', incoming and outgoing calls will be blocked.
3. If you selected G.723 codec as a Preferred Vocoder, select G.723 Bit Rate as: Bit Rate: 5.3 Kbps or 6.3
Kbps. Default: 6.3kpbs.
When G.723 is negotiated, the selected Bit Rate will be applied only when sending the RTP packets.
When receiving RTP packets from the remote end, both Bit Rates of G.723 will be accepted.
4. Select the check box Silence Suppression to suppress the 'Silence' packets, and to allow only the Voice
packets through. ETERNITY supports Silence Suppression for all Vocoders except GSM FR.
The DTMF option you select will determine how the DTMF digits will be sent over the IP Network, when a
DTMF digit is pressed.
66. Vocoders are the various voice codecs used to compress the data in RTP packets for optimum use of bandwidth and for ensuring
voice quality. You can set 7 Vocoder options in the order of preference.
Echo Cancellation
ETERNITY supports Echo Cancellation for SIP to CO trunk calls and SIP to Digital Trunks (Mobile, SIP) and
Extensions (DKP, SIP). If you want to apply Echo Cancellation,
2. Set the tail length for echo cancellation for SIP to CO trunks as Tail Length (msec) for CO. The echo
cancellation Tail Length for SIP to CO trunks can be 32/64/128 milliseconds. Default: 128 milliseconds.
3. Set the tail length for echo cancellation for SIP to Mobile/SIP trunks and SIP to DKP/SIP extensions as
Tail Length (msec) for Extensions/Digital Trunks (msec). The echo cancellation Tail Length for SIP to
Digital Trunks/Extensions can be 32/64/128 milliseconds. Default: 32 milliseconds.
Jitter Buffer
The speed at which voice packets travel through a network depends on the condition of the network. All voice
packets may not come at the same speed. This variation in the delay in receiving packets, known as Jitter, affects
voice quality. Jitter Buffer helps overcome the delay in receiving voice packets and improves voice quality. Jitter
Buffer receives voice packets, stores them and sends it to the DSP to process it at evenly spaced intervals, thus
improving voice quality.
ETERNITY supports two types of Jitter Buffer: Static and Dynamic. Static Jitter Buffer's internal delay is static,
whereas, the Dynamic Jitter Buffer's internal delay adapts itself to the jitter in the network.
If you select Static Jitter Buffer, configure the 'Minimum Delay'. The value configured in the Minimum
Delay determines the size of the Static Jitter buffer.
If you select Dynamic Jitter Buffer, configure the 'Optimization Factor' and 'Minimum Delay'. The
minimum size of the Dynamic Jitter buffer depends on the 'Minimum Delay' configured. The Optimization
Factor determines the rate of adaptation of the Dynamic Jitter Buffer to the jitter in the network.
The actual size of the Dynamic Jitter Buffer will be determined by the DSP on the basis of the Optimization
Factor configured and actual network condition. Dynamic Jitter buffer can go up to maximum 300
milliseconds.
3. Set the Minimum Delay (msec) (for both Static and Dynamic Jitter Buffer) to the desired value from 10 to
280 milliseconds. Default: 10.
The Minimum Delay determines the size of the Static Jitter Buffer. When Jitter Buffer type is selected as
Static, the Minimum Delay defines the size of the Static Jitter Buffer. The Static Jitter Buffer will store each
received voice packet for the configured time and then it will send it to DSP for voice processing.
When Jitter Buffer type is Dynamic, the Minimum Delay specifies the minimum time for which the Dynamic
Jitter Buffer will store the received voice packet before sending it to the DSP for voice processing.
4. You can send/receive Fax over IP (FoIP) from a Fax machine connected to the SLT port of the ETERNITY
NE.
• Select as Fax Type, the protocol of Fax over IP.
The ETERNITY NE supports the fax options: T.38 (UDPTL), T.38 (RTP), and Pass Through.
• 'Pass Through' and 'T.38' will work only if the peer devices also support the same option.
• If you select 'Pass through' as Fax type, you must disable 'Silence Suppression'.
• If the fax sent using T.38 is rejected, ETERNITY will use Pass Through for sending the Fax.
T.38 Parameters
If you selected T.38 protocol as Fax Type, you may configure Eye Quality Monitor (EQM) related parameters to
improve quality of Fax. The higher the EQM, the better the Fax quality. To improve the quality of T.38 fax reception,
1. Set the Max Rate (Kbps) to the desired speed. Default: 14.4 kbps.
The Max Rate controls the Fax image transfer speed. As EQM is inversely proportional to fax Max Rate, if
you receive poor quality fax, the Max Rate should be reduced.
2. Set the Packet Period (msec) to the desired value. Default: 40msec.
The packet period sets the sampling rate of TDM signal. If you cannot improve fax quality by lowering Fax
Max Rate, you may reduce the Fax Packet Period.
3. Set the Image Redundancy Level to the desired value. The Fax Image Level is redundancy level for
output Image, which can be from 0 to 3. Default: 1.
Fax Image transfer speed is inversely proportional to this parameter. If this parameter is low then fax is
transferred faster. EQM is directly proportional to this parameter. If this parameter is high, good quality fax
can be achieved.
You may increase the Fax Image Level from 1 to 3 if the quality of fax does not improve with Fax Max Rate
and Fax Packet Period.
4. Set the Data Redundancy Level to the desired value. Default: 3. This is a redundancy level for T.30
control data. Fax Data Level can be set from 0 to 7. Level 0 means no redundancy. Redundancy level
increases from 1 towards 7. The higher the level set, the slower would be the fax transmission.
EQM is directly proportional to this parameter. The higher the Fax Data Redundancy Level, the better the
EQM.
Gain Settings
• To increase the level of incoming speech volume of calls from this SIP trunk to Mobile trunks/DKP
extensions, set the Rx Gain for SIP to Digital Trunk/DKP Voice call (dB) to the desired level from -31 to
31dB. Default:0.
• To increase the level of outgoing speech volume of calls from this SIP trunk to Mobile trunks/DKP
extensions, set the Tx Gain for SIP to Digital Trunk/DKP Voice call (dB) to the desired level from -31 to
31 dB. Default: 0.
• You can also set the Rx and Tx Gains for SIP to Analog Trunks and Extensions (CO and SLT).
• To increase Rx and Tx Gain for SIP to CO trunks, go to the "CO Trunks” page of Basic Settings.
• To increase Rx and Tx Gain for SIP to SLT extensions, go to “SLT” parameters page of Basic Settings.
Configure this parameter if you selected Pass Through67 as Fax Type, and the fax is to be received on a digital
trunk.
1. Select the dB level for Data Gain (dB) from-31 to 31 dB. Default: -11 dB.
2. Select the dB Level for Bypass Gain (dB) from -31 to 31 dB. Default: -9 dB.
Configure this parameter if Pass Through Fax is to be received on a fax machine connected to an SLT port.
1. Select the dB level for Data Gain (dB) from -31 to 31 dB. Default: -11 dB.
2. Select the dB level for Bypass Gain (dB) from -31 to 31 dB. Default: -9 dB.
• Select Use Ethernet Port IP Address if ETERNITY NE is connected directly to the public internet.
• Select Use IP Address Fetched Using STUN if ETERNITY NE is located behind a NAT router
other than Symmetric.
• Select Use Router's Public IP Address, if ETERNITY NE is located behind a NAT Router (any
type).
Simultaneous Calls
2. In the field No. of Simultaneous Calls select from the combo box, the number of simultaneous calls you
want to allow on this SIP Trunk. Default: 8.
67. Pass Through Fax packets are transported using RTP protocol. Normally, fax calls require less gain compared to voice calls. How-
ever, to improve fax reception, ETERNITY NE enables you to configure gain settings for fax. Fax gain settings consist of Data Gain
and Bypass Gain. ETERNITY supports Fax Receive Gain for SIP to Digital Trunk calls as well as SIP to SLT Calls.
The ETERNITY NE supports 8 simultaneous calls. Ask your ITSP about the number of simultaneous SIP
calls supported on this SIP Trunk. Configure this parameter only if the ITSP supports less than 8
simultaneous calls.
Symmetric RTP
3. The use of Symmetric RTP makes it possible for a SIP device to send the RTP on the same connection on
which it is listening for RTP. This is done only on peer-to-peer SIP trunks.
Select the check box Use Symmetric RTP, if the ETERNITY NE is located on a public IP and you want
outgoing calls to the SIP Client located behind the NAT Router. OR if you need to receive incoming calls
from the SIP Client located behind the NAT router.
Digest Authentication
4. The Digest Authentication feature allows incoming calls from callers only after the callers have
authenticated themselves (with their User ID and Password). If a caller enters invalid authentication
information, the system will re-challenge authentication once and reject the call, if the authentication
attempt fails again.
If you want to enable the 'Digest Authentication' feature on the SIP trunk,
• TCP (Fallback to UDP): TCP is used for outgoing messages. However, if the TCP connection fails, the
system will attempt to send the message again over UDP.
Default: UDP
Clear the check box if you do not want REGISTER Message to be sent.
If you set the SIP Trunk mode as Peer-to-Peer and selected Treat Incoming Peer-to-Peer Call as Station, you
may configure Caller ID on Call Transfer.
You can choose whether the system should display the CLI of the ‘Held Party’ or the CLI of the ‘Transferring Party’
to the transfer destination extension while the call is being transferred.
Select the radio button of the desired option. See “Calling Line Identification and Presentation (CLIP)” to know
more.
Toll Control
If you set the SIP Trunk mode as Peer-to-Peer and selected Treat Incoming Peer-to-Peer Call as Station,
configure Toll Control.
1. Click Toll Control to expand options. Set the desired Toll Control for the SIP trunk for the Day and Night.
2. Select the type of Calls Allowed during Day: All Calls, No Calls, Local Calls, Regional Calls, National
Calls, and Limited Calls 1, 2, 3. Default: All Calls.
3. Select the type of Calls Allowed during Night: All Calls, No Calls, Local Calls, Regional Calls, National
Calls, and Limited Calls 1, 2, 3. Default: All Calls.
4. Select the call privilege for Calls allowed for Lock Level 1, Lock Level 2 and Lock Level 3.
If you set the SIP Trunk mode as Peer-to-Peer and selected Treat Incoming Peer-to-Peer Call as Station, select
the Trunks for Outgoing Calls.
2. Double click the Trunks allowed for dialing field. A multiple selection box opens.
• To select a trunk from the left box, place your cursor on the desired trunk, and click the Select>>
button.
• You may change the sequence of the trunks you selected, if required, using the Up and Down arrow
buttons on the right display box.
• You can also delete trunks from the ones you have selected.
• You may enable Rotation, if you have selected more than one trunk. Default: Disabled. Rotation has
no relevance if only one member trunk is selected.
• Click OK. The multiple selection box closes. All the trunks appear in the field Trunks allowed for ‘0’
dialing, in the sequence you selected, separated by commas.
• To apply Least Cost Routing, if required select the desired LCR method from the combo box, and
configure the settings, as required.
Priority
Each trunk of the ETERNITY NE can be assigned a Priority Level starting from 1, 2, 3... to 9, with '1' being lowest
Priority and '9' being highest Priority. Whenever there are incoming calls on multiple trunks, the call on the trunk
with higher priority will be answered by the Operator extension first. To know more, see “Priority”.
Time Table
Select a Time Table for the SIP trunk. Default: System Time Table.
1. Select the check box Do not allow Outgoing Calls without Account Code to enable Forced Account
code on the trunk. Default: Disabled.
When you enable this flag, the system will prompt extension users to dial the Account Code whenever they
grab this trunk to dial out a number. The system will allow extension users to dial out numbers only after
they have dialed the Account Code or Name.
Account Codes feature must also be enabled in the Class of Service of extension users who are to be
allowed this feature.
Now that you have finished configuring the SIP trunk. You may configure the next SIP trunk.
3. Use the Copy button to apply the same SIP trunk settings you configured on this trunk.
The Copy dialog box opens. The dialog box displays the number and name of the trunk to be copied in the
Copy from: field.
The dialog box also lists the numbers and names of the trunks to which you can copy the values under
Copy To:
2. Select the check boxes of the trunk numbers to which you want to Copy To.
• This page displays the default Emergency Numbers for the 'Region' you selected for the system. In
other words, it displays the Emergency Numbers specific to the country/region where the ETERNITY
NE is installed.
• The default Emergency Numbers that appear on this page are non-editable. All you need to do is to
select the Outgoing Trunks for each default Emergency Number.
For example, '112' is the default Emergency Number for the mobile network. So, you may select the
Mobile Trunk for dialing this number.
• You may add additional Emergency Numbers to the existing default numbers in the edit boxes and
select the Outgoing Trunks for these numbers.
1. To add an Emergency Number, Click the Emergency Numbers link below Basic Settings.
2. Enter the desired number in the field Route the Emergency Number.
• A multiple selection box opens. On the left, trunks appear with their names (if configured in “Naming
Trunks”) and port numbers in a sequence, starting with CO trunks, followed by Mobile trunks and SIP
trunks68.
• To select a trunk, place your cursor on the desired trunk, and click the Select>> button.
Or
• Press the ctrl key and click the left mouse button to select multiple trunks.
Make sure that the trunks selected for each Emergency number route the Emergency call to the right
network.
You may select as many as 10 trunks. You will get an alert, if you select more than 10 trunks or select
the same trunk more than once.
To change the sequence of the trunks you selected, use the Up and Down arrow buttons on the right
display box.
To delete trunks from the ones you have selected use Delete key on your keyboard.
• Enable Rotation if you have selected more than one trunk. Default: Disabled.
When you enable Rotation, each new outgoing call is routed through the subsequent trunk in the
group,69 ensuring equal distribution of outgoing call traffic on all trunks.
When Rotation is OFF, calls are routed through the first trunk in the group. If this trunk is busy, the call
is routed to the next trunk in the group.
68. If you have not assigned any names to the trunks, they will appear with their default names (CO, MOB, SIP) and port numbers. If
you have enabled On-Site Configuration, only those trunks that are connected will appear in the box.
69. The first call through the first trunk, the second through the second, the third through the third trunk, and so forth. Thus each new
call is routed through the trunk next to the one that routed the previous outgoing call.
Preferred WAN
1. Set the Preferred WAN according to the WAN interface you have used for your VoIP installation, which
may be:
• Ethernet WAN
• Wireless WAN over Mobile Port 1 and Mobile Port 2
Default: Ethernet
If you select Ethernet as Preferred WAN, you must configure the parameters: Network Connection Type,
DNS Connection Type, Dynamic DNS, Router’s Public IP Address, and Simple Traversal of UDP through
NAT (STUN).
If you select Wireless WAN Mobile 1 or Mobile 2, configure Wireless WAN parameters.
1. Select the network connection type, i.e. the IP Addressing Scheme used by your network to assign the
WAN IP address to the Ethernet Port: Static, DHCP, PPPoE. Default: Static.
• Static: If your network uses Static IP addresses, select Static and configure the following parameters.
• IP Address: Enter the IP Address you obtained from your network Administrator for the Ethernet
Port of ETERNITY NE in this field. Make sure that the IP Address does not conflict with that of any
other device on the LAN.
• Subnet Mask: Enter the Subnet Mask you obtained from your network Administrator for the
Ethernet Port in this field.
• Default Gateway: Enter the IP Address of the Router’s LAN Interface as the Default Gateway IP
Address.
• DHCP: If your network uses DHCP addressing, the DHCP server will dynamically assign an IP
Address, the Subnet Mask to the Gateway Address to the Ethernet Port whenever ETERNITY NE is
restarted. You have to configure the Domain Name Server (DNS) Address only, if not already provided
by your Internet Service Provider.
• PPPoE: If your network uses PPPoE addressing, the PPPoE server will automatically assign an IP
Address, Subnet Mask and Gateway Address to the Ethernet Port of ETERNITY NE. You need to
configure the following parameters provided by your Internet Service Provider:
• PPPoE User ID: Enter the User Name provided by the Internet Service Provider. The User ID may
be a maximum of 16 characters.
• PPPoE Password: Enter the User Password provided by the Internet Service Provider. The
password may be a maximum of 16 characters.
• PPPoE Service Name: Enter the Service Name, if provided by your Internet Service Provider. The
Service Name may consist of a maximum of 16 characters. If Service Name is not provided, leave
this field blank.
Configure the DNS related parameters as provided by your Internet Service Provider. You may consult your LAN
administrator in this regard.
1. Select DNS Connection Type as Static or Auto according to the Connection Type (IP Addressing
scheme) used by the network.
If your network does not assign DNS Address automatically, set DNS Address Assignment as Static
and enter the DNS Server Address in the DNS IP Address field or the DNS Name provided by your
ISP in the DNS Domain Name field.
• DNS IP Address: This field will be editable only if you selected DNS Address Assignment as 'Static'.
Enter the DNS Server IP Address here. The DNS Address can be a maximum of 15 characters.
If you selected DNS Address Assignment as 'Automatic', the DNS Address assigned by the DHCP/
PPPoE server will appear here.
• DNS Domain Name: Enter DNS Domain Name if provided by your ITSP/LAN Administrator.
Otherwise, keep this field blank. The Domain Name may be a maximum of 40 characters.
Wireless WAN
If you selected Wireless WAN as your preferred WAN, configure the Wireless WAN for the Mobile port you are
using as the WAN interface. Make sure you have Internet Services enabled on the SIM present in the Mobile Port.
1. In the field Access Point, enter the access point provided by your Service Provider.
2. Enter the Dial-up Number provided to you for the internet service by your Service Provider.
4. In the field, Password, enter the authentication password for the User ID provided by your Service
Provider.
Dynamic DNS
ETERNITY NE supports Dynamic DNS Server client of the Service Provider Dynamic DNS.org. Dynamic DNS
(DDNS) is a service that maps internet domain names to IP addresses. Dynamic DNS may be required if you have
registered SIP extensions with the Registrar Server of ETERNITY NE.
When you register IP extensions with ETERNITY NE and your network uses dynamic IP addressing (DHCP or
PPPoE) to assign the WAN IP Address70, all the SIP clients registered with ETERNITY will need to be updated with
the new WAN IP Address to be able to function. Dynamic DNS resolves this mapping a host name (domain name)
(host name) and the preferred WAN IP Address. Each time a new WAN IP Address is assigned by the network,
ETERNITY updates the mapping on Dynamic DNS server.
You may register with DynamicDNS.org server to use this service. If you have registered with Dynamic DNS.org,
do the following:
The DynamicDNS server stores the mapping between hostname and IP Address, which can be updated
periodically. However, if the ETERNITY frequently sends IP Address update requests to the DDNS server,
the server is likely to block the hostname in its database and terminate the DDNS services provided to
ETERNITY.
So, if you restart the ETERNITY frequently, chances are that DDNS server may block the hostname of
your system. This will in turn affect the ability of the system to receive the calls using DDNS hostname
since the entry (mapping between hostname and IP Address) in the DNS server will be deleted in such
scenarios.
ETERNITY NE updates the WAN IP Address in the DDNS server at each Power ON. You can also update
the WAN IP Address at any time you want, as required.
2. Enter the User ID you created on DynDNS.org in the field User ID. A maximum of 40 characters, including
all ASCII characters are allowed.
70. According to your installation scenario and Preferred WAN you configured.
4. Enter the Host Name you created on DynDNS.org. The host name must be under 40 characters.
5. Define the Retry Trials. This is the number of attempts ETERNITY NE will make to send the WAN IP
Address Update Request to the Dynamic DNS Server. The Retry Count may be set from 1 to 9. By default
the count is set to 1.
6. Whenever you want to update the WAN IP Address on the DDNS server, use the button Click to Update
IP Address.
This parameter is of relevance if the Ethernet WAN of ETERNITY NE is located behind a NAT Router and SIP
Messages are to be forwarded to the public internet.
Router's Public IP Address specifies the fixed IP Address of your NAT router required for NAT Traversal in SIP
messages.
You also need to select 'Router's Public IP Address' as the 'Source Port IP Address' in the 'SIP Extension
General Parameters', when you configure SIP Extensions'. Refer the topic "Configuring SIP Extensions".
You can also use STUN as an alternative to the Router's Public IP Address as NAT Traversal mechanism.
Ask your Network Administrator about the NAT Traversal mechanism that suits best for your voice network
and configure this parameter.
This parameter is to be configured only if the Ethernet-WAN of ETERNITY is located behind a NAT Router and SIP
Messages need to be forwarded to the public internet.
STUN specifies the mechanism required for NAT traversal in SIP messages. The STUN Server facilitates
traversing through most NATs, except symmetric NATs.
• Enter the Listening Port of the STUN Server in the field STUN Port. The valid range for this field is from
1024-65535. The default STUN Port is 03478.
2. To allow the SIP Port number to be fetched using STUN in the SIP message, you must enable Use SIP
Port fetched using STUN. Default: enabled.
If you are using Port-Forwarding in the Router for SIP messages, disable Use SIP Port fetched using
STUN by clearing the check box.
You also need to select 'Use IP Address fetched using STUN’ as the 'Source Port IP Address' in the 'SIP
Extension General Parameters', when you configure SIP Extensions'. Refer the topic "Configuring SIP
Extensions".
Since STUN does not work with symmetric NAT, as an alternative to STUN you can use the Router's Public
IP Address as NAT Traversal mechanism. Ask your Network Administrator about the NAT Traversal
mechanism that suits best for your voice network and configure this parameter.
By default, ETERNITY NE uses the unique MAC Address of its Ethernet Port on all ethernet frames. If you want the
system to use a Clone MAC Address, configure MAC Address Selection.
• In the Clone MAC Address field, enter the Clone MAC Address (in hexadecimal format) to be used on all
ethernet frames.
DHCP Server
ETERNITY NE has an in-built DHCP server for registering the Matrix Extended IP Phones. Configure DHCP Server
settings if you are connecting Extended IP Phones.
• In the Start IP Address field, define the starting IP Address to be assigned to DHCP client. ETERNITY will
allocate the first available free IP Address to DHCP clients.
• In the Number of Hosts field, define the maximum number of DHCP Clients to be assigned IP Addresses
by the DHCP server. You can define the Number of Hosts from 2 to 48. Default: 48.
• In the DNS Server Address field, enter the DNS address to be assigned to the DHCP Clients.
• Define the maximum Lease Time (hours) for which the IP Addresses should be allocated to the DHCP
Clients. The Lease Time may be from 1 to 192 hours. Default: 24 hours.
DHCP client should renew its lease within the Lease Timer. If the client does not renew its lease before the
expiry of the configured Lease Time, ETERNITY will free the IP Address. This IP Address may be
assigned to another DHCP client.
• ETERNITY NE has an embedded web server called Jeeves, for system configuration. You may change
the Web Server Port for accessing Jeeves, as per your requirement. Valid range of the port is: 80, 1024-
65535. Default: 80
Before you begin configuration of the VMS related parameters, consider the following points:
• Decide which extension users are to be provided voice mail. Make a list of these extensions by their port
type and port number, access codes.
• To upload or download VMS Software, configuration files, mailbox messages, you will need to access the
embedded FTP server of ETERNITY NE.
• Open Jeeves.
You can change the default settings of certain VMS features and functions, like function codes for nodes in a graph,
access code for General mailbox, call transfer message options, etc.
• Check the default values of the parameters described in the following and change them, as required.
General Settings
• Enable Extension Number Validation: The VMS Auto Attendant allows callers to reach directly the
desired party in an organization by, by giving them the option of dialing the extension number of the
desired party.
When the Extension Number Validation flag is enabled, the VMS compares the extension number dialed
by the external caller with the extension numbers configured in the system. If no match is found, the VMS
responds with a message “Invalid Number” (Invalno.wav).
• System Alerts through Email: The VMS allows alerts to be sent to the System Engineer via email.
Enable this flag, if you want system alerts to be sent via email to the System Engineer. You must also
specify the email address to which the system alerts are to be sent.
• SE Email ID: Enter the email address on which the System Alerts should be sent to the System Engineer.
The System Engineer must have access to this email ID. The email ID may consist of a maximum of 64
characters.
• Abort All Processes and send Email to Extension Number of Distribution List: For “Sending
Messages” to a group of persons at the same time, the VMS allows you to create Distribution Lists71 to
which voice messages along with email notification can be sent.
Using Distribution Lists, an extension user can send voice message along with email notification to as
many as 120 extensions of the ETERNITY. Distribution Lists take up a sizeable part of the system
resources, and may slow down the process of sending voice messages to Distribution Lists.
When you enable this flag, the VMS stops all other processes - drops calls, closes Jeeves, stops sending
and receiving of emails over the SMTP client - and sends the voice message to the Distribution List. Thus,
the process of sending messages to Distribution Lists happens much faster. Once the message is sent,
the VMS resumes the other processes.
• Home Node Code for Graph: By default, 0 is defined as the function code for Home Node in a graph.
Dialing 0 takes the caller to the home position in a graph. If required, you may assign a different function
code for Home Node in this field. Refer the topic “Graphs and Nodes” to know more.
General Mailbox Access Code: By default, 1176 is the access code for the General Mailbox.
A General mailbox is a common mailbox in the VMS, with which more than one extension is associated.
When the personal mailbox of any extension is full, all new messages are diverted to the General Mailbox.
The General Mailbox is also used as the destination for recording messages for features of ETERNITY like
Call Taping and Conversation Recording.
Extension users whose voice mail has been diverted to the General Mailbox, and those who want to listen
to the Call Taping and Conversation Recording messages can access the General Mailbox by dialing the
access code and listening to the messages.
If required, you may assign a different access code to the General Mailbox. The access code must not
exceed 6 digits. Valid digits are: 0 to 9, * and #.
• Code to Stop Message Recording: The VMS requires callers/extension users to signal end of message
recording by dialing a code. By default, # is the programmed as the code to Stop Message Recording. If
required, you may assign a different code.
Home Node
Home Node or Home Position is the point from where the journey of the caller/extension user starts in a graph of
the VMS. The VMS greets the caller/extension user and takes them to the Home Position. At the Home position,
Welcome message is played, the VMS offers different options to the caller, like go to Operator, dial the name of the
extension user, go to Home Position, disconnect the call, leave a message.
• Dial by Name: By default, 7 is the function code for Dial by Name. The VMS instructs the caller to dial 7 to
dial by name of the extension user.
• Root Node: By default, 0 is the function code for reaching the Root Node (Home Position). The VMS
instructs the caller to dial 0 to reach the Home Position.
• Mailbox Management: By default, 8 is the function code for Mailbox management. The VMS instructs the
extension users (mailbox owner) to dial 8 to reach their mailbox settings.
• Disconnect: By default, # is the function code for disconnecting the call. The VMS instructs the caller to
dial # to disconnect the call.
• Leave Message: By default, 6 is the function code for leaving messages. The VMS instructs the callers to
dial 6 to leave a message for the extension user.
When the caller/extension user fails to dial the digit, the VMS times out and terminates the call, or it can be
configured to take the caller back to the Home Node.
If you want to allow the VMS to take the caller back to the Home node when the caller does not dial any digit,
specify the Home Node option:
• Select Operator, if you want the VMS to take the caller to the Operator when they do not dial any digit.
• Select Disconnect, if you want the VMS to disconnect the call, when the caller does not dial any digit.
• Select Root Node, if you want the VMS to take the caller back to the Root Node in the graph (from which
they originally entered). When the caller is taken to the root node, they will be able to navigate the graph all
over again.
• Call could not be Attended Message: This message is played to the caller when the called party does
not answer the transferred call.
• Called Party Busy Message: This message is played to the caller with the called party is found to be
busy.
• Called Party Not Available Message: This message is played to the caller when the called party does not
• Leave Message: This message is played to the caller. If you disable this option, the VMS will not offer the
caller the option of leaving a message for the extension user
• Message Verification: This message is played to the caller after they have recorded their message. If you
disable this option, the VMS will not give the caller the option of verifying the recorded message.
• If you have completed configuring the VMS General Parameters, click Submit to save changes.
The Voice Mail Auto Attendant Profile determines how the calls are answered and processed by the VMS. The calls
answered by the VMS may be incoming trunk calls attended by the VMS Auto Attendant or calls made by extension
users to the VMS to access their mailbox.
• A Graph: It is the logical path that the caller/extension user takes within the VMS to reach the desired
destination (person or mailbox). A graph starts from the Root Node and traverses through different nodes
these may be Menu Node, Information Node, Transfer Node, Message Node. At each node the VMS offers
the caller the choice of performing some action like deciding what option to choose (menu node), leaving a
message (message node), accessing information (information node), or reaching an extension number
(transfer node).
The VMS plays voice messages to prompt callers at every step, and the caller/extension user dials single
digit codes to decide which path to take and reach the desired destination. To know more, refer the topic
“Graphs and Nodes”.
You can create 4 different graphs in the VMS, and assign a different graph to each Voice Mail Auto
Attendant Profile.
• System Greetings: It is played to the caller/extension user when the VMS answers the call. System
Greetings are played according to the time of the day, Morning, Afternoon, Evening. By default, the caller/
extension user is played the messages “Good Morning”, “Good Afternoon”, “Good Evening” according to
the time of the day.
You can also record custom message as System Greetings. Refer the topic “Recording Voice Messages”.
• Welcome Message: It is played to the caller when the VMS answers the incoming call according to the
current Time Zone: Working Hours or Non-Working Hours. By default, the welcome message for each time
zone are:
Working Hours: “Welcome! Please dial the extension number Or to dial by name press 7. To leave a
message press 6. To go to operator, press 9. For more options, press 0. To disconnect, press # (hash/
pound).
Non-Working Hours: “Welcome! We are closed due to holiday. To leave a message, press 6. For
Assistance press 9. To disconnect press # (hash/pound).
You can also record custom messages as welcome messages for Working and Non-Working Hours. Refer
the topic “Recording Voice Messages”.
• Directly Route to Root Node: This option takes the caller/extension user directly to the Root Node in the
graph, without playing the System Greeting or Welcome Message. This option can be selected according
to the Time Zone-Working, Break and Non-working Hours. The caller/extension user is taken to the Root
Node in the graph, if the option Route to Root Node is selected for the current Time Zone.
You can create 4 different Voice Mail Auto Attendant Profiles, by changing the values of the parameters described
above.
• the path and options you want to give to external callers and to extension users, and accordingly create
the graphs. Refer “Graphs and Nodes” for instructions.
• the System Greeting you want to play to callers in the morning, afternoon and evening.
• the Welcome Greeting is to be played for Working Hours and Non-Working hours.
• if you want to customize System Greetings or Welcome Messages or any Graph-Nodes messages, and
record the messages. Upload the new voice message files using the embedded FTP server of the VMS
card. “Recording Voice Messages”.
• whether you want the VMS to take the external callers and extension users directly to the Root Node
(without playing the System Greetings or Welcome Message).
For each Voice Mail Auto Attendant Profile that you decide to create, give a number from 1 to 4.
• Click the Voice Mail Auto Attendant Profile sub-link. The Voice Mail Auto Attendant Profile page opens.
• Select a Profile number. You may refer to the number you gave to each profile you created.
• select the desired Graph you want to use for this profile. By default, Graph 1 is assigned to all profiles.
• select the number of the Greeting message to be played for the Morning Time, Afternoon Time and
Evening Time. By default, Greeting message 01 (Good Morning), Greeting Message 02 (Good
Afternoon) and Greeting Message 03 (Good Afternoon) are selected as morning, afternoon and
evening greeting messages respectively.
• select the number of the Welcome Message to be played during Working hours and Non-Working
hours.
By default, Welcome Message 01 and 03 are selected for Working and Non-Working hours
respectively.
Welcome Message 01: “Welcome! Please dial the extension number Or to dial by name press 7. To
leave a message press 6. To go to operator, press 9. For more options, press 0. To disconnect, press #
(hash/pound).
Welcome Message 03: “Welcome! We are closed due to holiday. To leave a message, press 6. For
Assistance press 9. To disconnect press # (hash/pound).
• Select the Directly Route to Root Node check box, if you want the VMS to take the caller/extension
user directly to the Root Node in the graph during Working hours and Non-Working hours. By default,
this parameter is disabled in both time zones.
When Directly Route to Root Node is disabled for a time zone, the VMS will play System and Welcome
Greetings to callers and extension users.
The profile assigned to the extension determines how the VMS will answer and process the call from the extension;
what greetings and welcome message will be played to the extension user; whether or not the extension user
(mailbox owner) will be taken to the root node directly.
• according to the Voice Mail Auto Attendant Profile assigned to the extension, the VMS greets the caller
with the Greeting relevant to the time of the day, plays the Welcome message programmed for the current
Time Zone (Working hours, Non-working hours), and offers the different options as per the graph
assigned.
• If the Route to Root Node is enabled in the Profile for the current Time Zone, the VMS takes the caller
directly to the root node, without playing the greeting and welcome message.
You can assign different Voice Mail Auto Attendant Profiles to different extensions and to Department Groups.
• For SLT extensions, see “Voice Mail Auto Attendant” under “SLT Extensions” for instructions.
• For DKP extensions, see “Voice Mail Auto Attendant” under “DKP Extensions” for instructions.
• For SIP extensions, see “Voice Mail Auto Attendant” under “SIP Extensions” for instructions.
• For Department groups, see the feature description for “Department Call” for instructions.
• first select Voice Mail Auto Attendant as the destination for Routing Incoming Calls during the Day and
Night.
• select a Voice Mail Auto Attendant Profile for the trunk.
• the VMS answers the external incoming call landing on the trunk.
• According to the Voice Mail Auto Attendant Profile assigned to the trunk, the VMS greets the caller with the
Greeting relevant to the time of the day, plays the Welcome message programmed for the current Time
Zone (working hours, non-working hours), and offers the different options as per the graph assigned in the
Voice Mail Auto Attendant profile.
• If the Directly Route to Root Node is enabled in the Profile for the current Time Zone (working hours, non-
working hours), the VMS takes the caller directly to the root node, without playing the greeting and
welcome message.
You can assign different Voice Mail Auto Attendant Profiles (Graph number, Greetings and Welcome Message,
Directly Route to Root Node option) to different trunks.
To assign Voice Mail Auto Attendant Profile to trunks, go to the desired trunk port type, “CO Trunks”, “Mobile
Trunks”, “SIP Trunks”, under Basic Settings.
• Select Voice Mail Auto Attendant as destination for incoming calls during the Day and or Night, as
desired.
• Select the auto attendant profile number as Voice Mail Auto Attendant Profile number.
A Distribution List enables extension users to send the same message to a group of extensions at the same time.
Any extension with a mailbox can be included in a Distribution List. You can create five Distribution Lists of 24
members each.
How to configure
To configure Distribution Lists, decide which mailbox owners to be included in a list.
It would be more helpful if you also gave a name to the distribution list, for example, sales, marketing, customer
care, etc.
Draw a two-column table on a piece of paper for each Distribution List. On one column, write the number of the
member in serial order from (001 up to 099). You may write the numbers corresponding to the number of members
you wish to include in this list.
Now, in the next column, for each member number, write the extension number (this is the access code you have
assigned to the extension).
Distribution
Members
List 1
01
02
03
010
Distribution
Members
List 2
01
02
03
08
• Now, refer to the Distribution list(s) you created on paper, and enter the same information in the
appropriate fields on this page.
A Graph is a logical road map within the VMS which enables callers to easily navigate through the system and
reach the desired destination (person in an organization) by following voice prompts and dialing appropriate digits.
A Node is a logical stopover point in a graph where some activity like making a decision, leaving a message,
accessing information, etc. is made.
A Graph is thus a logical and meaningful association of different nodes. A graph initiates from the ‘Root Node’ and
traverses through different nodes.
Six-node Graph
Root Node
1
‘1’
2
‘5’
‘2’
Message ‘4’ 6
Node ‘3’
3 Transfer
5 Node
4
Information Transfer
Node Node
Transfer
Node
Callers must dial appropriate digits while traversing through the graph to reach the desired destination. Single digit
codes are used to navigate within a graph.
While traversing through the graph, the VMS uses voice messages to prompt callers at every step so as to enable
them to reach the desired destination.
Let us take the example of the 6-node graph illustrated above to understand graphs and nodes. In this 6-node
graph, the caller reaches the Root Node by dialing the digit '0'. The Root Node is the entry point of the graph.
At the Root Node, the VMS offers the caller different options like leaving a message, accessing Company
information, have the call transferred to a mailbox, or a particular extension, operator, or department.
For each option, the VMS prompts the caller to dial a particular digit. Each option takes the caller to a particular
node.
• Dialing '1' at the Root Node takes the caller to the Message Node.
• Dialing '2' at the Root Node takes the caller to the Information Node.
• Dialing '3', '4' or '5' at the Root Node will take the caller to the Transfer Node.
The caller decides which path to take and dials the single digit code for the desired path announced by the VMS in
the voice prompt.
Each node has a voice file (a prompt of maximum 60 seconds) attached to it. When the caller traverses through the
graph, at each node the VMS plays the prompt attached to it. The VMS also provides you the flexibility to program
a prompt of your choice at each node. You can also change the digit to be dialed to traverse from one node to the
other.
Menu Node
This node is the point in the graph, at which the VMS presents the callers with different Options in the voice prompt.
The caller is given a menu to select from, hence the name ‘Menu Node’. The caller must choose an option offered
in the voice prompt to navigate through the graph further.
There may be several menu nodes in a graph. In this sense, the Root Node is the first Menu node in a graph (see
6-node graph).
A different voice prompt can be recorded and played to the caller for each Menu node.
At a menu node, a caller can be offered 9 different options. The digits 1 to 9 can be programmed for the selecting
the options. The caller dials the digit for the desired option to traverse from one node to another.
Transfer Node
This node is a point in the graph from where the VMS takes the caller to a specific extension. Each transfer node is
linked to an extension. When a caller reaches the transfer node, the VMS transfers the call to the destination
extension which is assigned to the transfer node.
Thus, the transfer node can be used for allowing callers to reach an extension of a department group, a help line,
an information desk, etc.
No voice prompt is played to the caller at the Transfer node. The VMS simply transfers the call to the extension
linked to the node.
Transfer node can be reached only from a Menu node. A graph may terminate at the Transfer node.
Message Node
This is a point in the graph at which the VMS allows the caller to leave a message. Each message node is linked to
a mailbox. When the caller reaches the Message node, the VMS offers the caller the options: i) leave (i.e. record) a
message, ii) go to Home Position (root node), or iii) disconnect the call.
When the caller chooses the option to leave a message, the VMS transfers the call to the mailbox linked to the
Message node, where the caller can leave the message. The message node is useful to businesses that want to
use interactive voice response for taking orders, registering complaints, feedback, etc. Customers can place
orders, have their complaint registered, leave special request messages.
A different voice prompt can be recorded and played to the caller for each Message node.
Information Node
The Information node is a point in the graph at which the VMS delivers information to the caller. When the caller
reaches the Information node, the VMS plays the
The Information node is a useful when you want to provide information related to the company/organization,
company profile, products, new launches, distribution network, etc.
At information node, the caller has two options: i) to go to Home Position (Root Node), or ii) to disconnect.
The Information node can be reached only from a Menu node. A graph can be terminated at the Information node.
Twenty-two-node Graph:
1 Root
Node
Menu Menu
Node Node
2 4
Menu Menu
Node Node Menu
8 9 Node
10
Each node has an audio file (containing a voice message of a maximum of 60 seconds) attached to it.
The voice message is played to the When the caller traverses through the graph, at each node the VMS plays the
prompt attached to it. For example, at menu node VMS plays the prompt related to menu options. At information
node, the VMS plays the information prompt. At transfer node since no prompt is required, the VMS simply
transfers the call to the designated destination. You can program the prompt of your choice at each node.
A graph forms the basis of the “Voice Mail Auto Attendant Profile”. The Voice Mail Auto Attendant Profile
determines how calls to the VMS72 will be answered and processed by the VMS, on the basis of the type of graph
assigned to it, and the greeting and welcome messages programmed in it.
• Your graph should have optimum number of nodes, neither too few nor too many.
• While you can offer up to nine different options to the callers at each menu node, we recommend that you
offer not more than 4 options at the menu node, because the callers might forget the first option by the
time they have finished listening to the entire prompt.
• Make sure your graph does not confuse the callers or makes them lose their patience.
• Decide the Graph for extension users having mailbox and the graph for external callers who will be
answered by the VMS.
Decide a simple and easy flow to access the maximum number of extensions—persons and services—in your
organization, and draw a graph.
Prudent Investment is a wealth management company, offering individual investors financial solutions:
• stock broking
• insurance
• mutual fund advisory
The company wants to reinforce its brand promise, Your Interest First.
It wants to use Voice Mail to enhance customer experience and project its image as a company that puts the clients
best interest first. The company wants to use Voice Mail to:
• respond to existing and potential clients 24x7.
• provide information about the Company, service offerings, new retirement plans, hot stock tips, and mutual
funds.
72. These may be incoming trunk calls answered by the VMS Auto Attendant, or calls made by extension users to the VMS to access
their mailbox.
Existing New
Customer Customer
(MN) (MN)
Representative Insurance
for Complaints (IN)
(TN) Information on
Services Provided
Stock Mutual (IN)
Market Funds
(IN) (IN)
Representative
(IN)
Information
on Insurance
Representative (TN)
for Insurance
Representative (TN)
for Stock booking Information
on Shares Information
(TN) Representative
(TN) on Mutual Funds
for Mutual Funds
(TN)
(TN)
• The page of Graph 1 opens. To select a graph number, you may click the tab.
• A graph has maximum 64 nodes. For each Node Number, select the Node Type.
• Select Menu as Node Type if you want to give more options to the caller/extension user at this point in
the graph. Doing so, the field 'Extension Number' will become un-editable. The fields 'Destination Node
for the Digits' will become editable.
In the Destination Node for the Digits fields, assign destination nodes to which you want the VMS to
take the caller/extension user on pressing the digits 1 to 9 from the Menu node.
By default, the destination node number is '00' for all the digits at menu node. To de-assign a
destination node for a particular digit at menu node, enter '00' for that digit.
• Select Information as Node Type, if you want to give some information to the caller/extension user.
Doing so, all 'Extension number' and 'Destination Node for the Digits' will become un-editable.
• Select Message as Node Type, if you want to give the caller/extension user the option of leaving a
message at this point in the graph. Doing so, the fields 'Destination Node for the Digits' will become un-
editable.
• Select Transfer as Node Type, if you want give the caller/extension user the option of reaching the
desired extension of ETERNITY. Doing so, the fields 'Destination Node for the Digits' will become un-
editable.
In the Extension Number field, enter the number of the extension to which the call is to be transferred
from this node.
For email transmission, the VMS uses Simple Mail Transfer Protocol (SMTP). If you intend to use Email Based
Notification and send System Alerts to the System Engineer, you must:
• configure the parameter Message Wait Notification via Email73 in the VMS settings of the extension.
• configure the parameter System Alerts through Email and SE Email ID in the VMS General Parameters.
• configure the SMTP Settings.
73. You can also have the new voice message mailed as an attachment with the message wait notification.
• Requires Authentication?: Select 'Yes' if SMTP server requires authentication for the e-mail service.
By default, it is set to 'No'.
If you enable Requires Authentication, you must also enter the User ID and Password.
• Enable Secure Socket Layer (SSL)?: Select 'Yes' to enable SSL, if all data to the SMTP server are to
be transmitted over secure layer. By default, it is set to 'No'.
• Display Name: Enter the name to be displayed to the mail recipient in this field. The Display Name
may consist of up to 24 characters (maximum). By default, this field is blank.
• E-mail ID: Enter the Email ID provided by your SMTP server in this field. This e-mail ID will appear to
recipients as the originator of the e-mail. The e-mail ID may consist of 64 characters (maximum). By
default, it is blank.
• User ID: Enter the User ID provided the SMTP server. User ID is required if you have enabled
Requires Authentication? The User ID may consist of a maximum 40 characters. By default, the field is
blank.
• Password: Enter the Password provided by the SMTP server. The User Password can be of maximum
24 characters. By default, it is blank.
• SMTP Server Address: Enter the SMTP Server Address here. The SMTP Server Address may have a
maximum of 46 characters. By default, this field is blank.
• SMTP Server Port: Enter the SMTP Server Port Service Provider in this field. The valid Port values
are: 25, 587, 465 or in the range of 1024 to 65535. By default, SMTP Server Port is 25.
Timers
• Connection Timeout Interval: This is the time duration for which the VMS will wait for a response
from the SMTP server. You may change the Connection Timeout Interval timer, if required. The range
of Connection Timeout Interval timer is 01 to 99 seconds. By default, it is set to 60 seconds.
• Reconnection Interval: This is the time duration for which the VMS will wait before attempting to
reconnect with the SMTP server. You may change the Reconnection Interval timer, if required. Range
of Reconnection Interval timer is 01 to 10 seconds. By default, it is set to 10 seconds.
• Click the button 'Click to Test SMTP’ to check if the SMTP Parameters have been configured
correctly.
When you click this button, the alert message will appear: "Testing SMTP can take up to 99 seconds.
Would you like to continue?" Click 'OK' button.
The message "Please refresh the web browser after few seconds to check the test mail status" will
appear. Click 'OK' button.
• Test Status: Any one of the results listed below may appear in this field:
Test Status Message Description
"SMTP Server Connection Not Established" When connection to SMTP server fails due to any
reason.
"Login to SMTP Server Failed" When connection to SMTP server is established
but login to SMTP server fails due to any reason.
"Sending Test Mail Failed" When connection to SMTP has been established
successfully but there is no acknowledgement for
the test mail sent.
"Test Mail Sent Successfully" When acknowledgement for the test mail sent to
SMTP server received successfully.
The VMS of ETERNITY NE supports voice messages for different functions, which are broadly classified as:
• System Greetings: These are voice messages played when a new call lands on the VMS. Callers are
greeted according to the time of the day - morning, afternoon, evening (Time Zone). A different System
Greeting can also be played to callers on holidays.
System Greetings are played to callers when the VMS Auto Attendant feature is enabled on trunks.
• Mailbox Greetings: These messages are played to callers when they are diverted to the extension user’s
mailbox to leave a message. Extension users can record personal mailbox greeting messages of their
choice.
• Welcome Messages: These are voice guidance messages played to the callers who call the VMS.
Welcome messages help the callers navigate through the VMS (graph and its nodes). Welcome messages
are played according to the time of the day, i.e. the Time Zone programmed in the system.
Welcome Messages are played to callers when the VMS Auto Attendant feature is enabled on trunks.
• Graph-Node Messages: These are voice guidance messages that are played to help the caller take some
action once they have entered the VMS (graph). Refer the topic “Graphs and Nodes” to know more.
• Prompts/Responses: These are voice guidance messages that are played to the caller in response to the
action taken (i.e. when the caller dials a digit).
For all of these message types, audio files containing the appropriate recorded voice guidance messages are
loaded in the configuration of the VMS Card. The VMS plays the messages related to the function it is performing.
For example, if Multi-stage DID (the VMS Auto Attendant feature) is enabled on a trunk, the VMS plays messages
relevant to the Multi-level DID profile programmed for the current Time Zone. This helps the caller navigate through
the graph and nodes assigned to the Multi-level DID Profile for the Time Zone. At each node, the VMS plays the
related message, as explained below:
• The VMS plays the default System Greeting and Welcome message to the caller according to the time of
the day, e.g.: “Good Morning”. “Welcome. Please dial the extension number or to dial by name, press 7. To
leave a message, press 6. To go to Operator, press 9. For more options, press 0. To disconnect, press #”.
• As the caller navigates, the VMS plays the pre-recorded voice messages related to the particular node the
caller has reached. If the caller dials 6 to leave a message, the VMS takes the caller to the message node,
and plays voice message for this node, e.g.: “Dial 1 to leave a message, ‘2’ to disconnect, ‘0’ to go to
Home Position”.
• If the caller dials 1 to leave a message, the VMS plays the prompt: “Record your Message after the beep
and press any digit to end”.
• After the caller has recorded the message and dialed a digit, the VMS plays back the recorded message:
“The message you recorded is....”
• The VMS responds with the option of message verification: “To re-record the message, press 1, to confirm
press 2.”
• If the caller fails to dial any digit, and the call is timed out, the VMS ends the call with the response: “Thank
you your call”.
In the same way, when you set a voice guided alarm, the VMS plays the alarm-related voice prompts, like: “Enter
the time, HH MM in twenty four hour format”. Thus, for every voice mail related function or feature, the VMS plays
the appropriate voice message.
The VMS gives you the option of either using the default voice guidance messages loaded in the VMS Card
configuration, or recording custom messages that better suit your purpose.
All VMS default voice messages are in English only. If you want, you may record voice messages in your local
language.
No special programming is required for using the default voice messages. However, if you want to use custom
messages, you must first:
• record the message (Greetings, Welcome Messages, Graph Node messages, Voice Guidance prompt).
• upload the new recorded message file in the VMS configuration.
• Each voice message file has a unique file name, e.g.: ‘Thankyou.wav’ containing the message “Thank you
for your call”; ‘Extbusy.wav’ containing the message “The person you called is busy”.
• The voice message file is tagged with a unique number, referred to a Prompt Number, e.g. ‘Thankyou.wav’
is tagged with the prompt number 108, ‘Extbusy.wav’ is tagged with the prompt number 100.
The complete list of voice messages by their prompt numbers and file names is provided at the end of this
topic.
• There are two ways to record custom voice messages. You may either:
You may upload the files using the embedded FTP server of the VMS Card.
• Voice messages are in WAV format, so the custom messages must also be in the same format.
• When you record voice messages from a Telephone, the VMS stores the message you record in the
required file format, with its unique file name and prompt number. All you need to do is define the
Prompt Number when you record your message.
• When you record voice messages from any other source and upload the them in the VMS configuration
files, make sure that the audio files are recorded in .wav file format, with the attributes listed below:
• Bit Rate: 128 kbps
• Audio Sample Size: 16 bit
The audio file of the custom message you have recorded must have the same unique file name as the
existing default audio file.
• Voice Messages can be recorded and played back from the System Administrator mode.
• Extension users can record their personal mailbox greetings on their own. Refer the topic “Mailbox
Settings” to know more.
• Enter SA mode.
System Greeting
Filename Prompts/ Response
Number
01 SysGrt01.wav "Good Morning"
02 SysGrt02.wav "Good Afternoon"
03 SysGrt03.wav "Good Evening"
04
05
06
07
08
09
10
11
12
Only three System Greeting files are programmed. Greeting number 04 to 12 are blank. You may record
your custom greetings on any of these files.
• The VMS will prompt you to start recording your messages after the beep and press # (hash/pound) sign
to indicate end of message.
• Follow VMS prompt to record your message.
• The VMS will prompt you to start recording your messages after the beep and press # (hash/pound sign)
to indicate end of message.
To check Mailbox Greeting you recorded for a time zone for an extension’s mailbox, dial:
• 1072-309-Extension-Timezone Index
• The VMS will playback the message you recorded.
• The VMS will prompt you to start recording your messages after the beep and press # (hash/pound sign)
to indicate end of message.
To check the Voice Prompt you recorded for a node in a graph, dial:
• 1072-303-Graph-node
• The VMS will playback the message you recorded.
• The VMS will prompt you to start recording your messages after the beep and press # (hash/pound sign)
to indicate end of message.
• Follow VMS prompt to record your message.
As mentioned earlier, when you record voice messages from an external source, make sure that
• The audio file is recorded in the prescribed format (.wav) and attributes.
You may recall that each voice message file has a unique file name, and is tagged with a unique Prompt
Number. For example: You have customized the thank you messages as: “Thank you for calling” (default
message: Thank you for your call). Make sure that the new message is saved with the same unique file
name ‘Thankyou.wav’.
You can upload Custom Voice Messages using Windows FTP or FireFTP. Make sure FireFTP Add-on is installed
in your browser.
2. Type the current IP Address of the Ethernet Port of ETERNITY NE in the Address bar as ftp://
192.168.1.101
5. In the Password field, enter the SE password (default 1234) and click the Log On button.
6. On successful login, the FTP window will open. You will see the different Configuration folders in this
window.
• The existing audio files, with their unique names and .wav extension appear.
• Go to the location (CD, Pen Drive or Local disk) where you have stored the custom voice message(s).
• Click the desired folder. The Thank you message is a response, so open the System Prompts and
Responses folder.
Right click, and paste the file you copied in the System Prompts and Responses folder.
As you are pasting a file with the same file name, you will be prompted if you want to replace the existing
file. Click OK.
For instructions on uploading voice messages using FireFTP, see “Configuration Upload”.
The VMS can be configured to announce the extension user’s name to caller when transferring the call to the
extension. To do this, you must
Announce Name can be enabled on all extension types, SLT, DKP, SIP extensions, under the Voice Mail Auto
Attendant parameters of extensions. For instructions see “SLT Extensions”, “DKP Extensions”, “SIP Extensions”
under Basic Settings.
Station Names can be recorded by the System Administrator as well as by the extension users. The instructions
provided here are for the System Administrator only. Instructions for extension users may be found under the topic
“Mailbox Settings”.
• Enter SA mode.
• The VMS will prompt you to start recording the Station Name after the beep and press # (hash/pound) sign
to indicate end of message.
As you record the names of each extension, the VMS will create audio files with unique file names, ‘stn001.wav’,
‘stn002.wav’, etc. These files are stored in the Station_Names sub-folder in the VMSCard configuration folder.
Abbreviated Dialing
What's this?
Abbreviated Dialing is the use of Short Codes (abbreviated numbers), typically 2-3 digits, to dial out long-digit
numbers. It is also referred to as Memory Dialing.
Abbreviated Dialing allows you to quickly and easily dial frequently called, long-digit numbers.
This feature requires you to store the frequently called, long-digit numbers74 and their corresponding short codes in
special lists, known as 'directories'. These directories may be 'personal' or 'global'.
ETERNITY NE supports two types of Abbreviated Dialing based on the type of directory used: Personal
Abbreviated Dialing and Global Abbreviated Dialing.
Personal Directories can be configured and assigned to groups of extensions. The use of Personal Directories is
limited to the extensions to which they are assigned.
A single personal directory accommodates 25 numbers. Each number may be up to 16 digits long. A personal
directory has Index numbers from 01 to 25 against which the frequently dialed telephone numbers are stored along
with their corresponding names and Trunk Access Code (TAC).
As many as 50 different personal directories, numbered from 01 to 50 can be created and assigned to SLT and
DKP extensions and Extended IP phone extensions.
With a personal directory assigned to an extension, the extension user simply dials out the Feature Access Code
for Abbreviated Dialing and the Index Number at which the number to be dialed is stored in the personal directory of
the extension.
For example: personal directory number 02 is assigned to extension 201. The number 02652630555 is stored at
Index number 16 of this directory. The user of extension 201 can call this number by simply dialing '8' (feature
access code) followed by '16' (the index number).
74. These may be numbers of your branch offices, your clients, as also numbers of emergency services such as fire, police.
Each extension can access only the personal directory assigned to it.
Being a system-wide list, the Global Directory can be accessed by any extension connected to the ETERNITY NE.
The Global Directory has the capacity to store up to 900 numbers of a maximum of 16 digits each. The Global
Directory is divided into three parts:
The Global Directory has Memory Location codes starting from 100 to 999. The telephone numbers along with their
corresponding names are stored against Memory Location codes.
Whenever an extension user of ETERNITY NE wants to use Global Abbreviated Dialing, all that the user needs to
do is dial the feature access code ('8' or '6') and the Memory Location code at which the desired number is stored.
For example: the number 02652630566 is stored at Memory Location 102 of the Global Directory. Now, extension
users of ETERNITY can call this number by simply dialing the '8' or '6' (feature access code for Abbreviated
Dialing) followed by '102' (Memory Location code at which the desired number, 02652630566, is stored).
The ETERNITY NE will dial out the number using any of the trunks selected for Routing Global Directory’s
numbers.
• Extensions can use Global Abbreviated Dialing only if this feature is included in the “Class of Service
(COS)” allowed to them.
• Further, an extension can access only that part of the Global Directory which is allowed to it in the CoS.
For instance, if extension 201 is allowed Global Directory Part 1 in its CoS, the user of extension 201
can dial out only those numbers contained in Global Directory Part 1.
• Extensions must be assigned all three parts of the directory in their Class of Service to be able to
access the entire Global Directory. By default, only Global Directory Part 1 is included in the CoS of all
extensions. Therefore, all extension users can dial numbers stored in Global Directory Part 1.
• When an extension user dials a number stored in Global Directory Part 2 or Part 3, the system checks
for Class of Service and Call Budget.
• When an extension user dials an abbreviated number from the Personal Directory, the system first
checks the outgoing trunk and Toll Control Level (Call Privilege) of that extension and then dials out the
number.
How to configure
For both Personal and Global Abbreviated Dialing to work, the System Engineer must:
2. Assign Personal Directory to the desired extensions, which may be SLT, DKP, and SIP extensions.
3. Enable Global Directory Part 1, Part 2, or Part 3 as desired in the Class of Service allowed to the
extensions.
It is possible for extension users to configure their Personal Directories using their own phones.
It is also possible for DKP and Extended IP Phone users to add, delete and edit contacts in Global Directory Part 1
from their extension phones. For this, Global Directory Part 1 must be allowed in their CoS by the System Engineer.
• Ask the extension users the numbers they would like to be included in the personal directory of their
extension.
• Make separate lists of numbers along with their corresponding names and trunk access codes, for each
personal directory. You may draw four-column tables on paper and enter the Numbers and corresponding
names and trunk access codes against each Index number. For example:
Personal Directory 01
: : : :
: : : :
25
Personal Directory 02
: : : :
: : : :
25
• Compile the numbers to be included in the Global Directory. Numbers that are commonly dialed by all
extensions can be included in the Global Directory.
• Draw a four-column table on paper and enter the telephone numbers along with their names, and the
outgoing trunks to route the call. For example:
Global Directory
100
101
: : :
: : :
999
• Prepare the Global Directory keeping in mind that is divided into three parts: Part 1 (100 to 799), Part 2
(800 to 899), and Part 3 (900 to 999). As Part 1 is allowed to all extensions in their default CoS, ensure that
numbers to be allowed to all extensions are included in this part of the directory.
You may assign the same personal directory to more than one extension.
• Open Jeeves.
Personal Directory 1 appears on this page. Against each Index Number from 01 to 25,
• Enter the contact’s Number you wish to store. The number may consist of 16 digits (maximum).
• Enter the contact Name against the Number. The Name may contain up to 12 characters (maximum).
• To configure another Directory, click the personal directory number tab 02 to 25.
• Follow the same steps as described above to configure each Personal Directory.
Keep a print out of each personal directory for your records. This will also help you take care of overlaps
and include some of the numbers that are dialed by all users in the Global Directory instead of the
Personal Directory.
After you have configured the Personal Directories, assign a personal directory to each extension. The extension
may be an SLT or DKP or SIP extension.
• To assign Personal Directory to another SLT, click the tab of the desired SLT and follow the same steps as
described above to assign the Personal Directory.
• Select the Personal Directory number to be assigned to the DKP from the list box.
The table for the entries 100 to 199 appear on this page. Against each Index number,
• Enter the contact’s Number you wish to store. The number may consist of 16 digits (maximum).
• Enter the contact Name against the Number. The Name may contain up to 12 characters (maximum).
• To configure more numbers, click the next tabs 200-299, 300-399, 400-499, 500-99, and so on. Follow the
same steps as described above.
• Select the trunks to be used for routing the numbers of the Global Directory dialed out by extensions.
Double click the Route Global Directory’s numbers using Trunks field.
• To select trunks, place your cursor on the desired trunk listed on the left box and click the Select>> button.
listed on the left box.
• You may change the sequence of the trunks using the Up and Down buttons on the right display box.
• You may enable Rotation, if you have selected more than one trunk.
• If you want to apply the Least Cost Routing logic on the trunks, select the desired LCR type from the list
box: Time based, Number based, Time+Number based, Service Provider based. See “Least Cost Routing
(LCR)” to know more.
• Now, enable Global Directory in the “Class of Service (COS)” of the extensions. To do this,
• On the respective extension page, click “Class of Service (COS)” to expand options.
Enable the part of the Global Directory you want to assign to the extension in its Class of Service.
You may also configure the Personal Directory using an SLT, but you will not be able to enter Names of contacts in
the Directory.
OR
• Dial 1071.
• Enter Personal Memory Index (001 to 025)
• Enter Number of the contact (max. 16 digits).
• Press 'Enter' key.
• Enter Name of the contact.
• Press 'Enter' key.
• Enter Trunk Access Code.
• Press 'Enter' key.
• You get confirmation tone and the message on your phone's display.
• Extension users can only add, delete and edit names and numbers of contacts in Global Directory Part
1. However, they cannot select the outgoing trunks for the contact numbers in the directory.
• When an extension user configures Global Directory Part 1, the system will automatically assign the
number and name to a free Memory Location. The system will use the trunks selected by the System
Engineer to dial out the number added by the extension user.
To configure Global Directory Part 1 from the DKP or Matrix Extended IP Phone, follow these steps:
Adding a contact
To add a contact, select ‘Add’ and press Enter key.
• Enter your contact’s name on the prompt.
A maximum of 12 characters are allowed.
• Dial #* or Press Enter key to terminate command.
• Enter your contact’s number on the prompt.
A maximum of 16 digits are allowed.
• Dial #* or Press Enter key to terminate command.
• You will get the confirmation tone and the confirmatory message: “Stored at Index xxxx”.
Editing a contact
To edit a contact,
• Scroll to ‘Contacts’ in the phone menu and press Enter key.
• Select ‘Edit’ and press Enter key.
• You get the prompt: 'Name:'
• Enter the initial letters of the contact's name.
• The number of matching entries that will appear at a time on your phone's display will vary according to
your phone's LCD display capacity.
• Scroll with the Up/Down navigation keys to reach the desired contact's name on the list.
• Press 'Enter' key to select the name.
• The system displays the name you selected.
• To delete a character, use the Back/Forward navigation key to place the cursor under the character you
want to delete.
• Press the ‘Transfer’ key to delete the character you selected with the cursor.
• To enter a character, use the Back/Forward navigation key to place the cursor in the position you want to
enter the character.
• Enter the desired character by pressing the relevant digit pad keys in quick succession.
• After you have finished editing the name, press Enter key.
• The number of the contact whose name you edited will be displayed.
• Repeat the same steps as you did for editing the name.
• After you have finished editing the name, press Enter key.
• You will get the confirmation tone and the confirmatory message: “Stored at Index xxxx”.
Deleting a contact
To delete a contact,
• Scroll to ‘Contacts’ in the phone menu and press Enter key.
• Select ‘Delete’ and press Enter key.
• You get the prompt: 'Enter Name to select and Press Enter Key to delete.'
• Enter the initial letters of the contact's name.
• The number of matching entries that will appear at a time on your phone's display will vary according to
your phone's LCD display capacity.
• Scroll with the Up/Down navigation keys to reach the desired contact's name on the list.
• Press 'Enter' key to delete the name.
• You will get the confirmation tone and the confirmatory message: “Done”.
OR
OR
OR
OR
OR
OR
What's this?
Access codes are short digit sequences dialed from an extension phone to instruct the PBX to perform a function
such as:
• Calling an extension.
• Station Codes: Codes used for calling DKP, SLT, SIP extensions, Digital Output Port, Door Phone. These
codes are also commonly referred to as Extension numbers.
• Logical Group Codes: codes used for calling a group of extensions as in a Department group, a group of
trunks.
Default feature codes: there are different feature codes for every feature and function of the ETERNITY
NE. For example, '2' for Auto Call Back, '5' for Raid, '13' for Call Forward.
You can change the default access codes to the codes of your choice. For example: the default Operator
code '9' can be changed to '0'.
How it works
Whenever an access code is dialed from an extension, the system matches each digit in the code with the access
codes configured within the system to determine the instruction, like whether it is an extension it must call, or a
trunk line it must grab, a port it has to activate. The system processes the instruction when a match is found.
• When the first digit '1' is dialed, the system finds a match. As several default access codes begin with '1'
the system waits for the next digit to be dialed.
• When the second digit '3' is dialed, the system finds a match for '13'.
• As '13' is common for all Call Forward options75, the system waits for the next digit to be dialed
• When the user dials the third digit '1', the system finds a match for '131'.
• If there is more than one access codes matching with '131', e.g. '1311', '1314', '1315' the system will wait
for the next digit to be dialed.
• If no further digit is dialed on expiry of the Inter Digit Wait Timer, the system understands the instruction as
'Call Forward - Unconditional' and waits for the destination phone number to be dialed.
Access Codes are related to various phases of a call. When a call is processed by a PBX, it goes through a number
of pre-defined phases.
No Digits are The system The dialed The dialed Connected Connected No reply
activity. pressed on is processing extension extension is with the with two from dialed
the phone the call. The is busy. ringing. dialed extensions. extension.
keypad/dialed call is neither extension.
from the placed nor
rotary. blocked.
Dial tone is Beeps are Busy tone Ring Back Two-way Three-way Error Tone
played. played. is played. Tone is speech. speech. is played.
played.
Different access codes are dialed at different call phases. Station Codes and Logical Group Codes are dialed in the
'Dial' phase.
As different features are invoked in each call phase, Feature Access Codes are dialed at different call phases. For
example:
• Auto Call Back code is dialed at the 'Blocked' phase as well as 'Placed' phase.
• Three-party Conference code is dialed at the 'Matured 2-way' phase, with one held party.
75. Call forwarding options: Unconditional, When Busy, When No Reply, When Busy or No Reply.
Each access code in a single call phase may be of different lengths, but must be unique. For example, the same
access code cannot be used for two different features like Call Forward and Redial, since both these features are
invoked in the 'Dial' phase.
However, the same access code can be used for features in different call phases. For example, '4' is the default
feature access code for DND Override (Routing Phase), Call Pick-Up-Group (Dial Phase) and Barge-In (Blocked
Phase).
Similarly, Station and Logical Group Codes too must be unique and should not match with any of the features
invoked in the 'Dial' phase.
How to configure
ETERNITY NE provides default Access Codes for extensions, logical groups—department and trunk groups—and
features.
It also provides country-specific default Access Codes which are applied automatically when you select the
'Region' to configure the system.
The default Access Codes for India are presented in the table below. The default Access Code tables also indicate
the call phase in which each feature is invoked.
Call Phases
Access
Feature Matured Matured
Code Dial Routing Blocked Placed
2-Way 3-Way
Redial 7 Y Y
Abbreviated Dialing 8 Y Y
Operator 9 Y Y Y
Call Forward 13 Y Y
Dynamic Lock 14 Y Y
Hot Line 15 Y Y
Alarm 161 Y Y
DND 18 Y Y
Interrupt Request 3 Y
Barge-In 4 Y
Raid 5 Y
Trunk Reservation 6 Y
Call Toggle 1 Y
Conference- 3 Party 0 Y
Conference-Multi Party 19 Y Y
Paging 1074 Y Y
Flashing on Trunk * Y Y
DND Override 4 Y
Presence 1097 Y
Hold Flash Y
Mute 1052 Y Y
SA Mode 1072 Y Y
Keypad Lock - Y
Reminder 162 Y Y
Invoke RCOC ** Y
Voicemail 3931 Y Y
391 to
Dept. Group Y Y
395
You can either use the default Access Codes or change them to suit your preferences.
To change Access Codes, see “Extension Numbering Plan” under Basic Settings for instructions.
What's this?
Account Codes are a very useful feature for business consultants, law firms, advertising and media agencies, and
the like, which interact with third parties on behalf of their clients. Such organizations need to keep track of calls
made to and on behalf of each client.
An 'Account Code' is a unique three-digit number that an organization can assign to each of its clients. Each
Account Code may be given a name, which is entered in the Account Name List.
With an Account Code and Name assigned, whenever calls are made to the client or to a third party on behalf of the
client,
• The extension user dials the Account Code or Name assigned to the client.
• The extension user may dial the Account Code before dialing the external number,
Or
when in speech with the client/third party (by putting the party on hold).
• Call details for these calls are recorded in the Station Message Detail Recording Report (SMDR) for
Outgoing Calls.
• The SMDR report can be printed using the Account Code as filter.
This way, the organization can know the details of calls made to and on behalf of each client.
To illustrate this with an example: An advertising media agency makes nearly 100 calls every day to and
on behalf of its clients that include 'Midas Business Solutions', 'Jet-Set Holidays', 'Bacchus Vineyard',
among several others.
Now, the agency can assign a three-digit account code to Midas Business Solutions, e.g. '001' and the
enter the name code 'Midas Biz' in the Account Name List. The same can be done for all other clients.
Each time someone in the agency makes a call to Midas, they may dial either the account code '001' or the
account name 'Midas Biz'. The account code/account name can be dialed either before dialing out the
number or when in speech with the client.
• an extension user dials out the number or the Trunk Access Code to grab a trunk.
• ETERNITY NE displays the Account Name List on the DKP of the extension user.
• To use Account Codes, extensions must be allowed this feature in their Class of Service (CoS).
• If you want to use Account Names, you must configure the Account Name List.
• When the Forced Account Code is enabled on a trunk, the system will ask the extension user to enter
the account code irrespective of the method of dialing, whether Global Abbreviated dialing, Personal
abbreviated dialing, Least Cost Routing, or Selective Trunk Access.
• However, if Forced Account Code is enabled on the selected trunk, and the number is dialed using
Selective Trunk Access, the system will dial out the number using Store and Forward dialing.
In the case of Abbreviated Dialing or Direct Dialing, if the extension user fails to dial the Account Code,
an error message will be displayed on the extension user's DKP.
How to configure
For Account Code to work, the System Engineer must:
1. Enable 'Account Codes' feature in the Class of Service (CoS) of the extension(s) to which this feature is to
be allowed.
2. Prepare and configure the Account Name List, if you are going to assign names to the account codes.
3. If Forced Account Code is to be used, enable 'Forced Account Code' flag on the trunks through which calls
using account codes are to be made.
• Write Account Codes on one column. Account codes may be any three-digit number between 001 and
999.
• Write the Account Names, i.e., names of the clients on the second column, against their respective
Account Codes.
010 Bacchus
You need not follow a cardinal numbering sequence when assigning Account Codes.
You may assign any code to any client. For instance, you can assign code '111' to Midas Business
Solutions, '222' to Jet-Set Holidays, '333' to Bacchus Vineyard.
• Open Jeeves.
• Enter the Names of the clients against the account codes you have assigned to them. Refer the paper with
the two-column table you created.
• Click the link of the extension type—SLT, DKP, SIP—to which you want to allow this feature.
• Select the extension number to which you want to allow this feature.
• Enable Account Code in the Class of Service for the Day and Night.
• Click Submit.
• Select the Do not allow outgoing calls without Account Code check box to enable this feature on the
extension.
• Click Submit.
• Click the link of the trunk type—CO, Mobile, SIP—to which you want to allow this feature.
• Select the trunk you on which you want to assign this feature.
• Select the Do not allow outgoing calls without Account Code check box to enable this feature on the
trunk.
• Click Submit.
How to use
Account Codes can be dialed in two ways: by Number and by Names.
Print and hand out copies of the Account Code List to everyone in the organization for reference while
making calls.
OR
• Dial 1058
• Enter Account Code
• Dial Trunk Access Code
• Dial the number of the client.
OR
• Dial 1058
• Enter Account Code
• Speech will be resumed.
To enter Account Code Number when Forced Account Code Flag is enabled:
OR
• Dial 1059.
• Enter the initial letter of the client's name.
OR
• Dial 1059.
• Dial the initial letter of the client's name.
The Account Name List will be displayed on your DKP, alphabetically with the corresponding account
codes.
• Scroll to select the desired client name and press Enter key.
Speech will be resumed with the called party.
If you have dialed the wrong account code or name while in the middle of a call, you can correct it by
dialing 'flash' again and following the steps described above. The system will override the previously dialed
account code or name.
See “Station Message Detail Recording–Report”, for more detailed instructions on printing reports using filters.
What's this?
ETERNITY offers Alarm feature on all extensions.
• Once Only - A one-time call, where the extension phone rings at the set time.
• Daily - A repeat call, where the extension phone rings at the set time everyday.
• Personalized - The Operator greets the extension user to serve the alarm request.
• Automated - The system serves the alarm request by playing a voice message or music.
Alarms can be voice-guided, if your ETERNITY NE has the Voice Mail System (VMS) module.
How it works
Personalized Alarm
When the Alarm serving mechanism is configured as 'Personalized',
• The Operator phone rings first1, displaying the number of the extension to which the alarm is to be served.
• When the Operator answers this call, a call is placed on the extension on which the alarm is set.
• The extension rings for the duration of the Alarm Ring Timer.
• When the extension user answers the call, the Operator greets the extension user with the time and alarm
message.
• If the extension user does not answer the call till the Alarm Ring Timer has elapsed, the Operator phone
will display a text message notifying 'No Reply' from the extension. The Alarm is now considered as
served.
• If the extension is busy2, the Operator phone will display a text message notifying that the extension
number is 'Busy'.
1. The Operator phone rings for the duration of the Alarm Ring Timer. If the Operator does not answer the call, the ETERNITY NE will
make two more Alarm Attempts at an Alarm Attempt Interval of 5 minutes to call the Operator.
2. An improperly placed receiver may also be the cause for the busy tone on the extension phone. In that case, the system will notify
the Operator Phone with the 'OFF-Hook Alert'.
• inform the extension user about the alarm in person or send someone to do it.
OR
OR
Automated Alarm
When the Alarm serving mechanism is configured as 'Automated',
• The extension phone rings at the set time till the end of the Alarm Ring Timer. If the extension phone is a
DKP or the Matrix Extended IP Phone, an Alarm message will appear on its display.
• When the extension user answers the call, s/he may be played music-on-hold, or a pre-recorded voice
message, or be connected to the Voice Mail, or routed to the Operator, depending upon the Alarm
Notification Type configured by the System Engineer.
The System Engineer may consult with the Enterprise to decide which of these options is to be configured
as the Alarm Notification Type.
• If the extension user does not answer the alarm call, the ETERNITY NE makes two more attempts (in all, 3
attempts) at an interval of 5 minutes between each attempt, to call the extension.
• If all Alarm attempts go unanswered, the ETERNITY NE places the call on the Operator phone. The
Operator phone rings till the end of the Alarm Ring Timer. The Operator phone displays the extension
number with the message 'No Reply'. The Alarm is now considered as served.
• If the extension phone is busy ETERNITY NE will continue to make Alarm Attempts at the Alarm Interval
configured. When all Alarm Attempts go unanswered, the ETERNITY NE will place a call on the Operator
phone. The Operator phone will display the number of the extension phone with the message 'Busy'.
Snooze
The Snooze function can be added to Automated-Alarms to ensure that the extension user answers the call.
Snooze is a system-wide feature; when set, this function will be added to all Automated Alarms.
• The extension phone rings for the Number of Alarm Attempts configured, at set Alarm Attempt Intervals.
• The extension stops ringing, when the extension user answers the call and dials the Code '0' to
acknowledge the Alarm. The Alarm Acknowledgement Code is not configurable.
The status of Alarms set by Operator as well as extension users appears on this report, with details of time (hours
and minutes), type (once only, daily), and serving mechanism (personalized, automated).
The Operator can view the Alarm report whenever required and can also print this report.
• ETERNITY NE can register as many as 60 Alarm requests set by the Operator and extension users.
• Multiple Alarms can be set for an extension by the Operator and/or by the extension user. For example,
Daily Alarm at 09:00am is set for an extension. The extension user wants to change the alarm time to
08:30am for a day. The extension user/Operator can set another alarm, i.e. a Once Only Alarm, at
08:30am without disturbing the daily alarm. Both the Alarms will ring at the set time.
• When multiple alarm requests have been set on an extension, if the Operator or the extension user
attempts to cancel alarms from the phone, the system will cancel all the alarms set for the extension.
Multiple alarm requests set for an extension can be canceled selectively only from the System
Administrator pages of Jeeves.
• It is not possible to modify an alarm request. Instead, the alarm request should be canceled and a new
one should be made.
• The duration of Alarm Ring Timer, the Number of Alarm Attempts and the Alarm Attempt Interval are
configurable.
• Alarms can be set for all extensions of the ETERNITY NE, including the Operator phone also.
• Alarm settings will be retained in the system during power down and system upgrades.
How to configure
The following parameters are important for the functioning of the Alarm feature. Read their description before
configuring this feature.
Snooze Function
Snooze forces the extension user to acknowledge the Alarm call. With snooze functionality enabled, the system
expects the user to answer the Alarm call by going OFF-Hook and to dial Acknowledgement code '0'.
When snooze is disabled, the system considers the Alarm as answered when the extension user simply answers
the alarm call by going OFF-Hook (dialing acknowledgement code is not mandatory). You may choose whether or
not to enable snooze. By default, snooze is disabled.
User experience however, shows that 'Once Only' Alarm call requests are more common than 'Daily' Alarm
requests. So, ETERNITY NE allows you the flexibility of setting 'Once Only' as the default Alarm Type, by disabling
the 'Configuring Alarm Type' flag.
When this flag is disabled, the system will prompt the Operator/Extension user to enter the Time of the Alarm call
and consider the Alarm Type as 'Once Only'.
If you want to offer only 'Automated' Alarms to extension users, ETERNITY NE allows the flexibility to set
'Automated' as the default Alarm call serving mechanism. This can be done by disabling the 'Configurable Alarm
Category' flag.
When this flag is disabled, the system will consider the Alarm call serving mechanism as 'Automated' and will
prompt the Operator only for the Time of the Alarm call.
• If the 'Configurable Alarm Type' flag is disabled, but the 'Configurable Alarm Category' flag is enabled,
the system will set 'Once Only' alarm calls, but will give the option of selecting 'Automated' or
'Personalized' as the serving mechanism.
• Similarly, if 'Configurable Alarm Type' is enabled, but the 'Configurable Alarm Category' flag is disabled,
the system will allow both 'Once Only' and 'Daily' alarms to be set, but the serving mechanism will be
'Automated'.
• The flags ‘Configurable Alarm Type’ and ‘Configurable Alarm Category’ are not applicable for Voice-
guided Alarms. In the case of Voice-guided Alarms, the Operator/Extension user will be prompted to
select the Alarm type and serving mechanism, each time, even when both aforementioned flags are
disabled.
• Voice Mail: The extension user is connected to the Voice Mail System.
• Voice Message: The extension user is played the message recorded in the Voice Module assigned to
Alarm.
Macros
These are short codes for simulating the Alarm call. These are used for SLTs which have special function keys to
send a fixed string to the ETERNITY NE, when each function key is pressed. The system interprets this string and
translates it into a string that can be understood by the ETERNITY NE. For example, an SLT has a special function
key for Alarm calls which sends the string 53 to the system. The system can be configured to translate 53 received
from the SLT in to the feature access code for Voice-guided Alarm calls, 163.
• Configure, as required, the Alarm Call related parameters: Number of Attempts, Attempt Interval,
Configurable Alarm Type and Category, and Snooze.
• Configure Macros, if the SLT extension has special function keys, and you want to a function key for the
Alarm feature.
Now,
• Open Jeeves.
• To select an extension number on the page, select the Extension Number tab.
• Click More button to expand parameter options for the selected Extension number.
• In the Alarm Notification Type box, select the desired notification type for the extension: Music-on-Hold,
Voice Message, Voice Mail, Route to Operator.
If you select Voice Message as notification type, make sure that you also assign a Voice Module to the
Alarm application. See “Voice Message Applications” for more details.
Select Voice Mail only if you have the Voice Mail System installed in your ETERNITY NE.
• Repeat the same steps to configure another extension type and extension number.
• Use Alarm with Snooze: Select this check box if you want to use the Snooze function for the Alarm
Call.
• Configurable Alarm Type (Once Only/Daily): Disable this flag, by clearing the check box, if you do
not what the system to provide the Operator and the extension users the option of setting 'Once Only'
or 'Daily' Alarms. When this flag is disabled, the system will allow only 'Once Only' alarms to be set.
• Configurable Alarm Category (Personalized/Automated): Disable this flag, by clearing the check
box if you do not want the system to provide the Operator the option of setting 'Personalized' or
'Automated' Alarm calls. When this flag is disabled, the system will follow the 'Automated' Alarm call
serving mechanism. The Operator will not be prompted to choose between 'Automated' and
'Personalized' Alarm calls when setting Alarm calls for an extension phone.
• Alarm Ring Timer (sec.): Change, if required, the time for which the Alarm Call will ring on the
extension phone and the time for which the Operator phone will ring to notify an unanswered Alarm
Call.
• Number of Alarm Attempts: Increase or decrease, as required, the number of attempts the system
should make to serve an Alarm call.
• Alarm Attempt Interval (minutes): Increase or decrease, as required, the time gap between each
attempt the system makes to serve an Alarm call.
The Macros page opens. Each macro is stored against an index number.
• In the Number String field, enter the strings to be replaced with on receiving the strings from the SLT.
• In the Access Codes field, enter the strings sent by the SLT on pressing the special function key for
'Alarms'.
For example, if the SLT sends the string ‘53’ to the ETERNITY NE when the function key for Alarms is
pressed, enter the string 163 (the feature access code for Voice-guided Alarms) in the Number String
field, and enter the string 53 in the corresponding Access code field.
How to use
Alarms can be set by the extension users by themselves. The extension users can also ask the Operator to set the
alarm for them.
If the Voice Mail System (VMS) is installed in the ETERNITY NE, it can offer voice-guided Alarms to extension
users and the Operator.
For SLT
• Pick up the handset.
• Dial 1072-034
• Follow Voice Mail System Prompts to set/cancel alarm.
• Replace Handset.
If the SLT of the extension user has a special Alarm function key, s/he can set the alarm using this key.
• Press 'Alarms' key. (The label on the SLT key may differ from model to model)
• Follow the Voice Mail System prompts to set/cancel alarm.
• SLTs with special function keys will work only if the corresponding Macros are programmed by the
System Engineer.
• Without the Voice Mail System installed, the extension user having SLT with the special Alarm function
key will not be able to set/cancel alarm. This extension user can set/cancel alarm only by dialing the
feature access code for voice-guided alarms.
To cancel Alarms,
For SLT
• Lift handset.
• Dial 1072-003
• Dial Extension Number.
• Dial Time in HHMM format
• Dial 1 for Personalized, Dial 2 for Automated.
• You get confirmation tone.
• Replace handset.
• Lift handset.
• Dial 1072-003
• Dial Extension Number.
• Dial #
• You get confirmation tone.
• Replace handset.
To set Alarm,
To cancel Alarms,
To cancel Alarms,
• Pick up the handset.
• Dial 161.
• Dial #.
• You get confirmation tone.
• Replace the handset.
• Extension users can set only automated alarms from their phones. For personalized alarms, they must
request the Operator.
• If there are multiple alarms set, alarms cannot be canceled selectively. Only the Operator can cancel
alarms selectively from the SA mode.
• Alarm(s) set on an extension will be served, even if DND is also set on the same extension.
• Open Jeeves.
• select the Cancel Alarm check box of the extension number for which you want to cancel the alarm.
• click the Cancel Selected Alarms button at the bottom of the page.
What's this?
Alternate Number Dialing allows you to dial different phone numbers in an attempt to reach a person whose line is
busy.
Alternate Number Dialing is useful when the person or organization you are trying to reach has more than one
number, where they may be reached. The system dials out different phone numbers of the same party, saving you
time and effort of dialing each of these numbers manually.
How it works
This feature works as an extension of the features “Last Number Redial” and “Auto Redial”. It requires you to
configure first, the Alternate Number Groups in the Global Directory. With the alternate numbers configured in the
Global Directory, all you need to do is to use Last Number Redial or Auto Redial, every time you want the system to
try Alternate Number Dialing.
For example: Midas Business Solutions has four telephone numbers: 2640459, 2631235, 2635589 and 2565590.
To be able to use Alternate Number Dialing, you must first configure all four numbers as Alternate Number Group in
the Global Directory.
Now, when you dial one of these numbers, '2640459', and get a busy tone, you can either initiate Last Number
Redial or set an Auto Redial request.
• The system will dial an alternative number for the dialed number.
• If the redialed number is busy, you can set Last Number Redial again.
• If the second alternative number is also busy, you can set Last Number Redial again.
• This process will be repeated each time you set Last Number Redial, until the call gets through.
• If the alternative number is busy, the system will redial another alternative number.
• The system will dial a different (alternative) number on each auto redial attempt3, until the call gets
through.
3. The number of auto redial attempts depends on the Auto Redial Count configured in the system. By default, the system will make
5 redial attempts if Auto Redial 'normal' is set. If Auto Redial 'Priority' is set, the system will make 20 redial attempts.
• when any of the alternate numbers gets through, the system will give a ring on your extension.
(Busy) 2630555
( Bu
Calling Party sy)
2630556
2630557
ETERNITY
• Alternate Number Dialing will work only on extensions that are allowed the features Last Number
Redial in their “Class of Service (COS)”
• Also, Alternate Number Dialing will work only for those numbers that exist in the Global Directory
assigned to each extension. The Global Directory is divided into three parts, 100-399 (Part 1), 400-699
(Part 2), and 700-999 (Part 3). If an extension is assigned only Global Directory Part 2, Alternate
Number Dialing will work only for those numbers grouped as Alternate Number Groups in Global
Directory Part 2.
• Alternate Number Dialing will work also with “Abbreviated Dialing”. For example, an extension user
dials the abbreviated code 8100, and the dialed out number is busy. When the extension user sets
Redial or Auto Redial, the ETERNITY will try the alternate numbers related to 8100.
How to configure
To use Alternate Number Dialing, you must do the following:
4. Enable the features 'Last Number Redial', 'Global Directory', in the Class of Service (CoS) group of the
extensions to which Alternate Number Dialing facility is to be provided. If desired, 'Auto Redial', 'Auto
Redial Priority' may also be enabled in the CoS of these extensions.
• To create Alternate Number Groups, the alternate numbers must exist in the Global Directory. If any of
the alternate numbers do not exist in the Global Directory, first configure the numbers in the directory,
before you begin creating Alternate Number Groups. Refer the topic “Abbreviated Dialing” for
instructions on configuring the Global Directory.
• Write the name of the contact on one column and the Alternate Numbers for the contact on the other
column.
• Make a list of the numbers which need to be grouped as alternate numbers. For example:
• Taking the above example further, the Alternate Number Groups on the list may be numbered as follows:
To create Alternate Number Groups and configure them in the Global Directory,
• Open Jeeves.
• Click the Global Directory link under Abbreviated Dialing to open the page.
• Double click the Route Global Directory’s numbers using Trunks field, and select the trunks to be used
for routing the numbers of the Global Directory dialed out by extensions.
• Enter the contact’s Number you wish to store. The number may consist of 16 digits (maximum).
• Enter the contact Name against the Number. The Name may contain up to 12 characters (maximum).
• In the Alternate Number Group column, enter the number of the Alternate Number Group you assigned
to the numbers of you con.
For example, you have assigned Alternate Number Group '001' to all the numbers of the contact Midas
Business Solutions, enter this number against each number belonging to this contact.
Similarly, enter Alternate Group number '004' against the numbers belonging to the 'GoodLife Inn' to which
it is assigned.
Memory Alternate
Number Name
Location Number Group
: : : :
The numbers of the contacts may not necessarily appear alphabetically or in a sequence. It is possible that
the numbers of the same contact may be configured at different memory locations in the Global Directory.
In the above example, one number of the GoodLife Inn is entered at memory location Index 104 and the
other on Index 129. Since these two numbers are grouped and assigned the number alternate group
number '004', this number must be entered against the GoodLife Inn numbers at the respective memory
location Index.
• After assigning Alternate Number Groups, click Submit button to save entries.
• Enable the features 'Last Number Redial' and 'Global Directory', in the Class of Service of the extensions
to which Alternate Number Dialing facility is to be provided. If desired, 'Auto Redial', 'Auto Redial Priority'
may also be enabled in the COS of these extensions.
By default, all extensions are allowed Last Number Redial in their Class of Service. However, only Global
Directory Part 1 is enabled by default in their COS.
Alternate Number Dialing will work only for those numbers that exist in the Global Directory assigned to
each extension. So, the Global Directory Part containing the Alternate Number Groups must be allowed to
Refer the topic “Class of Service (COS)” for instructions on configuring COS of extensions.
How to use
Confirm with your System Engineer that
• Alternate Number Groups are configured in the Global Directory allowed to your extension.
• 'Basic Features' (these include Redial) are enabled in the Class of Service allowed to your extension.
Now, follow the instructions for using the feature “Last Number Redial”.
What's this?
Auto Answer allows incoming calls to be answered without any manual interventions by the extension users.
This feature is particularly useful for Operators in high call traffic settings, as it saves them the effort of picking up
the handset or pressing the speaker key repeatedly.
How it works
With Auto Answer set on an extension DKP, whenever a call lands on the DKP extension,
• the extension rings for the duration of the Auto Answer Timer4. This timer is configured, and by default it is
set to 1 second.
• on the expiry of the Auto Answer Timer5, the system plays a beep to the user.
• the DKP goes OFF-Hook to answer the call, without any intervention by the extension user such as picking
up the handset or pressing the speaker or the headset key.
• If a headset is connected, and headset connectivity is enabled on the DKP, the incoming speech audio will
be diverted to the headset automatically.
Auto Answer works only if the DKP is in idle state; the phone must not be busy with an active call or using a feature.
How to configure
For Auto Answer to work, you are required to do the following:
• Change Auto Answer Timer, if required. The range of this timer is 1 to 9 seconds. By default, the Auto
Answer Timer is set to 1 second.
• Enable Headset Connectivity flag on the DKP, if headset is to be used for Auto Answer.
You can do the above configuration using Jeeves, or DKP extension users can also configure the above
parameters using the Phone Menu of EON. See “How to use” Auto Answer later in this topic.
4. This timer defines the time in seconds that the DKP should wait before going OFF-Hook to answer incoming calls.
5. This timer defines the time in seconds that the DKP should wait before going OFF-Hook to answer incoming calls.
• Now, set the Auto Answer Timer to the desired duration. By default the Timer is set to 3 seconds.
• To configure Auto Answer on the second DKP, click the DKP extension tab and repeat the above steps.
How to use
Extension users can set/cancel Auto Answer and enable Headset connectivity from their DKP/Extended IP Phone
by navigating the Menu of the phone.
OR
It is recommended that Auto Answer Timer be set to at least 2 seconds.
OR
What's this?
If the extension number you have dialed is busy or is not responding, you may use the Auto Call Back feature,
instead of repeatedly dialing the number. Similarly, when you dial a code to access a trunk and the trunk is busy,
you may set Auto Call Back.
How it works
When you set Auto Call Back,
• As soon as both extensions, yours and the remote extension, are available, the system will ring first on
your extension for the duration of the Auto Call Back Ring Timer. This timer is set by default to 30 seconds
and is configurable.
• When you go OFF-Hook, the system will ring on the remote extension (provided it is also available at that
moment) for the duration of the Auto Call Back Ring Timer.
• When the remote extension user goes OFF-Hook, your call will get connected.
However, if the remote extension gets busy before the system can ring on it, the system will continue to try
again.
Auto Call Back set for a busy trunk works the same way. As soon as the busy trunk port you are trying to
access is available, the system will ring your extension. When you go OFF-Hook you will be connected to
the trunk port.
• Each extension of the ETERNITY can set only one Auto Call Back request at a time. If you set another
Auto Call Back request, before the first one has been served, the system will override the first request
and serve the second.
• The ETERNITY NE has the capacity to serve 50 Auto Call Back requests from its extensions at a time.
The service duration for each request is 60 minutes. Requests that are not served within 60 minutes
are automatically cancelled by the system. Also, the system will not serve any more requests if all the
50 requests are pending. In such a case, the system will play an error tone, when an extension
attempts to make a request.
Auto Call Back request set by you will be cleared by the system, if:
• it was successfully served, i.e. your extension was connected to the remote extension or the trunk you
were trying to reach.
• you do not answer the Auto Call Back ring, before the expiry of the Ring Timer, i.e. within 30 seconds
(default setting).
• the remote extension does not answer the Auto Call Back ring before the expiry of the Ring Timer.
• Auto Call Back works for internal calls and for accessing trunk ports only.
How to configure
Auto Call Back is a “Class of Service (COS)” dependant feature. An extension user can set/cancel Auto Call Back
only if it is enabled in the extension's Class of Service.
The only configuration involved in this feature is enabling/disabling Auto Call Back in the Class of Service and
changing the duration of the Auto Call Back Ring Timer, if required.
By default, all extensions of ETERNITY NE are allowed Auto Call Back feature for the day and night in their Class
of Service. So, all extensions of the ETERNITY can set/cancel Auto Call Back if the called number is busy or does
not reply.
However, if Auto Call Back Busy/No Reply is to be denied to any of the extensions, you may this feature in the class
of service of these extensions. For instructions on enabling/disabling this feature on the different extension types,
refer the topics “SLT Extensions”, “DKP Extensions”, “SIP Extensions”under Basic Settings.
If you want to increase or decrease the duration of the of the Auto Call Back ring on both extensions, i.e. the
extension requesting Auto Call Back and the destination extension, configure the 'Auto Call Back Ring Timer'.
• Open Jives.
• Click the System Timers and Counts link to open the page.
How to use
Extension users can set two types of Auto Call Back:
• Auto Call Back on Busy - when the extension/trunk they are trying is Busy.
• Auto Call Back on No Reply - when there is no reply from the extension they are trying.
• On Busy Tone.
• Dial 2.
• You get confirmatory tone
• Replace handset.
If you hear an error tone while setting an Auto Call Back request, it is likely that the system already has 50
pending requests and is unable to accept yours.
What's this?
The Auto Redial feature retries a call automatically if the dialed number is busy. It repeatedly checks the busy line
till it is free. When the called number is no longer busy, the extension of the caller rings.
Auto Redial saves time and the effort of repeatedly dialing the entire phone number over and over until the called
party gets off the phone.
How it works
When an extension user dials a number and gets a busy tone, the user may set Auto Redial. When Auto Redial is
set,
• It waits for the 'Dial Tone Wait Timer8' to expire to begin sensing the dial tone from the CO Network. This
timer is configurable, and is set to 3 seconds as default.
• On sensing the dial tone the ETERNITY NE will dial out the requested number and will wait until the 'Ring
Back Tone Wait Timer9' expires to sense the Ring Back Tone from the requested number. This timer is
configurable and is set to 60 seconds as default.
• If the system does not detect Ring Back Tone for 60 seconds, it releases the trunk and tries again after
some time. If the system detects a busy tone, it releases the trunk and redials the number automatically
after some time. This process is repeated until the system detects the Ring Back Tone.
• When the ETERNITY NE detects the Ring Back Tone instead of the Busy Tone, it will ring on the
extension that set Auto Redial. The extension will ring for the duration of the 'Redial Ring Timer10'. This
timer is configurable and is set to 45 seconds as default.
• If the extension is in the middle of any activity such as dialing, ringing or speech, the ETERNITY NE will
suspend Auto Redial until the extension becomes idle again. After which it dials the requested number
again.
Two types of Auto Redial are supported by the ETERNITY NE - Auto Redial (normal) and Auto Redial 'Priority' -
that differ from each other in terms of the number of redial attempts and the interval between attempts.
8. Time for which ETERNITY waits to sense the dial tone from the PSTN/CO Network. Valid range of the timer: 000 to 255 seconds.
Default: 003 seconds.
9. Time for which ETERNITY waits to sense the RBT from the PSTN/CO Network after dialing the requested number. This timer is
particularly relevant to CO ports. Valid range of the timer: 000 to 255 seconds. Default: 060 seconds.
10. Time for which the extension that has requested Auto Redial should ring. The valid range of the timer: 000 to 255 seconds. Default:
045 seconds.
• Auto Redial 'Priority': the system makes a greater number of attempts to redial and the duration of the
interval between each attempt is less. By default, the system is configured to make 20 redial attempts at
intervals of 20 seconds. The number of attempts as well as duration of the interval are configurable, e.g.:
number of attempts can be set to 30 and the interval to 15 seconds.
To change the number of redial attempts and the interval between them, you must configure the Auto Redial Count
and the Auto Redial Timer respectively. In addition to these, the system has three other related timers, which can
be configured to match User preference:
• An extension user can request Auto Redial for multiple numbers at a time from the same extension and
more than one extension can attempt auto redial simultaneously.
• The system uses the same trunk access code you used for dialing the number. If you dialed ‘0’, the
system grabs one of the free trunks selected for dialing ‘0’.
• If the number was dialed the first time using selective trunk access, the system will use the same trunk
for Auto Redial.
• If the extension has 'Dynamic Lock', and you have set the 'Auto Redial', the system will check the Toll
control as per dynamic lock level.
Auto Redial may not work well on CO trunks, as its functioning greatly depends on line condition. Unlike
Mobile and SIP trunks, the line condition of Analog trunks may not always measure up to the standard
requirement for Auto Redial to function.
How to configure
For Auto Redial to work, the you must:
• Enable the features 'Auto Redial' and 'Auto Redial Priority' in the “Class of Service (COS)” of the
extensions to which this feature is to be allowed.
By default, none of the extensions of ETERNITY NE are allowed Auto Redial and Auto Redial Priority in
their Class of Service. You must enable it to allow these features to the extensions.
For instructions on enabling/disabling this feature on the different extension types, refer the topics “SLT
Extensions”, “DKP Extensions”, “SIP Extensions”under Basic Settings.
• Change the 'Auto Redial Normal/Priority Count' and the 'Auto Redial Normal/Priority Timer' to match User
preference. This will change the number of redial attempts made by the system and the interval between
them.
• Open Jeeves.
• Click the link System Timers and Counts on the left side panel to open the page.
• Change the Count and Timer of the type of Auto Redial - Normal or Priority - you have set.
• You may change any of the related timers - Auto Redial Dial Tone Wait Timer, Auto Redial Ring Back Tone
(RBT) Wait Timer, Auto Redial Ring Timer - as per your preferences on this page.
What's this?
ETERNITY NE offers connectivity to different networks—CO, GSM/UMTS, VoIP—each having a different
numbering plan. For example, the GSM/UMTS network requires area codes to be dialed also for local numbers,
whereas CO requires dialing of area codes for long distance calls.
When ETERNITY NE is connected to more than one network, outgoing calls may be routed through any of these
networks, depending on the outgoing call routing configured in the system. However, as extension users do not
know through which telecom network their calls will be routed, they cannot be expected to dial numbers according
to the numbering plan of the destination networks.
The feature, Automatic Number Translation of ETERNITY NE takes care of this. It modifies the dialed numbers or
part thereof to match with the specific route numbering plan understood by the destination network (CO, GSM/
UMTS VoIP). This includes adding or stripping of country codes and area codes.
For example, when an extension user dials a local landline number, Automatic Number Translation can be
configured such that the ETERNITY prefixes the number with the appropriate country-area code when it routes the
call through the GSM/UMTS network.
How it works
Automatic Number Translation makes use of the Automatic Number Translation Table which comprises:
• Substitute Number Strings - the corresponding numbers for the Dialed Numbers that the system will dial
out as the destination numbers instead of the dialed numbers.
The Dialed and Substitute Number strings must be configured, and the Table must be applied to the trunk ports,
through which calls are routed to the destination networks. The trunk ports may be CO Trunks, Mobile, SIP,
depending on system configuration.
You can configure different 8 Automatic Number Translation Tables; each table accommodates 32 Dialed Number
strings and their corresponding Substitute number strings. A trunk port can be assigned only 1 Table.
For example:
• Automatic Number Translation (ANT) Table-1 has '95' stored as Dialed Number string in Index-1 and '91'
as the corresponding Substitute Number string at Index-1.
How to configure
The working of the Automatic Number Translation feature is controlled by two parameters: 'Automatic Number
Translation Table' and 'Automatic Number Translation flag' in the outgoing trunks.
You must first configure the Dialed Number strings and the Substitute Number strings in the Automatic Number
Translation Table, and assign the configured Table to the trunks, on which you want to enable this feature.
Decide the number of ANT tables you need to configure. You can configure 8 different ANT Tables, with a
maximum of 32 number strings in each.
For your convenience, draw three-column tables on paper. In each table, enter the Index numbers 1 to 32 in the
first column, Dialed Number in the second column, and Substitute Number in the third column.
Enter the Dialed Numbers and their corresponding Substitute Numbers against each Index number. For instance, if
you entered '95' as the Dialed Number at Index 1 and its Substitute Number string '91' must be entered at Index 1.
Decide which of the trunks are to be assigned the Automatic Number Translation feature and configure Automatic
Number Translation on those trunks.
• Open Jeeves.
• To configure another table, click the Table Number tab and enter the dialed and substitute number strings.
• On the trunk page, click the Automatic Number Translation (ANT) option to expand.
Select OFF if you want to disable ANT on this trunk. Default: OFF
You can also configure the ANT Table number you select for this trunk from this page by clicking the arrow
icon. This will open the ANT Table Number you selected for this trunk.
What's this?
Barge-In allows you to break into an on-going conversation between two extension users, between an extension
user and an external caller as well.
Barge-In can be used by Operators to transfer Incoming calls to busy extensions. The Operator can put the caller
on hold, barge into the busy extension to inform about the call, and then transfer the call.
ETERNITY offers flexibility to allow/deny Barge-In feature to an extension user, i.e. allow the extension user to
barge into on-going conversations. It also provides the flexibility to prevent conversations of extension users from
being barged in, referred to as Privacy against Barge-In.
How it works
• A, B and C are users of the system.
• C calls A.
• C gets RBT and A gets beeps for Barge-in timer. (By default, 10 seconds)
• If A does not respond till the end of the Barge-In Timer (set to 10 seconds, by default), A gets connected to
C. B is put on hold and is given hold-on music.
• If B disconnects while A and C are in speech, the held call between A and B is cleared.
• If B keeps holding the call and C disconnects, the call between A and C is cleared and A will be in speech
with B.
• After Raid, if A goes ON-hook, before the expiry of the Barge-In Timer, C gets RBT and A gets a ring.
Feature Interactions
• Call States:
• Barge-In works only if the dialed extension is busy. The dialed extension may be busy with another
extension or trunk (external number).
• It will not work if the busy signal is due to the user being OFF-Hook, or in the middle of dialing, or
accessing a feature of the PBX.
• “Call Toggle”: Once A and C comes in speech with each other, A can toggle between B and C using Call
Toggle feature.
• Privacy against Barge-In: If the feature 'Privacy against Barge-in is enabled for an extension, it cannot be
barged into.
• “Priority”: No Interaction with Barge-In. If 'A' has lower priority than 'B' but has Barge-In enabled; A can
barge in B.
• “Do Not Disturb (DND)”: Barge-In will not work if the called user has set DND. If 'A' has set DND. A is
busy with C. B calls A. B cannot barge in A.
How to configure
The functioning of this feature is controlled by three parameters, 'Barge-In', 'Privacy against Barge-In' and 'Barge-In
Timer'.
By default, all extension port types have Barge-In and Privacy from Barge-In disabled in their Class of Service.
While it makes sense to offer all extensions Barge-In, providing Privacy from Barge-In also to all extensions will not
serve the purpose of Barge-In.
Decide which extensions are to be allowed 'Barge-In', and which are to be allowed 'Privacy from Barge-In' and
enable these features in their Class of Service for the Day and Night.
Barge-In Timer
Barge-In Timer is the time after which the caller gets connected to the called party. By default the Timer is set to 10
seconds. To configure the Barge-In Timer, see Other Features under “System Timers and Counts”.
How to use
11. This default feature access code can be changed to suit your preference. Refer the topic “Access Codes”.
What's this?
BCCH Selection feature enables you to lock the Mobile Port of ETERNITY to a particular cell or channel or BTS
(Base Transceiver Station) to ensure better network availability, and minimize call drops due to bad signal/ network
failure, etc.
This feature is supported when SIMCOM-2G engine (SIM340-B01 or higher) or SIMCOM-3G engine is
installed in ETERNITY’s Mobile Card.
How it works
In the GSM network, each BTS is assigned one particular channel called as ARFCN (Absolute Radio Frequency
Channel Number), which is transmitted by BTS in BCCH (Broadcast Control Channel).
Now, when ETERNITY is switched on, the Mobile Port gets registered with the network on a particular BTS which
has the highest signal strength. However, the signal strength is not consistent. It keeps fluctuating, resulting in call
drop or poor voice quality.
Therefore, to avoid this, ETERNITY enables you to lock the Mobile Port to a particular cell or channel manually
after checking Signal Strength and Signal Quality of each cell.
How to configure
You can lock Mobile Port to a cell or a channel only through Jeeves.
• Mobile Port Number: This is number of the Mobile port for which BCCH Selection status is displayed.
You can choose another Mobile Port number from the box. The page will display the BCCH Selection
related parameters for the selected mobile port.
• Mobile Port Name: This is the name you have assigned to the Mobile port in the Mobile port
parameters.
• Mobile Port Status: The current state of the Mobile Port is displayed in this field. Given below is the
description of the various status indication messages that will appear in this field.
STATUS DESCRIPTION
GSM Displayed when GSM module is in initialization state i.e. before SIM
Initialization detection.
SIM Absent Displayed when SIM Card is not detected by the system.
Registering Displayed when the Mobile Port is in registration process with the Network.
Idle Displayed when the Mobile Port is registered with the Network and it is
free.
Busy Displayed when any active call is present on the Mobile Port.
• BCCH Locking Status: The current BCCH Locking status of the mobile port is displayed in this field.
STATUS DESCRIPTION
Trying to Lock Displayed when user selects Manual BCCH Locking as 'No' from 'Yes' and
module is in initialization process after system or module restart.
Trying to lock on Displayed when BCCH Locking is selected as Manual and the Mobile Port is in
BCCH xxxxx the registration process with the Network. xxxxx is the BCCH selected by the
user for locking the cell.
Manually Locked Displayed when BCCH Locking is selected as Manual and Mobile Port is
on BCCH xxxxx successfully registered with the Network. xxxxx is the BCCH selected by the user
for locking the cell.
Auto Locked on Displayed when BCCH Locking is selected as Auto and Mobile Port is
BCCH xxxxx successfully registered with the Network. xxxxx is the BCCH of the Main Cell.
xxxxx is updated as per the changes in the Main Cell's BCCH.
• Main Cell- Bit Error Rate (%): Bit Error Rate of the Main Cell is displayed in this field. Bit Error Rate
(BER) is the percentage of received bits on a digital link that are in error relative to the number of bits
received. Bit Error Rate is calculated from the received signal quality.
• Manual BCCH Locking: This parameter allows you to lock the Mobile Port to a particular cell of your
preference. By default, manual BCCH locking is set to 'No'. When manual BCCH locking is set to 'No',
Mobile Port gets locked to the cell as per the highest signal strength. Select 'Yes' if you want to lock the
Mobile Port to the particular cell selected by you.
• Auto Refresh: By clicking this button, BCCH Selection page is refreshed automatically and all its
parameters are downloaded automatically after every 15 seconds. By default, Auto Refresh button is
enabled.
• Stop Auto Refresh: By clicking this button, you can stop the system from automatically refreshing the
BCCH Selection page every 15 seconds. When you stop Auto Refresh, you must click the link 'Refresh' at
the bottom of this page to refresh the page whenever you want
• Cells: Indicates the cells with which the Mobile Port can be locked. You can decide to lock the Mobile Port
with a particular cell after considering the following cell related parameters, which appear on the page:
• MCC-MNC: In this field, MCC-MNC of a cell is displayed. Mobile Country Code (MCC) is a three digit
number uniquely identifying a country and Mobile Network Code (MNC) is either a two or three digit
number used to identify a given network from within a specific country.
• LAC (Location Area Code): In this field, LAC (Location Area Code) is displayed. LAC uniquely
identifies a location area within a GSM PLMN (Public Land Mobile Network). The maximum length of
LAC is 16 bits ranging from 0 to 65535. LAC is displayed in hexadecimal characters for SIMCOM-2G.
For SIMCOM-3G engine, LAC is displayed in decimal digits which ranges from 00000 to 65535.
• Cell ID: In this field, Cell ID is displayed. It is a 16-bit identifier that identifies the cell. Cell ID is
displayed in hexadecimal characters for SIMCOM-2G engines which ranges from 0000 to FFFF. For
SIMCOM-3G engine, Cell ID is displayed in decimal digits which ranges from 00000 to 65535.
• BCCH (Broadcast Control Channel): In this field, the BCCH value of the cell is displayed. BCCH
defines the frequency channel number.
• Receive Level: In this field, the Receive Signal Strength level of the cell is displayed. It is the average
Receive Signal Strength of the cell. Its value ranges from -110 dBm to -47 dBm.
• Manual Cell Locking: This radio button is for locking a Mobile Port to a selected cell manually.
• Select the desired Mobile Port Number (if your system has two Mobile Ports).
• Set the parameter Manual BCCH Locking to 'Yes'.
• Go to the Cell to which you want to lock the Mobile Port you selected.
• The BCCH Locking for the selected Mobile Port will appear on this page, if Auto Refresh is enabled.
• If you have stopped Auto Refresh, click Refresh at the bottom of the page to refresh the page and view the
current BCCH Locking settings of the selected Mobile port.
Example:
Consider the following example when using this feature:
Problem:
• ETERNITY is installed in roaming area, where more than one network is available, network A and network
B.
• Mobile Network Selection is set to 'Manual' mode, network A is selected as the first priority and network B
is selected as the second priority.
• The Mobile Port gets registered with network A. After registration, the user locks the Mobile Port to one of
the cells of network A.
• After registration, if the module or the system restarts or gets de-registered from the network, module
starts registration process again.
• While re-registering, ETERNITY tries to lock the Mobile Port to the last selected cell of network A.
Solution:
• In this situation, user should set Manual BCCH locking mode to 'No' to register Mobile Port with the
suitable network automatically.
• Later, the user can set the Manual BCCH locking mode to 'Yes' and lock the Mobile Port to the desired cell
after assessing the cell information.
What's this?
It is common for small and medium PBXs to be connected to larger PBX systems, where the trunks of the larger
PBX are connected to the extension ports of the smaller system. This is usually done for the purpose of expanding
the capacity of the large PBX already in use.
How it works
Consider the following illustration.
PBX-A is connected behind ETERNITY NE. In this 'Behind the PBX' configuration, the Trunk Lines T4, T5, T6 of
ETERNITY NE are connected to the Extensions (SLT) S4, S5, S6 of PBX-A.
However, Trunk lines T1 and T2 of PBX-A are connected directly to the CO.
In such application scenarios, implementing toll control restrictions for the trunks becomes difficult for ETERNITY
NE.
For example: Extension Number 21 of ETERNITY NE in the above illustration is not allowed the facility of long
distance dialing. It has access to all the CO trunks.
When the user of Extension 21 wants to access T1, T2 or T3 (which are direct trunks from the CO to ETERNITY
NE) the user dials '0' (or the Trunk Access Code assigned) and gets CO dial tone. When the user dials the number,
ETERNITY NE applies Toll Control.
Similarly, when the user of Extension 21 tries to grab a trunk T4, T5 or T6 (which are connected to extensions of
PBX-A) by dialing Trunk Access Code '0', the user gets the dial tone of PBX-A. This means, the user of Extension
21 must dial '0' again to grab CO dial tone of the T1 or T2 connected to PBX-A.
However, when the user dials '0' again, ETERNITY NE applies Toll Control. It detects the dialed number as ‘00’ and
interprets this as an attempt to dial a long distance number. Since Extension 21 is not allowed long distance dialing
in its Toll Control, ETERNITY NE rejects dialing on the trunk and plays an error tone to Extension 21.
The PPDC defines the number of digits to be dialed by the extensions to reach the CO. The system applies Toll
Control check on extensions only after checking the PPDC configured for the CO.
On trunks that are connected to another PBX, in this case, T4, T5, and T6, PPDC must be configured with the
same number of digits as the Trunk Access Codes assigned for PBX-A. For example, if the Trunk Access Code is
a single digit number '0', the PPDC on must be configured as '1'. If the Trunk Access Code is a two-digit number,
61, the PPDC should be configured as '2'.
On trunks directly connected to the CO, i.e. T1, T2, T3 of ETERNITY NE, the PPDC must be configured as ‘0’.
To take the above example further, when PPDC is configured as ‘1’ on T4, T5 and T6, when the user of Extension
21 dials ‘0’ followed by another ‘0’ to grab T1 or T2 trunk of PBX-A, the system will check the PPDC configured on
the trunk. On finding ‘1’ the system will ignore the first 0 dialed by the extension, and let the extension user grab T1
or T2 by considering the second ‘0’. The Extension user will get the CO dial tone from T1 or T2 of PBX-A.
How to configure
The PPDC should be configured only for 'Behind the PBX Applications'. For all normal applications, this
count must be set to '0' for all the trunks. Otherwise, external number dialing may be hampered. Features
like Least Cost Routing and Station Message Detail Recording will also be affected.
For CO Trunks that are directly connected to the CO, PPDC must be configured as ‘0’.
For CO Trunks that are connected to the extensions of another PBX, PPDC must be configured as per the number
of digits in the Trunk Access Codes defined for the second PBX.
• Click the number of the CO trunk you want to assign PPDC ‘0’ (i.e. trunk connected directly to the CO) by
clicking on the CO trunk number tab.
• Similarly, click the tab of the CO trunks you want to assign PPDC count from 1 to 6 (i.e. trunks connected
to the extensions of another PBX, and assign the appropriate PPDC count to the trunk. This would depend
on the number digits in the Trunk Access Code defined for the trunks in the other PBX. If the TAC is single
digit, select '1'. If TAC is double or triple digit, select '2' or '3' as applicable as the PPDC.
What's this?
The feature Call Back on Trunk Ports is used to respond to missed calls from particular numbers on the different
trunk ports of ETERNITY: Two-wire Trunks, Mobile trunks and SIP Trunks.
When the Call Back feature is enabled on a trunk port, and there is a missed call on that trunk port, the ETERNITY
determines if the calling number is eligible for a call back or not. It calls back the same number or an alternative
number configured for that number, either from the port on which it was received or from a different port, depending
on the configuration. ETERNITY can be configured to choose the most cost effective line to call back the missed
call numbers.
Employees at remote locations can use this feature to have the ETERNITY installed in their office call them back,
thereby saving on charges (e.g. roaming charges on mobile calls), where applicable.
How it works
For this feature to work:
• The ‘Call Back Timer’ may be configured. When the caller disconnects within the Call Back Timer, the Call
Back will be applied for that number.
• You must define ‘Call Back to’, i.e. you must select whether the number which must be called back should
be the same CLI number as on the received call or an alternative number.
• The CLI of those callers whom the system should call back as well must be configured in a List. Also, if you
want the call back to be made to an alternative number, the alternative number must also be configured in
a list.
• You must select the trunks through which the call back should be made. If necessary, you may enable
Least Cost Routing on the outgoing trunks you selected for call back.
• Select a ‘Call Back Mode’, i.e. how the call should be routed when the call back is answered by the remote
party; whether it should be routed as Operator, DID, or DISA.
Following is an example of a Call Back on a mobile port, when the above parameters are configured.
• The system matches the CLI of A with the Call Back profile assigned to the mobile port 1 to determine if
the calling number is eligible for a call back.
• A must disconnect before the expiry of the Call Back Timer so that the system can treat it as a Missed Call.
• If A disconnects within the Call Back Timer, the system applies Call Back for A’s number.
• The system checks the ‘Call Back to’ parameter in the Call Back profile, to determine whether it has to call
back the same number or an alternative number.
• If alternative number is configured as ‘Call Back to’, the system checks the list of alternative numbers. As
the CLI of A matches with the number in the list, the corresponding alternative number configured for A’s
CLI number.
• The system checks if the number is to be called from the same trunk port or a different group of trunks.
• If the same trunk port is configured as outgoing route, the system will make a call to the number using
mobile port 1.
• If a different group of trunks has been assigned, the system will check if Least Cost Routing is enabled on
these trunks and accordingly make the call back accordingly.
• The system checks the type of Call Back Mode enabled in the Call Back Profile assigned to mobile port 1
(the port on which the call back request was made).
1. “Direct Inward Dialing (DID)” is enabled as Call Back Mode on mobile port 1.
2. 'Pin Authentication - Multiple Calls' or 'CLI Authentication - Multiple Calls' is enabled as Call Back Mode
on mobile port 1.
• A can now reach any extension or trunk of ETERNITY from DISA Mode.
3. 'CLI Authentication - Single Call Answer Signaling' is enabled as Call Back Mode on mobile port 1.
12. If the system does not find a match for the CLI of the caller in the list, the 'Call Back' feature will not be appli-
cable and the call will be processed according to the normal incoming call logic.
Read the topics “Direct Inward Dialing (DID)”, “Direct Inward System Access (DISA)” and “Configuring
'Operator'” to know more about the call respective call logic.
• Since this feature is essentially for callers, they must be aware of its functioning to be able to use it, i.e.
disconnect the call within the Call Back Timer. If the caller does not disconnect within the Call Back
Timer, the call will be processed according to the normal incoming call logic.
• ETERNITY NE supports only one call back request at a time, for one trunk port. The second incoming
call on that trunk port will be processed by the system as per normal incoming call routing.
• For call back requests made from a group of trunks, when any of these trunks is busy, ETERNITY will
support only the last call back request in the group. Previous requests will be processed as per the
normal incoming call management logic.
How to configure
For this feature to function, you must configure the Call Back Profiles and assign them to each Trunk port type (CO,
Mobile, SIP) on which you want to use this feature.
• The CO trunk page opens. Select the number of the CO trunk you want to configure by clicking the trunk
number tab.
• Now, configure the Call Back Profile you selected by clicking the settings icon.
• Set the Call Back Timer to the desired duration. This is the duration for which the system waits for the
caller to disconnect the call after the system has found a matching number for the caller’s CLI in the list
you configured.
When the caller disconnects within Call Back Timer, the system applies Call Back on the port. If the caller
does not disconnect within the Call Back Timer, the incoming call management logic is applied for the call
on the trunk port.
• Select the number which the system must Call Back To. You may select:
• Call Back on same number from which missed call is received. If you select this option, you must
configure the numbers eligible for call back in the list Program the numbers for which call back is to
be provided.
• Call back on alternative number if programmed. If you select this option, for each number you
configured as eligible for call back, you must configure an alternative number in the list Program
alternate number in respective row.
In countries where CLI received on trunks can be dialed out without any modification, you may select
‘CLI Number’ as ‘Call Back to’ option.
In countries where CLI received on trunks cannot be dialed without modification, you may select
“Alternate Number’ as the ‘Call Back to’ option. You may also select ‘Alternate Number’ as Call Back
on when you want the call back to be made to a different number.
• Configure Trunk Selection on Call Back. This parameter determines the trunk port to be used to make
call back.The call back can be made using the same port or a group of trunks.
• Select Call Back from same port from which missed call is received, to use the same port to make
the call back.
• To make the call back using a group of trunks, select Call Back from trunks, and double click this field
to select the trunks.
On the left, the trunks appear with their names (if configured in “Naming Trunks”) and port numbers in a
sequence, starting with CO trunks, followed by Mobile trunks and SIP trunks.
• To select a trunk, place your cursor on the desired trunk, and click the Select>> button.
Or
• Press the ctrl key and click the left mouse button to select multiple trunks.
• You may change the sequence of the trunks you selected, if required, using the Up and Down arrow
buttons on the right display box.
• You can also delete trunks from the ones you have selected.
• You may enable Rotation, if you have selected more than one trunk. Default: Disabled.
• To apply Least Cost Routing on the trunks, select desired LCR method from the combo box: Number-
based, Time-based, Time+Number based, Serivce Provider based. Configure LCR method that you
selected for the trunk group, if not already done.
• Select the Call Back Mode, from the options; how a ‘Call Back’ call answered by the remote party should
be routed:
• Operator: When the remote party answers the Call Back call, the system will route the call to the
Operator13.
• DID: The system will process the call as per the DID call logic - give a dial tone to the remote party, who
can now call any extension. Refer the feature description for “Direct Inward Dialing (DID)”.
• PIN Authentication-Multiple Calls: The system will process the call as per DISA call logic - allow
remote party to enter DISA mode with PIN-Authentication. On successful authentication (DISA Login)
the user is allowed to make calls or use features as allowed to him/her.
• CLI Authentication-Multiple Calls: The system will process the call as per DISA call logic, allowing
the remote party to enter DISA mode with CLI Authentication-Multiple calls as authentication method
and level of access.
• CLI Authentication-Single Call-Answer Signaling: The system will process the call as per DISA call
logic, allowing the remote party to enter DISA mode with CLI Authentication-Single call as
authentication method and level of access. Refer the feature description for “Direct Inward System
Access (DISA)”.
• In the column Program the numbers for which call back is to be provided, enter the numbers that
are eligible for call back. The system checks the CLI of the caller with this list to determine if the caller
is eligible for a call back.
Number strings configured in this list are compared with the actual received CLI.
The number string configured in this list may be shorter than the number string received as CLI, but
only if the configured number string completely matches with the received CLI from the right towards
left, the system will consider it as a complete match.
For example, if the configured string is 263055 and the number string received in the CLI is
2652630555, the system will consider it a complete match. If the received CLI 912652630555, the
system will consider this caller too as eligible for a call back. Thus any CLI received with 263055 as the
last 7 digits will be considered as match found.
• In the column Program alternate number in respective rows, enter the numbers that the system
must call back. When the system finds a missed call eligible for a call back, it will make the call back on
the basis of the ‘Call Back to’ option you selected and the alternate number list.
• you have selected ‘Call back on same CLI number on which missed call is received’ as the ‘Call
Back to’ option, BUT the CLI received cannot be dialed out without modification. In such a case,
modify the CLI received and enter the string in the alternate number List.14
OR
• you have selected ‘Call Back on alternate number’ as ‘Call Back to’ option. In this case, configure
the alternative number for the CLI received in the alternate number list.
The modified CLI or the Alternate number should be entered in the same row as the corresponding
received CLI number. For example, for the received CLI number string entered in row 2, enter the
corresponding modified CLI/Alternate number string in row 2.
When the CLI received matches with the number string entered in row 2 of the alternate number list,
the call back will be made using the (modified/Alternate) number stored at row 2 of this List.
If you have selected ‘Alternate Number’ as ‘Call Back to’ option, but do not want to provide alternative
numbers to call back particular callers (i.e. CLI received), configure the CLI of these callers in the first list but
keep the corresponding rows in the alternative number list blank.
• To configure Call Back on other trunk port types, go to “Mobile Trunks” and “SIP Trunks” under Basic
Settings, and follow the same instructions as above.
14. Where the CLI received can be dialed out without any modification, you do not need to configure any number string in the Alternate
number List.
What's this?
Call Budget is a cost control feature that allows you to keep a tab on the total cost of phone call made by extension
users.
With this feature, each extension can be allotted a 'budget' limit for outgoing calls, which is automatically reloaded
at the start of every month.
Long distance calls form a major part of the increased cost of telephone calls. Though excessive use or misuse of
long distance dialing can be restricted using Toll Control, there may be extension users whose nature of work
requires them to make long distance calls. Instead of denying them the facility, their telephone bill can be limited to
a certain amount using Call Budget.
With a Call Budget allotted to the extension, the user is free to make calls as long as s/he does not cross the budget
limit. Once the user exceeds the budget limit, the extension can be denied access to long distance dialing.
The extension user can be assigned a fresh budget, after which s/he can resume making long distance calls.
Call Budget can be enabled on all the extensions as well as on selected extensions. Each extension can be
assigned a different amount depending on user requirement.
How it works
When an extension allotted Call Budget makes a call,
• The system checks the current call budget amount of the extension.
• If the consumed amount is within the budget limit allotted to the extension,
• The system allows the extension to make the call as per the “Toll Control Levels” assigned to it.
• After the call ends, the system calculates and adds the call amount to the extension's account. Thus it
calculates and updates the total cost of calls made from the phone.
• If the consumed amount exceeds the budget limit allotted to the extension,
• The system allows the extension to make the call as per the Toll Control-Call Budget Consumed
assigned to the extension.
• After the call ends, the system calculates and adds the call amount to the extension's account.
• Until a new Call Budget is allocated to the extension user, the extension user can make calls only as per
Toll Control assigned for the Call Budget Consumed state.
• If the budget exceeds anytime during the month, and if no fresh budget amount is allotted, the system
allows calls to be made as per the Toll Control for Calls Allowed When Call Budget is Consumed till the
end of the month. From the 1st day of the following month, the system automatically reloads the budget
amount. The extension can now make calls.
• The Call Budget allotted to extension is valid for one month. The system automatically reloads the budget
at the start of every month.
• The budget amount can be changed or allotted afresh to extensions from the System Administrator (SA)
mode, at any time. The Call Budget allotted by the System Administrator will be reloaded in the following
month.
• Call Budget is not based on real time (online) call cost calculation. The ETERNITY calculates the call
cost only after the call has ended.
• So, if the Call Budget allotted to an extension user gets exhausted in the middle of a call, the call will
not be disconnected, though the budget is exceeded. To prevent this from occurring, you may
configure the “Call Duration Control (CDC)” feature.
• Call Budget is dependent on precise Call Cost Calculation. So, SMDR parameters and long distance
codes must be configured correctly to prevent errors in calculation.
• This feature works independent of any Call Accounting Software (CAS) installed with the ETERNITY.
• The ETERNITY will calculate cost of phone calls made by extension phones even when no call budget
is allocated15.
How to configure
To be able to use this feature, you must do the following configuration:
• Apply Call Budget on the Extensions on which you want to use this feature. For instructions see “SLT
Extensions”, “DKP Extensions”, “SIP Extensions” under Basic Settings.
• Define the Toll Control, i.e. Types of calls to be allowed when Call Budget is consumed by the
extension.To do this, select the
How to use
Call Budget amount can be allotted to extensions from the System Administrator mode, using Jeeves or by dialing
SA commands from an extension phone.
• The Extension numbers appear on the tabs on the right pane, starting with SLT extensions.
• To select a particular extension number or type, click on the tab.To select a particular extension number or
type, click on the tab.
• In the Call Budget Allotted field, enter the Call Budget Amount you want to allot to this extension.
• To allot call budget to another extension, click the extension number tab, and follow the same instructions
as above.
OR
• Dial 1072-004.
• Enter Extension Number
• Enter Call Budget Amount.
Use leading zero if amount is fewer than 6 digits.
• You get confirmatory message and confirmation tone.
• Exit System Administrator mode.
OR
What's this?
Call Budget on Trunks is an expense control feature of ETERNITY NE that enables you to keep track of the cost of
phone calls made from the different Trunks of ETERNITY NE.
With this feature, each trunk can be allotted a 'budget' limit for outgoing calls. This budget limit can be configured to
be reloaded manually each time it is exceeded or on a scheduled date, which may be either daily or a particular
date of the month.
There are three types of Call Budget limits that can be set on the trunks:
• Amount: In this type of Call Budget, a fixed amount is assigned to the trunk. By default the amount of
999999 (to be considered in the local currency) is set as Call Budget Amount on trunks. With Amount-
based Call Budget you can control the actual expense incurred on making calls from a trunk.
• Minutes: In this type of Call Budget, a fixed number of Minutes are assigned to the trunk. By default,
999999 minutes are assigned as Call Budget Minutes on trunks. This type of Call Budget is useful when
the Service Provider offers 'Free' minutes. For example, the Service Provider allows the customer to make
calls for the first 1000 minutes every month. The customer can avail of this offer using Minutes-based Call
Budget on the trunk port.
• Number of Calls: In this type of Call Budget, you can define the maximum number of calls that can be
made from a trunk. By default, the maximum number of Call Budget - Calls is set to 9999 calls on the
trunks. This type of Call Budget is useful when the Service Provider offers a certain number of free calls or
a certain number of free calls for a fixed period. For instance, the Service Provider offers 150 free calls per
month.
With a Call Budget allotted to a trunk, the users can make calls from the trunk as long as the budget limit set for the
trunk (i.e. the Amount or Minutes or the maximum number of Calls) is not crossed. Once the budget limit is
exceeded, the trunk gets disabled automatically and no outgoing calls are allowed to be made from the trunk.
The consumed Budget can be reset, after which it becomes functional again and allows outgoing calls to be made.
The consumed Call Budget can be reset manually, i.e. anytime, as required/desired, or on a scheduled date either
daily or on a particular date of the month.
How it works
Call Budget can be enabled on all trunk port types - CO, Mobile, SIP. Each trunk can be assigned a different Call
Budget, depending on the requirement of the users.
When Call Budget is enabled on a trunk port, for each outgoing call,
• The system checks the type of Call Budget set on the trunk - Amount, Minutes or Number of Calls.
• At the end of each outgoing call made from the trunk, the system will calculate the cost of the call on the
basis of the Pulse Rate Type configured. The system will thus calculate the total amount consumed after
the end of each call. Refer the topic “Call Cost Calculation (CCC)” to know more.
• With the number of Minutes defined, at the end of each call, the system will calculate the duration of the
call on the basis of the units configured in the Pulse Rate. The system will calculate the consumed minute
on the basis of the duration of the call. Refer the topic “Call Cost Calculation (CCC)” to know more.
• With the number of calls configured, the system will maintain a count for the number of matured outgoing
calls made from that trunk port.
• Thus for each matured call, the Number of Calls-Count increments, irrespective of the actual duration of
the matured call.
• When the assigned 'cost' or 'minutes' or 'number of calls' assigned to trunk is exhausted, ETERNITY will:
• The consumed Call Budget Amount/Minutes/Calls can be reset manually at any time from the System
Administrator mode or the System Engineer mode or can be configured to be automatically reset either
daily or on a particular date of the month.
• The current Call Budget Amount/Minutes/Calls limit can be changed from the System Administrator (SA)
mode, at any time. If scheduled reset of consumed Call Budget is configured, then the Call Budget allotted
by the SA will be reloaded on the scheduled date.
• Once a new Call Budget is allocated to the trunk, outgoing call facility is resumed on the trunk.
• Call Budget on Trunks is not based on real time (online) call cost calculation. The ETERNITY NE
calculates the call cost only after the call has ended.
• If the Call Budget allotted to a Trunk Port gets exhausted in the middle of a call, the call will not
disconnected, though the budget is exceeded.
• Call Budget on Trunks is dependent on precise Call Cost Calculation. So, SMDR parameters and long
distance codes must be configured properly to prevent errors in calculation.
• The ETERNITY will calculate cost of phone calls made by the trunks even when no call budget is
allocated16.
How to configure
Call Budget on Trunks is to be programmed in the Trunk Parameters of the trunk type on which you want to enable
this feature. For instructions, see “CO Trunks”, “Mobile Trunks” and “SIP Trunks” under Basic Settings.
• The consumed Call Budget on trunk can be reset from the System Engineer mode as well as the
System Administrator mode manually at any time, referred to as Manual Reset.
To reset the Call Budget on trunks, from the System Engineer mode,
• Open Jeeves.
• To reset Call Budget on CO Trunks, click CO Status link to open the page.
If you have set Call Budget to reset on a scheduled date, the date will appear in the Budget Reset
Schedule (Date) field.
• To reset call Budget, click Mobile Status link to open the page.
If you have set Call Budget to be reset on a scheduled date, the date will appear in the Budget Reset
Schedule (Date) field.
• To reset Call Budget manually, any time, click the Reset button assigned to Reset Consumed Budget.
• To reset Call Budget on SIP Trunks, click the SIP Trunk Status link.
• To reset Call Budget manually, any time, click the Reset button assigned to Reset Consumed Budget.
What's this?
Call Chaining is when an external/internal call transferred by the Operator to another extension or external number
is made to return to the Operator's extension when the conversation between the caller and the extension/external
number to which it is transferred has ended.
Call Chaining is useful situations where the Operator intervention is required after the transferred call has ended.
For instance:
• The caller needs to take an appointment or requires some information from the Operator after talking to the
desired extension.
• A marketing executive who calls his supervisor to consult on a technical problem needs to be informed
about his travel itinerary and ticket booking by the Operator. The Operator can transfer the call to the
supervisor, and use Call Chaining to retrieve the call once the conversation has ended to give the
information to the executive.
• A Direct Station Selection (DSS) Key must be programmed for this function.
• Any DKP extension can use Call Chaining, provided it has a DSS Key for this function.
How it works
• A is an External Caller
• B is an extension.
• If A disconnects the call with C, the call will be released. It will not return to the Operator.
• If the Operator is busy, A will be played music on hold for the duration of the Call Park Release Timer.
• If the Operator is busy and the Timer elapses, the call will be released.
Call Chaining can be performed when call is transferred from a DKP to DKP, SLT or Trunk.
The process of Call Chaining would be the same if A were an internal caller and B were an external number,
or if both were external numbers or both were internal numbers.
For instructions on assigning a feature to DSS keys, see “Phone Key Settings” under “DKP Extensions”.
To change the Call Park Release Timer, see “System Timers and Counts”
How to use
Call Chaining is possible from DKP or Extended IP Phone only, though the destination to which the chained call is
transferred may be an SLT or Trunk (external number).
OR
• Press Hold-1050.
• Caller gets on-hold music.
• You get feature tone and the message 'Called Party in Chaining' on your phone's display. If DSS Key is
used, the LED of the key will glow.
• Dial the requested extension/external number.
• You get Ring Back Tone.
• The called party answers.
• Perform Call Transfer17 to the requested extension/trunk.
• When extension/trunk disconnects, your extension rings.
• Go OFF Hook. You get connected with the caller.
• Go Idle after the conversation ends.
17. Refer the EON48/Extended IP Phone User Card for instructions on various options for Call Transfer of internal and external calls.
What’s this?
The ETERNITY NE can calculate the cost in amount, for external calls made by extensions. The cost calculation is
done at the time of printing reports.
How it works
The cost of a call depends on:
• Trunk Port (CO, Mobile, SIP) used for making the outgoing call.
• Time and day when the call was made, whether daytime, nighttime, or on a holiday.
• When the call is made from a trunk, the system checks the Call Cost Calculation Pulse Rate Option, 1 to
4 assigned to the trunk, on the basis of the Call Cost Calculation Time Schedule configured for the
outgoing call.
Each Call Cost Calculation Pulse Rate option contains a Pulse Rate Type, which you must assign.
• The system matches the Number dialed by the extension user with the Area Code Table configured in the
system. When the area code matches with an entry in the table, the system obtains the Pulse rate type
configured for the Call Cost Calculation Pulse Rate option assigned to the trunk.
• This Pulse Rate type obtained from the Area Code Table is checked in the Pulse Rate table. to obtain
the corresponding duration and cost to be applied for the call duration. The Pulse Rate Table may be for
'Normal' or 'Holiday', depending on the day of the call.
• The Pulse Rate Type applied (duration and cost) is divided into two parts for each time zone:
• First unit.
• Additional units.
The duration of the call is interpreted in terms of number of units, and the number of units depends on the
pulse rate.
Cost of Call = [Cost of First Unit + (Number of Additional units x Cost for Additional units)] + Service
Charge is applied.
• If the call duration is less than the pulse rate of the first unit then additional unit is zero and total units. Call
units when call answer supervision type is 12/16KHz metering.
ETERNITY uses the Cost of the Call for SMDR and also to deduct it from the Call Budget (Amount), if allotted to
the trunk.
001 26 Local 03 06 09 10
002 09 Mobile 05 03 07 08
Duration (sec) 300 300 300 300 300 300 300 300
02
Cost 01.00 01.00 01.00 01.00 01.00 01.00 01.00 01.00
Durations) 30 30 30 30 30 30 30 30
03
Cost 01.00 01.00 01.00 01.00 01.00 01.00 01.00 01.00
Duration (sec) 45 45 45 45 45 45 45 45
04
Cost 01.00 01.00 01.00 01.00 01.00 01.00 01.00 01.00
Duration (sec) 180 180 180 180 180 180 180 180
05
Cost 03.00 03.00 03.00 03.00 03.00 03.00 03.00 03.00
Durations) : : : : : : : :
:
Cost : : : : : : : :
Duration (sec)
32
Cost
• With this configuration, ETERNITY will calculate the call cost as follows:
• An Outgoing Call is made by an extension user, to the number ‘2630555’ through the trunk, CO-1 at
20:10 hours. The ETERNITY will check the Call Cost Calculation parameters singed on the trunk
and determine the Time Zone as per time of the call.The system will also check the corresponding
Pulse Rate Option configured on the trunk.
In this example, Time Zone for CO-1 at 20:10 Hours would be Time Zone 1, and the Pulse Rate
Option for CO-1 is ‘1’.
• The ETERNITY will match the dialed number ‘2630555’ in the Area Code table. A best match is
found with the entry configured at index 001 in the Area Code Table.
As per the Area Code Table, the Pulse Rate Type ‘03’ is programmed in ‘Pulse Rate Option 1’ for
the matching entry (at Index 001).
(However, if CO-1 would have been assigned Pulse Rate Option ‘2’, the Pulse Rate type ‘06’ would
have been selected as shown in Area Code Table)
• Finally, for Pulse Rate Type ‘03’, ETERNITY will consider the Cost for the First Unit as 1.00 (As. or
$as per applicable currency) for the duration of 30 seconds and for the additional unit also, the cost
will be considered as 1.00 for the duration of 30 seconds. (See Pulse Rate Table). This data will be
used for calculating the total cost of call based on the total duration of the call.
How to configure
To be able to use Call Cost Calculation, you must do the following:
• Define the Unit and Service Charge on the basis of which call cost is to be calculated.
• Assign Call Cost Calculation Pulse Rate Option and the Call Cost Calculation Schedule on the trunks on
which you want to apply this feature.
• Configure the Pulse Rate Types for the Pulse Rate Option you assign to the trunk.
• You may configure different Pulse Rate Types of Normal Days and for Holidays, as required. If configuring
Holiday Pulse Rate, you must also configure the Holiday schedule.
• Open Jives.
Service Charge
By default, no Service Charge is applied on call cost by the system. Service Charge on call cost is generally applied
in Hotels and other organization which charge users for the calls made by them.
• In the Service Charge Type field, select the type of service charge you want to apply from the options:
• Fixed for a call: A fixed amount is added as service charge to every call regardless of the cost of that
call.
If you select this option, you must define the Amount to be added as service charge in the Specify
Service Charge field.
• per Unit: service charge is added to each unit of the call. For example, if a call worth 10 units was
made, the service charge will be applied on each of the 10 units, instead of the one time service
charge as in the case of Fixed service charge.
If you select this option, you must define the amount to be added as service charge on each unit in the
Specify Service Charge field.
• Percentage of call cost: A percentage of the cost of the call is added as a service charge for that call.
If you select this option, you must define the percentage in the Specify Service Charge in % field
which appears.
• To assign Call Cost Calculation Pulse Rate Option and configure the Call Cost Calculation Schedule
on the trunks, go to Basic Settings, and configure these on the different trunk port types. See “CO Trunks”,
“Mobile Trunks”, and “SIP Trunks” under Basic Settings for instructions.
If your service providers do not offer any special rates for holidays, you may skip configuring the Holiday Pulse
Rate table.
Generally service providers offer different call rates for different types of calls, for example: local, national,
international. You can configure different Pulse Rate Types for different types of calls. Thus, each Pulse
Rate Type can have different rates for the First and the Additional unit.
The Pulse Rates offered by service providers may vary according to the time of the day. In such cases, you
will need to configure the Call Cost Calculation Schedule for the trunk, by dividing the day into Time
Zones, from 1 to 4, as required, to match the time of the pulse rates offered by your service provider.
If you have configured Time Zones for the Call Cost Schedule on a trunk, you may define the different
Pulse Rate Types for each Time Zone.
• In the Area Code column, enter the number strings (prefix) of the Area Codes, country codes, local
numbers. You can configure as many as 999 area codes in the table.
• Do not configure the Ignore Digit Count. This parameter is relevant only for Service Provider Based Least
Cost Routing.
• Different service providers offer different pulse rates for the same type of calls. To take care of this,
ETERNITY allows you to assign different Pulse Rate Options for each area code.
• For each area code, configure the Pulse Rate to be followed for the desired Pulse Rate Options in the
Pulse Rate Type for Pulse Rate (Option 1 to 4) column.
What’s this?
If your extension is a DKP or an Extended IP Phone, you can use the Call Cost Display feature to view the cost of
the last 10 external calls made from your extension.
The system will display the dialed numbers and the call cost for each number that it has calculated on the LCD of
your phone.
How to configure
For this feature to work, it must be enabled on the extension by the System Administrator (SA). You may enable
Call Cost Display on an extension from Jeeves or from a telephone.
• Enter SA mode.
• Exit SA mode.
How to use
OR
• Dial 1075.
• Scroll with the up/down navigation keys to view the cost of the last 10 calls.
• The display shows the last 10 dialed numbers and their corresponding call cost.
• Go Idle
What's this?
Call Duration Control (CDC) allows a maximum time limit to be set on internal and external (both incoming and
outgoing) telephone calls. When the maximum call duration is reached, the calls are disconnected, after a warning
tone indicating to the user that the calls in progress will be disconnected.
By limiting the duration of the conversations, CDC helps increase availability of trunks for making outgoing calls
and for receiving incoming calls, which is important in high call traffic situations. Besides increasing trunk
availability, CDC curbs unrelated and unproductive conversations.
How it works
• A is an extension user. B is an external number.
External-Outgoing Calls
• A dials B's number.
• It checks whether the flag, Apply CDC to Outgoing Calls made from Trunks, is enabled. It matches B's
number with the entries on the Apply CDC for calls matching with numbers list and the Do Not Apply
CDC for calls matching with numbers list in the CDC profile. Three results are possible:
• The flag is enabled and a match is found for the number in the Apply CDC for calls matching with
numbers List. So, CDC is applied on the call.
• The flag is enabled and a match is found in the Do Not Apply CDC for calls matching with numbers
list in the CDC profile. So, CDC is not applied on the call.
• The flag is enabled and a match is found in both these lists. The system gives precedence to the Do
Not Apply CDC for calls matching with numbers list. So, CDC is not applied to the call.
• When CDC is applied on the call (see point a above), the CDC Timer starts as soon as B has answered
the call. This timer is set to 160 seconds as default, but can be configured for the desired time limit.
• At the end of the CDC Timer, the CDC Goodbye Timer starts. This timer provides a grace period of 20
seconds for the user to finish the call. This Timer is fixed.
• At the end of the Goodbye Timer, the call is disconnected, if the Disconnect CDC after Timer flag is
enabled.
• Instead, the CDC Warn Timer will be loaded again. The user can know how long s/he has been talking.
External-Incoming Calls
CDC works similarly for incoming calls.
• B calls A.
• The system checks whether the flag, Apply CDC for incoming calls received from trunks, is enabled
and matches B's number with the entries on the Apply CDC for calls matching with numbers list in the CDC
Profile assigned to A. If the flag is enabled and a match is found for the number, CDC is applied on the call.
Internal Calls
A and B are extension users.
• A calls B.
• The system checks whether the flag, Apply CDC to Internal Calls, is enabled in the CDC profile assigned
to B.
Warning Beep
Feature Interactions:
• Call Transfer: In case of transferred call, the CDC timer gets reset and starts again afresh on the
transferred extension.
• Emergency Number Dialing: Emergency calls are not affected by this feature, i.e. CDC will not be
applied on the dialing of Emergency Numbers.
• Configure Call Duration Control Profiles. You may configure up to 4 different Profiles.
• Enable Call Duration Control on the extension, and assign a CDC Profile to the extensions on which you
want to apply this feature.
• First, decide the types of calls - Outgoing, Incoming and Internal - on which CDC is to be enabled.
• Make a list of numbers on which CDC is to be applied, and a list of numbers on which CDC is not to be
applied.
• Make a list of extensions on which CDC is to be applied and the CDC Profile number you want to assign to
these extensions.
• Open Jeeves.
• Enable Apply CDC to Internal Calls, if CDC is to be applied on internal calls. Default: Disabled.
• Enable Apply CDC for Incoming Calls received from trunks, if CDC is to be applied on incoming
external calls. Default: Disabled.
• Enable Apply CDC for Outgoing Calls, if CDC is to be applied to outgoing external calls. Default:
Disabled.
• If required, change the CDC Timer to the desired duration. The range of the timer is 0001 to 9999
seconds. Default: 160 seconds.
• To have calls disconnected on the expiry of the CDC Timer, enable Disconnect Call after CDC Timer
check box. Default: Disabled.
• In the Apply CDC for calls matching with numbers column, enter the numbers on which CDC should be
applied. You may refer to the list you prepared.
• To configure another CDC Profile, click the Profile number tab, and follow the same steps described above
to configure this profile.
• Now assign the CDC Profile to the extensions. For instructions see “SLT Extensions”, “DKP Extensions”,
“SIP Extensions”, under Basic Settings.
What's this?
By invoking Call Duration Display, extension users can view the duration of the current call instantly.
Needless to say, only DKP and Extended IP Phone users can view Call Duration Display. The system displays the
duration of the current call on the LCD display of the phone.
How it works
• The DKP/Extended IP Phone user goes OFF-Hook
• The dialed external number with duration (5-digits in the format of MM:SS) is displayed on the LCD of the
phone, when the call is answered.
6 1 6 A M I T PAT E L 02:52
F ri 22 JAN 12:19
What's this?
During a typical workday, it is common for people in an organization to move from one place to another. For
instance, a manager might go on the production floor or remain in the conference room for a few hours; a field
engineer may spend half of the day on site. So, they need to be able to attend their calls even when they are not
present at their usual workplace. The 'Call Forward' feature of ETERNITY ensures this.
Using this feature, calls landing on an extension can be forwarded to another extension, external number, Voice
Mail Group, or Department Group. This way, extension users can ensure that callers can reach them and that they
do not miss calls when they are not present at their extension.
The Call Forward feature of ETERNITY offers the following forwarding options:
• Unconditionally - calls are forwarded to the destination phone number automatically without waiting for a
response from the called party's phone.
• If Busy - calls are forwarded to the destination phone number only when the called party's phone is busy.
• If No Reply - calls are forwarded to the destination phone number only when the called party does not
answer the phone. Each extension can set a different time after which the call should be forwarded, in
case of no reply. The default time is 30 seconds for all extensions and can be changed by configuring the
Call Forward No-Reply Timer.
• If Busy or No Reply - calls are forwarded to the destination phone number when the called party's phone
is either busy or does not reply.
• Dual Ring - when calls are forwarded to another phone number. Both phones, i.e. the source phone
(whose calls are forwarded) as well as the destination phone (on which call is forwarded) will ring and the
user can answer from either extension.
Dual Ring is useful to users who may find themselves shuttling between two places frequently. As it is
cumbersome to forward the calls from one extension to another and cancel it repeatedly, users can set
Dual Ring, so they can attend to their calls at either place they are present.
How it works
A has set Call Forward to extension B unconditionally.
• The system forwards all calls for A to B, without checking for Busy Tone and without waiting for the Call
Forward No-Reply Timer to expire.
• The system waits for the Call Forward No-Reply Timer to expire and forwards all external incoming calls to
the external number.
• The system forwards the call for A to B on detecting Busy tone from A.
B belongs to a Department Group and has set Call Forward-If Busy to C within the Department Group.
• If the system detects Busy signal on B, it forwards the call for B to C in the Department Group.
• However, if the caller has called the Department Group instead of calling B directly, the call will land in the
sequence on all Department group extensions. When it is B's turn, the call will not be forwarded to C, B will
ring instead.
C belongs to a Department Group and has set Call Forward-No Reply to D within the Department Group.
• The system waits for the Call Forward No-Reply Timer to expire, and forwards the call for C to D in the
Department Group.
• Whenever there is a call for D, if the system does not detect a busy tone from D, it waits for the Call
Forward No-Reply timer to expire.
• When there is a call for E, the system rings on both E and the destination F.
Feature Interaction:
• Do Not Disturb (DND): When DND and Call Forward are set on an extension, DND is given priority.
• You can select the types of calls, i.e. internal calls only, or trunk calls, or both, to be forwarded to
external numbers. You can program the system to forward internal calls only, or trunk calls only or both
trunk calls and internal calls, to the external number. For this, the parameter when Call is forwarded
to an external number must be configured on the extensions.
• The system supports only single-point Call Forward, which means, if the destination extension is also
forwarded, the call will not follow the forwarding path. For example: Calls for extension A are set to be
forwarded to extension B. Call Forward is also set on extension B with C as the destination number.
Calls for A will land on B only and calls for B will land on C only.
• Only one Call Forward Type can be set from an extension. Every new Call Forward Type set overrides
the previous one.
• When the calls are forwarded the extension user gets the feature tone on lifting the handset to indicate
that Call Forward is set on the extension.
• Enable Call Forward in the “Class of Service (COS)” of the Extensions. By default, Call Forward is enabled
in the Class of Service of all extensions of ETERNITY for the Day and Night.
• If required, change the duration of the Call Forward No-Reply Timer on the extension. By default the timer
is set to 30 seconds.
• Select the type of call—internal, trunk, all calls— to be forwarded when Call is forwarded to an external
number on the extension. By default, Forward Only Trunk Calls is selected.
For instructions on configuring Call Forward related parameters different extension types, see “SLT Extensions”,
“DKP Extensions”, “SIP Extensions” under Basic Settings.
How to use
Call Forward can be set/canceled by extension users who are allowed this feature. It can be set/canceled by an
extension user for another extension (refer “Call Forward-Remote” to know more).
OR
• Dial 13.
• Scroll to select the desired Call Forward Type.
• Press 'Enter' key.
• Enter destination Phone Number/Voice Mail/ Department Group Number.
• You get a confirmatory text message and confirmation tone.
• Go Idle.
• If the call is to be forwarded to an external number, dial Trunk Access Code, then the external phone
number and terminate the command with #*.
• For users in USA, TAC for dialing external numbers are: 9, 5, 81, 82, 83, 84.
• If call is to be forwarded on voice mail, dial the Access Code for the Voice Mail. The default Access
Code is 3931. If this code has been changed and use the new code to dial the VMS.
OR
• Dial 13.
• Select 'Cancel'.
• You get a confirmatory text message and confirmation tone.
• Replace Handset on the cradle or you get dial tone after 3 seconds.
What's this?
Extension users may want their calls to be automatically forwarded to a desired destination number during working
hours or non-working hours. To cite an example, a Support Technician spends working hours on the field and
wants all incoming calls on his extension in the office to be forwarded to his cell phone during working hours. During
non-working hours, he wants call calls to be forwarded to his voice mail.
Remembering to set and cancel Call Forward and changing the destination number for each Time Zone, i.e.
working hours and non-working hours every day proves to be cumbersome for such extension users.
In addition to “Call Forward”, ETERNITY supports Call Forward - Scheduled, which allows extension users to set
call forward for the desired Time Zones at one time, and the system automatically forwards the calls to the
destination defined for each Time Zone.
How it works
Call Forward-Scheduled supports all the forwarding options as Call Forward: Unconditionally, If Busy, If No Reply, If
Busy or No Reply, Dual Ring.
Any of these options can be set for the Time Zones: working hours and non-working hours.
The destination for Call Forward-Scheduled can be an internal (extension) number or an external number.
Both 'Call Forward' and Call Forward-Scheduled can be set on the same extension. In this case, priority is given to
'Call Forward' over Call Forward-Scheduled.
The logic for forwarding calls to the destination number remains the same as described in the topic “Call Forward”,
illustrated in the following example.
• When there is a call on extension A, the system first checks if there is any 'Call Forward' type (i.e.,
Unconditional, Busy, No Reply, Busy/No Reply, Call Follow Me) set on extension A.
• If 'Call Forward' is set on extension A, the system will follow the logic described in 'How it works' under the
topic “Call Forward”.
• If no 'Call Forward' is set on extension A, the system will check if Call Forward-Scheduled is set on A.
• Since Call Forward-Scheduled is set on extension A, the system will compare the Time Zone for which the
Call Forward is scheduled with the current Time Zone of extension A.
• If the current Time Zone of extension A is the same as the Time Zone set for Call Forward Scheduled, i.e.
non-working hours, the call will be forwarded to extension B as per the call forward type set.
• As the Call Forward Type set by A is Unconditional, the system will forward the call to B, without checking
for the Busy Tone and without waiting for the Call Forward No-Reply Timer to expire.
• Call Forward - Scheduled can be set simultaneously for more than one Time Zone from the same
extension. For example, extension A can set Call Forward-Scheduled for working hours, then again set
Call Forward-Scheduled for non-working hours.
• A different Call Forward Type can be set for a different Time Zone. For example, extension A can set
Call Forward -Unconditional for non-working hours, and Call Forward -Busy for working hours. Also, a
different destination number can be set for forwarding calls in each Time Zone. For example, extension
A can set Call Forward-Unconditional for non-working hours to a mobile number and set extension B as
destination number for working hours.
• When more than one Call Forward type is set on the same extension for the same Time Zone, the
latest Call Forward type set for the Time Zone will override the previous Call Forward type set for that
Time Zone. For example, extension A sets Call Forward -Busy for working hours, then sets Call
Forward Busy or No Reply for working hours, the latter will override the former. The system will
consider the latest, i.e. Busy or No Reply as the Call Forward type for forwarding calls during working
hours.
• Call Forward-Scheduled can be cancelled individually for a desired Time Zone or all at once for all
Time Zones.
• Call Forward-Scheduled can be set by extension users as well as for extension users from the System
Administrator mode.
• It is also possible to select the types of calls, i.e. internal calls only, or trunk calls, or both, to be
forwarded to external numbers. You can program the system to forward internal calls only, or trunk
calls only or both trunk calls and internal calls to the external number. For this, the Call Forward option
If call is forwarded to an external number must be configured on the extensions that want to use Call
Forward-Scheduled.
How to configure
The configuration of this feature involves the same parameters as in “Call Forward”.
• Call Forward must be enabled in the Class of Service (Cost) of the extensions to which this feature is to be
allowed.
• If Call Forward No-Reply is to be set, if required, the Call Forward No-Reply Timer may be configured.
• You may select the types of calls—internal, trunks, all calls—to be forwarded to the external number by
configuring the option Call Forward option If call is forwarded to an external number on the extensions.
• Extensions that are to be allowed to set Call Forward-Scheduled for other extensions must be allowed
either the feature 'SA Mode’ or ‘SA Extension' in their Class of Service. Refer the topic “Call Forward-
Remote”.
• The destination number for forwarding calls can be a maximum of 24 digits. Terminate the command
with #* if destination number has fewer than 24 digits.
• If the destination number is an external number, enter the Trunk Access Code followed by the
destination number.
• The Extension numbers appear on the tabs on the right pane, starting with SLT extensions.
• To select a particular extension number or type, click on the tab.To select a particular extension number or
type, click on the tab.
• Select the type of Call Forward you want to set for the extension for the Day Time, i.e. working hours:
• To forward calls to voice mail, select the radio button Forward Calls to Voice Mail and the type of call
forward from the drop down list. Default: Unconditionally.
• To forward calls of this extension to another extension, select Forward Calls to Phone radio button
and the type of call forward for this option from the drop down list. Default: Unconditionally.
Enter the extension number to which calls must be forwarded in the empty field provided for this option.
• To forward calls of this extension to an external number, select Forward Calls to External Number
radio button, and select the type of call forward for this option from the drop down list. Default:
Unconditionally.
Enter the external number to which calls must be forwarded in the empty field provided for this option.
The label of the Call Forward button changes color and the message “Call Forward is set” appears.
• Click the Dual Ring button to set Call-Forward Remote with Dual Ring.
The label of the Dual Ring button changes color and the message “Dual ring is On” appears.
• To set call forward for non-working hours, click Call Forward - Night Time to expand.
• Follow the same instructions as above for setting Call Forward Scheduled for the Day Time.
• To set Call Forward - Day Time on another extension, click the extension number tab, and follow the same
instructions as above.
OR
• Dial 1072-223.
• Enter extension number (from which calls are to be forwarded)
• Scroll to the desired Time Zone.
• Press Enter key to select Time Zone.
• Scroll to the desired Call Forward type for the selected Time Zone.
• Press Enter key to select Call Forward type.
• Enter Destination Number on prompt.
• You get confirmation tone and message showing extension to which Call Forward is set.
OR
• Dial 1072-223
• Enter extension number (for which it is to be canceled)
• Scroll to the desired Time Zone.
• Press Enter key to select Time Zone.
• Scroll to select Cancel.
• Press Enter key.
• You get confirmation tone and message.
OR
• Dial 1072-223
• Enter extension number (for which it is to be canceled)
• Scroll to 'Cancel Call Forward'.
• Press Enter key.
• You get confirmation tone and message.
• Lift handset.
• Enter SA Mode.
• Dial 1072-223-Extension number-1-1-Destination Number for CF-Scheduled -Unconditional.
• Dial 1072-223-Extension number-1-2-Destination Number for CF-Scheduled -Busy.
• Dial 1072-223-Extension number-1-3-Destination Number for CF-Scheduled -No Reply.
• Lift handset.
• Dial 1072-223-Extension number-3-1-Destination Number for CF-Scheduled -Unconditional.
• Dial 1072-223-Extension number-3-2-Destination Number for CF-Scheduled -Busy.
• Dial 1072-223-Extension number-3-3-Destination Number for CF-Scheduled -No Reply.
• Dial 1072-223-Extension number-3-4-Destination Number for CF-Scheduled -Busy/No Reply.
• Dial 1072-223-Extension number-3-5-1 for CF-Scheduled-Dual Ring.
• Dial 1072-223-Extension number-3-5-0 to cancel CF-Scheduled-Dual Ring.
• Dial 1072-223-Extension number-3-0 to cancel CF-Scheduled.
• Replace handset.
OR
• Dial 1175.
• Scroll to the desired Time Zone.
• Press Enter key to select Time Zone.
• Scroll to the desired Call Forward type for the selected Time Zone.
• Press Enter key to select Call Forward type.
• Enter Destination Number on prompt.
• You get confirmation tone and message showing extension to which Call Forward is set.
OR
• Dial 1175.
• Scroll to the desired Time Zone.
• Press Enter key to select Time Zone.
• Scroll to select Cancel.
OR
• Dial 1175.
• Scroll to 'Cancel Call Forward'.
• Press Enter key.
• You get confirmation tone and message.
• Lift handset.
• Dial 1175-1-1-Destination Number for CF-Scheduled-Unconditional.
• Dial 1175-1-2-Destination Number for CF-Scheduled -Busy.
• Dial 1175-1-3-Destination Number for CF-Scheduled -No Reply.
• Dial 1175-1-4-Destination Number for CF-Scheduled-Busy/No Reply.
• Dial 1175-1-5-1 for CF-Scheduled -Dual Ring.
• Dial 1175-1-5-0 to cancel CF-Scheduled -Dual Ring.
• Dial 1175-1-0 to cancel CF-Scheduled for the Day.
• Replace handset.
• Lift handset.
• Dial 1175-3-1-Destination Number for CF-Scheduled -Unconditional.
• Dial 1175-3-2-Destination Number for CF-Scheduled -Busy.
• Dial 1175-3-3-Destination Number for CF-Scheduled -No Reply.
• Dial 1175-3-4-Destination Number for CF-Scheduled -Busy/No Reply.
• Dial 1175-3-5-1 for CF-Scheduled -Dual Ring.
• Dial 1175-3-5-0 to cancel CF-Scheduled -Dual Ring.
• Dial 1175-3-0 to cancel CF-Scheduled for the Night.
• Replace handset.
What's this?
An extension user can set Call Forward for another ('remote') extension from his/her own extension. Thus, Call
Forward set for an extension from another extension is called 'Call Forward-Remote'.
This feature can be used by the Operator or the Receptionist to forward the calls for the Manager(s) and other
extension users to the destinations where they will be available.
• Call Forward-Remote is possible only from the System Administration (SA) mode.
How it works
This feature works in the same way as Call Forward. The only difference is that it is set by one extension user for
another extension.
For example:
• A needs to forward calls for B's extension to another extension 'C' or an external number or a Voice Mail
Group or a Department Number.
• A dials the Call Forward-Remote feature code followed by B's extension number, the destination number
where the calls for B should land.
• The system routes all incoming calls for B to the destination number.
How to configure
As Call Forward-Remote can be invoked only from the SA mode, either the feature 'SA Mode' or 'SA Extension'
must be enabled in the “Class of Service (COS)” of extensions that are to be allowed this feature.
The feature 'SA Mode' requires a password to be dialed. Users must be provided a password to use this
feature from their extensions. The feature 'SA Extension' allows an extension user to enter into SA mode,
without a password.
By default, all extensions are allowed SA Mode in their Class of Service. So, all extensions can access Call
Forward Remote, provided they have the SA password.
You may decide which extensions should be allowed Call Forward-Remote feature. Generally, only few extensions
are allowed this feature.
For instructions on configuring SA Mode and SA Extension in the Class of Service of different extension types, see
“SLT Extensions”, “DKP Extensions”, “SIP Extensions” under Basic Settings.
• The Extension numbers appear on the tabs on the right pane, starting with SLT extensions.
• To select a particular extension number or type, click on the tab. To select a particular extension number
or type, click on the tab.
• Select the type of Call Forward you want to set for the extension:
• To forward calls to voice mail, select the radio button Forward Calls to Voice Mail and the type of call
forward from the drop down list. Default: Unconditionally.
• To forward calls of this extension to another extension, select Forward Calls to Phone radio button
and the type of call forward for this option from the drop down list. Default: Unconditionally.
Enter the extension number to which calls must be forwarded in the empty field provided for this option.
• To forward calls of this extension to an external number, select Forward Calls to External Number
radio button, and select the type of call forward for this option from the drop down list. Default:
Unconditionally.
Enter the external number to which calls must be forwarded in the empty field provided for this option.
The label of the Call Forward button changes color and the message “Call Forward is set” appears.
• Click the Dual Ring button to set Call-Forward Remote with Dual Ring.
The label of the Dual Ring button changes color and the message “Dual ring is On” appears.
• To set Call Forward on another extension, click the extension number tab, and follow the same instructions
as above.
• You can also forward the calls of all extensions at one go to the same destination. To do this,
• Select the Forward Calls of all Extensions radio button and select the Call Forward type from the drop
down list.
OR
• Dial 1072-006.
• Enter the Destination Phone Number.
• Scroll to select the desired Call Forward Type:
• All Calls.
• If Busy.
• If No Reply.
• If Busy or No Reply.
• Dual Ring.
• Press 'Enter' key.
• Enter Destination Phone Number18/Voice Mail19/Department Group.
• You get a confirmation tone and a text message for the Call Forward type set.
• Go Idle.
OR
• Dial 1072-006.
• Enter Extension Number.
• Scroll to select 'Cancel'.
• Press 'Enter' key.
• You get a confirmation tone and text message for Call Forward canceled.
• Go Idle.
18. If call is to be forwarded to an extension of the ETERNITY, dial the extension number. If call is to be forwarded on an external num-
ber, dial Trunk Access Code, then dial the external phone number and terminate the command with #*.
For users world wide, Trunk Access Code (TAC) for dialing external numbers are: 0, 5, 61, 62, 63, 64. For users in USA, TAC for
dialing external numbers are: 9, 5, 81, 82, 83, 84.
19. If call is to be forwarded on voice mail, dial the Access Code for the Voice Mail group. The default Access Code is 393121.
• Lift handset.
• Dial 1072-006.
• Enter Extension Number.
• Dial 1 for All Calls
• Dial 2 for If Busy
• Dial 3 for If No Reply
• Dial 4 for If Busy or No Reply
• Dial 5 for Dual Ring
• Dial destination Phone Number/Voice Mail.
• You get confirmation tone.
• Replace handset.
What's this?
Call Hold enables you to put an on-going conversation (with an internal or external number) on hold, and call
another person or receive a call from another person. You can retrieve the call you put on hold, after the
conversation with the other party has ended or in the middle of the conversation with the other party.
Call Hold is a feature of the DKP and the Extended IP Phone. ETERNITY NE enables two types of Call Hold on the
DKP: Exclusive Hold and Global Hold.
Exclusive Hold
The call placed on Exclusive hold can be retrieved only from the DKP/Extended IP Phone which put it on hold. The
call remains connected to the DKP/Extended IP Phone which placed it on hold.
When a call is put on Exclusive hold, the ETERNITY NE starts the Call Park Timer and Call Park Release Timer.
The call remains on hold for the duration of the Call Park Timer (programmable; default: 45 seconds). If this call is
not retrieved before the Call Park Timer expires, the call is parked in the Personal Orbit for the duration of the Call
Park Release Timer (programmable; default: 3 minutes). If the DKP/Extended IP Phone becomes idle within this
Timer, the call is returned to the DKP/Extended IP Phone and the phone gets Ring Back Tone.
If the DKP/Extended IP Phone is busy, the system waits for the duration of the Call Park Release Timer for the
DKP/Extended IP Phone to go idle or retrieve the call. If the DKP/Extended IP Phone goes idle within this Timer,
the call is returned to the DKP/Extended IP Phone and the phone gets Ring Back Tone.
The call is disconnected if the DKP/Extended IP PHone is still not free or does not retrieve the call before the Call
Park Release Timer expires.
• Pressing the Hold key again (when the DKP/Extended IP Phone is idle).
• Pressing the Key of the Call Appearance of the call put on hold (before it gets parked).
• Answering the call, when it returns at the end of the Call Park Timer.
Global Hold
The call placed on Global hold can be picked up from any DKP/Extended IP PHone extension of ETERNITY. The
call remains connected in the system. The call remains on hold for the duration of the Call Hold Retrieval Timer
(programmable; default: 60 seconds). If this call is not retrieved before the expiry of the Timer, the call is returned to
the DKP/Extended IP Phone which put it on hold.
To be able to place calls on Global Hold, you must enable 'Global Hold' in the System Parameters of ETERNITY.
The DKP/Extended IP Phone (which picks up the call) must have a DSS Key to access the Trunk or the Extension
which is put on hold.
• Pressing the DSS key assigned to the extension put on Global Hold.
The call on Global Hold must be picked up before the Call Hold Retrieval Timer expires.
ETERNITY NE provides the flexibility to use Exclusive Hold and Global Hold at the same time. You can put
calls on Exclusive Hold even when Global Hold is enabled in the system.
How to configure
For this feature to work,
• Call Hold must be enabled in the “Class of Service (COS)”of the DKPs and Extended IP Phones you want
to allow this feature.
Call Hold is a part of the ‘Basic Features’ allowed to all extensions by default in their class of Service. So,
all extensions of ETERNITY NE can use this feature.
• Global Hold must be enabled in the “System Parameters” if you want Calls on Hold to be picked up by any
DKP/Extended IP Phone extension.
• DSS Keys must be configured for Trunks and Extensions on the DKPs and Extended IP Phones, which
are allowed to retrieve calls on Global Hold. See “Phone Key Settings” under “DKP Extensions” for
instructions. Also see, “Matrix Extended IP Phone Key Settings” under “SIP Extensions”
How to use
Exclusive Hold
To put a call on Exclusive Hold, when Global Hold is disabled:
• press DSS Key of the Trunk/extension you put on hold from your phone.
Global Hold
To put a call on Global Hold:
What's this?
ETERNITY stores the details of 20 each, of the following types of calls:
• Missed calls: incoming calls that were not answered by extension users.
• Answered calls: incoming calls answered by extension users.
• Dialed calls: calls made by extension users.
The call history of each of the above types of calls is stored by Name, Number, and Date-Time of the Call.
If there is no name in the CLI of the above types of calls, the system stores and displays the Number and the Date-
Time. In case there is no number in the CLI, the system will display the Port number on/from which the call was
received/made.
The Call Logs contain details of both internal as well as external calls made or received by the extension users.
• view call history: you can see the calls you missed, answered or dialed.
• make calls: you can call any number that you have missed, answered, or dialed.
• edit the numbers: you can change or modify the number in the call log. This is useful when the CLI
received and stored in the call log is not in the same format that is to be used to make calls.
• save the numbers: you can store the external numbers in your call logs in the “Personal Directory" and
use them for “Personal Abbreviated Dialing”.
The maximum number of calls that can be stored under each Call Log type is 20. The logs will be cleared
automatically using the First-In, First-Out method, i.e. the latest call detail will replace the record of the oldest call
detail.
Given the limited Call Log capacity, the system allows you to choose if you want internal calls to be displayed or not
in the Missed, Answered and Dialed Call Logs, and accordingly stores the internal calls in the logs.
The system stores each Missed, Answered and Dialed call individually even if the same number is received
multiple times.
How to configure
This feature does not require any specific configuration, except:
• Selecting whether internal calls should be logged in the Missed, Answered and Dialed Call Logs. This can
be done on the '“System Parameters”' page under Advanced Settings.
• Programming of a DSS key for the Call Logs feature.See “Phone Key Settings” under “DKP Extensions”
for instructions. Also see, “Matrix Extended IP Phone Key Settings” under “SIP Extensions”.
• You may enable any or all of the following flags by clicking on the respective check box:
• Log Internal Calls in Missed Calls
• Log Internal Calls in Answered Calls
• Log Internal Calls in Dialed Calls
How to use
The Call Logs feature allows you to view calls and edit numbers, make calls to any number logged, and store
numbers.
• If there is no name in the CLI, the Call Log will only display the number.
• If you press the 'Enter' key, the system will dial out the number you just viewed.
OR
• Press the DSS Key assigned the Call Logs feature, when it glows.
• The phone will display the call log details of the last missed call by: <Name> <Date> <HH:MM> (only if
name is received).
• Press Enter key.
• The phone will display the Number: <XXXXXXXXXXX>
• You may exit the Phone Menu by going OFF-Hook or pressing the Cancel key20.
• You may also edit or store the number.
• You may scroll with the < Back navigation key to view the other call logs.
• The LED of the Call Logs DSS key will be turned off once you have viewed the missed call.
20. This key is available on EON48. This key must be programmed on EON42.
• When you store the number in the Personal Directory, the system will automatically assign Trunk
Access Code "0".
• If all 25 Location Index Numbers of the Personal Directory are already programmed, the message
"Memory Full" will appear on your phone's display and you will get an Error Tone. Refer the topic
“Abbreviated Dialing” to know more.
What is this?
Call Park allows you to place a call on hold, so it can be retrieved from the same or another extension of the
system.
A call is 'parked' when the extension user temporarily places the call into a location in the system called 'Orbit'. The
user can attend to other calls. The parked call can be retrieved on completion of the current call by dialing the Orbit
number.
Call Parking is useful in offices housed in different parts of a building or multi-storied offices. It is useful in situations
like:
• the person who picked up the call is not the desired called party or the desired party is at an unknown
location. The person who picked up the call can then either go to find the desired called party or call other
numbers to find him/her. When found, the desired called party can pick up the call from the same or any
extension by dialing the Orbit number.
• the person who picked up the call may have to go to another part of the office to look up a file or consult a
colleague. The person can park the call and continue the conversation from the other part of the office.
• Call Park-General Orbit: The extension user can park calls in any of the 8 'general' Orbits, which are like
fictional extensions located in the system. The calls parked in the General Orbit can be picked up from any
extension by dialing the General Orbit Number. At a time, only one call can be parked in each General
Orbit.
• Call Park-Personal Orbit: Each telephone instrument (EON or SLT) connected as extension has one
Personal Orbit. Calls parked in personal orbit can be picked up only from where the call is parked. So, no
other person can pick up this call. Multiple calls can be parked in the Personal Orbit at a time.
Extension users can park the call either in the General Orbit or the Personal Orbit by dialing an Orbit Number from
1 to 9, where:
After parking a call, the extension user can continue to make and answer other calls and use other system features.
How it works
A and B are extension users. C, D and E are callers.
However,
• If neither A nor B retrieves the parked call within the Call Park Timer, the system will hunt for the extension
that parked the call (A) on the expiry of the Call Park Timer.
• Meanwhile, if A is busy, the system again keeps the call parked in orbit number 2 for the period of the Call
Park Timer. This process continues for the duration of the Call Park Release Timer, which is set to 3
minutes by default.
• If A is free, the system will ring on A's phone. A gets connected to C again.
• If A does not retrieve the parked call till the end of the Call Park Release Timer, C gets disconnected.
When there are multiple calls to be retrieved from the Personal Orbit, they are retrieved one by one, without
following any particular sequence like FIFO or LIFO.
To be able to use 'Call Park', this feature must be enabled in the COS of the requesting extension.
However, for retrieving parked calls, the system does not check COS. So any extension can retrieve
parked calls.
How to configure
To be able to use this feature, you must do the following configuration:
• Enable Call Park in the “Class of Service (COS)” of the extensions which you want to allow this feature.
For instructions see “SLT Extensions”, “DKP Extensions”, “SIP Extensions” under Basic Settings.
• If required, change the value of the Call Park Timer and the Call Park Release Timer to the desired
duration. See “System Timers and Counts” for instructions.
How to use
• Press DSS Key assigned to 'Call Park' again. The LED of the key is turned off.
OR
• Dial Flash-116
• Enter Orbit Number where you parked the call (1-9)
(Personal Orbit:1; General: 2-9).
• You are in speech with the extension/external caller.
What's this?
Call Pickup allows extension users to answer calls ringing on other extensions from their own extension; without
physically going to the ringing extensions.
Extension users can 'pick-up' both internal and trunk calls ringing on other extensions.
As extension users can answer calls of their colleagues or co-workers without physically going to their extensions,
this feature ensures that all incoming calls are answered.
• Call Pick Up-Group - extensions are assigned to Pick-Up Groups. Any extension in a Pick-Up Group can
answer calls ringing on other extensions within the same group only.
• Call Pickup Selective - calls ringing on any extension of the system can be answered.
How it works
• For example, extensions 201, 201, 203, and 303, 304, 305 are assigned to Pick-Up Group number 03.
• When an extension in this group rings, any extension in the group 03 can pick up the call by dialing ‘4’ the
(default) feature access code for "Call Pickup Group".
• Whenever an extension in the system rings, the call can be picked up by any extension of the system by
dialing the feature access code and the number of the ringing extension.
When more than one extension in a Pick-Up Group are ringing, you can choose which one to answer first,
using Call Pickup Selective.
• Call States: Call Pickup will fail if the ringing extension goes into idle state just when you are dialing the
pick-up access code.
• Auto Call Back: Call Pickup will fail if the call ringing on the extension is an Auto Call Back request.
• Alarms: Call Pickup will fail if the call ringing on the extension is an Alarm Call.
How to configure
On a sheet of paper, list the extensions that are to be grouped into a Call Pickup Group. Make as many Call Pickup
Groups as required. Assign each group a number.
Call Pickup
SLT Extensions DKP Extensions SIP Extensions
Group Number
08
The numbering of Call Pickup Groups must start from 01 and end at 08.
• Configure “Call Pickup Group”. You may refer to the sheet of paper you prepared.
• For each SLT/DKP/SIP extension assigned to a CPU Group, make sure Call Pickup is enabled in the
“Class of Service (COS)” of the extensions for the Day and Night. Default: Enabled. See “Basic Settings”
for instructions on configuring COS of different Extension types: SLT, DKP, SIP.
How to use
OR
• Dial 4.
To pick up any one of several ringing extensions ringing or the extension that is not in your group:
OR
• Dial 12.
• Dial number of the Extension you want to pick up.
• Talk.
• Go idle.
To pick up any one of several ringing extensions ringing or the extension that is not in your group:
What's this?
Call Progress Tones (CPT) are audible tones sent from switching systems such as PSTN or PBX to calling parties
to show the status of phone calls, like dial tone, error tone, ringing error in number dialed, ringing called party, busy
line, etc.
Each CPT has a distinctive tone frequency and cadence assigned to it, for which some standards have been
established by the International Telecommunication Union (ITU).
On the basis of specific frequency, modulating frequency and cadence, the CPTs generated by ETERNITY NE are
categorized as:
Dial Tone 1 Played on lifting the Toooooooooooo Played for 7 seconds. Dial Tone Timer
handset. After which Error Tone
starts
Dial Tone 2 Played on lifting the Toooooooooooo Played for 7 seconds. Dial Tone Timer
handset, when 'Store After which Error Tone
and Forward Dialinga' is starts
done.
Ring Back Tone Played when the Turroo... Turrroo Played for 45 seconds Ring Back Tone
internal number you Timer
have dialed is free.
Busy Tone High pitch beeps with Tooooooo......... Played for 7 seconds. Busy Tone Timer
(Engaged Tone) equal ON and OFF Toooooooo
periods, played when
the dialed extension is
busy. Busy tone
continues for 7
seconds. This Busy
Tone Timer is
programmable.
Error Tone Fast beeps, played on a Too…Too…Too Played for 30 seconds Error Tone Timer
(Congestion/ wrong operation being …Too
Refusal Tone as performed or a feature
per ITU invoked without access.
Internal Call Short beep followed by Beep.……….… Played for duration of Interrupt Request
Waiting Tone longer OFF duration Beep the Interrupt Request Timer, Barge-In
(Intrusion Tone repeated every second; Timer or the Barge-In Timer
as per ITU) played to the busy Timer.
extension when another
extension attempts
Interrupt Request/
Barge-In
External Call Two ticks followed by a Beep...Beep… Played for the duration Transfer-On Busy
Waiting Tone longer OFF time of ……......Beep... of the Transfer-On Busy Timer.
(Call Waiting approx. 3 seconds; Beep Timer.
Tone as per ITU) played to a busy
extension when there is
a new incoming Trunk
call.
Confirmation Continuous, fast beeps, Beep... Beep... Played for 7 seconds. Confirmation Tone
Tone played to confirm Beep Timer
(Acceptance successful use of
Tone as per ITU) features.
Feature Tone Short beep followed by Beep................. Played until user goes
a longer off duration Beep ON-Hook or dials a
repeated every second; feature code.
played when dialing
feature access codes
Programming Short beep followed by Beep................. Played until user goes Programming Tone
Tone a longer off duration Beep ON-Hook or dials a Timer
repeated every second; command.
played to prompt
entering of fresh
commands during
programming.
Programming Continuous, fast beeps; Beep... Beep... Played for 3 seconds. Programming
Confirmation played to indicate that Beep Confirmation Timer
Tone system has received a
valid command and is
processing it.
Programming Fast beeps, played on a Too…Too…Too Played for 3 seconds. Programming Error
Error Tone wrong programming …Too Tone Timer
command being dialed.
a. In Store and Forward dialing, the digits are first stored in a memory location and then these are dialed on the trunk.
For example: When Least Cost Routing (LCR) is enabled, the system will store the dialed digits first, check the trunk
through which the call is to be routed and then dials the number on the appropriate trunk.
Tone standards vary with the country of application. For example, as per ITU standard, the Dial Tone for India
consists of 400Hz modulated by 25Hz, whereas it is 350+440Hz, without modulation, for USA/Canada. Further,
many countries use different frequencies and cadences for the same tone. For example, in the US, five different
frequency and cadence are used for Dial Tone.
ETERNITY offers the flexibility of setting the Call Progress Tone Generation (CPTG) type to match the country-
specific CPT standards established by ITU.
India being the default 'Region' for ETERNITY, the CTPG for India is set as default in the system.
For countries that use different frequencies and cadences for the same tone, e.g. USA, only one
frequency/cadence among the group is considered. See Table "Default CPTG Type".
How to configure
Configuration of Call Progress Tones involves three parameters: CPTG Type (Region), CPT related Timers, and
Dial Tone Type.
The country-specific CPTG type is set automatically by the system when the 'Region' is selected. However, if
required, you may change the CTPG type set by the system. To do this,
• Open Jeeves.
• Select the desired Dial Tone Type as Type 1 or Type 2 from the combo box.
• Click the System Timers and Counts link to open the page.
CPTG Dial tone 1 Dial Tone 2 Ring Back Tone Busy Tone
Region Region Cadence Cadence Cadence Cadence
Code Freq. Freq. Freq. Freq.
(sec) (sec) (sec) (sec)
1 Region1 440 Continuous 350+440 Continuous 350+440 0.4on 0.2off 440 0.75on 0.75off
0.4on 2.0off
2 Region2 400 Continuous 400 Continuous 400 0.6on 0.2off 400 0.5on 0.5off
0.2on 2.0off
3 Region3 350+440 Continuous 350+440 Continuous 440+480 2.0on 4.0off 480+620 0.5on 0.5off
4 Argentina 425 Continuous 425 Continuous 425 1.0on 4.0 off 425 0.3on 0.2off
5 Australia 425*25 Continuous 425*25 Continuous 400*25 .4on .2off 425 0.375on
.4on 2.0off 0.375off
6 Brazil 425 Continuous 425 Continuous 425 1.0on 4.0 off 425 0.25on 0.25off
7 Canada 350+440 Continuous 350+440 Continuous 440+480 2.0on 4.0off 480+620 0.5on 0.5off
8 China 450 Continuous 450 Continuous 450 1.0on 4.0off 450 0.35 on
0.36off
9 Egypt 425*50 Continuous 425*50 Continuous 425*50 2.0on 1.0off 425*50 1.0on 4.0off
10 France 440 Continuous 440 Continuous 440 1.5on 3.5off 440 0.5on 0.5off
How to use
It is important that users of ETERNITYNE also get acquainted with the different Call Progress Tones played by the
system, so that they understand the meaning of the terms used for various tones. Therefore, ETERNITY makes it
possible for users to listen to the various Call Progress Tones.
Demonstration of Tones
It is possible to demonstrate Call Progress Tones to users by dialing the SE commands from a DKP or an SLT.
By default, the system will play each tone as demonstration for 30 seconds. The duration of demonstration can be
changed by setting the 'Tone Demo Timer' to match user preference.
• Exit SE mode.
What's this?
When the WAN port of ETERNITY NE is connected to a public IP network, it is may be necessary to restrict traffic
to and from a particular IP address only.
With the feature 'Call Restriction based on IP Address', ETERNITY makes it possible to entertain requests on its
WAN port for predefined IP Addresses only,
How it works
For this feature to work,
• the "White List IP Address" list, i.e. a list of IP Addresses and their respective Subnet Masks from where
the traffic is to be allowed, must be configured.
• With flag enabled and the table configured, traffic coming from all IP Addresses, other than those
programmed in the White List, will be blocked.
• If the flag "IP Address based call traffic restriction" is enabled, but the White List IP Address Table is
blank, all incoming traffic will be rejected and it will not be possible to make VoIP calls to ETERNTY NE.
How to configure
• Create a White List Table. Make a three column table on a sheet of paper.
Make a list of IP Addresses from which you want to allow traffic. You are allowed to program a maximum of
5 IP Addresses. For each IP Address enter the corresponding Subnet Mask.
• Open Jeeves.
• Select the Enable IP Address based call traffic restriction check box to enable this feature.
What's this?
Call Taping allows extension users to record the telephone conversations they have with other extensions or
external numbers, without the opposite party coming to know about it.
This feature is useful for keeping records of important conversations. For this feature to work, the system must
have the Voice Mail System present.
Calls are taped in a common mailbox assigned to this feature. Extension users with access to the mailbox can
retrieve and listen to the recorded conversations.
To be able to record external incoming and outgoing calls, a list of phone numbers (both incoming and outgoing)
must be programmed in the Call Taping Profile. This Profile is assigned to the extensions on which Call Taping is to
be used.
Incoming calls without Calling Line Identification (CLI) can also be taped.
To be able to record internal calls, the Call Taping for Internal Calls must be enabled on the extension.
• Matrix Comsat is not responsible for any mis-/abuse of this feature by users.
How it works
• A and B are extensions. Both have Call Taping parameters configured in the Call Taping Profile assigned
to them.
A calls C
• The system matches the dialed number with the numbers configured in the Tape Outgoing calls made to
numbers in the Call Taping Profile assigned to A.
• The system finds a match. When speech is established, the system starts recording the conversation
between A and C automatically in Be's mailbox.
• The system matches the incoming number with the numbers configured in the Tape Incoming Calls
received from numbers in the Call Taping Profile assigned to B.
• On finding a match, system records the speech between D and B in E's mailbox.
• Call Taping Beeps will be played to D and B only if this feature is enabled.
• If an incoming call does not have any CLI, the system checks the flag ‘Tape Incoming Calls received
without CLI' in the Call Taping profile assigned to the extension.
A calls B
• The system checks if the Tape Internal Calls is enabled in the Call Taping Profile assigned to A.
• If the flag is enabled, the system records the speech between A and B in E's mailbox.
• Call Taping Beeps will be played to A and B only if this feature is enabled.
• If the flag is disabled, the speech between A and B will not be recorded.
• The same is done when B calls A. The speech will be recorded in E's mailbox.
Feature Interaction:
• Conversation Recording: If Call Taping and “Conversation Recording” both are enabled for an
extension, then priority is given to Conversation Recording.
How to configure
To provide this feature to extensions,
• Call Taping Profile must be configured. A Call Taping Profile consists of the following parameters:
• List of Incoming and Outgoing phone numbers for which the system should initiate Call Taping.
• Internal Call taping flag.
• Flag for taping Incoming Calls received without CLI.
You can configure four different Profiles and assign different profiles to different extensions.
• Call Taping Profile must be assigned to the extensions which are to be provided this feature.
• If required, you may disable the Beeps played when Call Taping starts.
• Open Jeeves.
• In the Tape Incoming Calls received from following numbers column, enter the phone numbers of
external callers whose speech you want to tape.
• In the Tape Outgoing Calls made to following numbers column, enter the phone numbers of external
parties dialed by extension users whose speech you want to tape.
• To tape calls between extensions, select the Tape Internal Calls check box. Default: Disabled.
• To tape incoming calls received without a CLI, select the Tape Incoming calls received without CLI
check box. Default: Disabled.
• Now, assign the Call Taping Profile(s) you configured to the desired extension types. For instructions on
configuring Call Taping on different extension types, see “SLT Extensions”, “DKP Extensions”, “SIP
Extensions” under Basic Settings.
• To assign a mailbox for Call Taping, click the System Parameters link. The System Parameters page
opens.
• On this page, under System Parameters, go to Mailbox for Call Taping, select the Extension Type—SLT,
DKP, SIP—whose mailbox you want to use for Call Taping from the drop down list, and enter the Port
Number of this extension. For example, if you want to select the mailbox of the phone connected to SLT
Extension Port 4, enter 4 in this field.
For SLT extensions, the valid Port numbers are 1 to 14. For DKP extensions, the valid Port numbers are 1
and 2. For SIP Extensions, the valid Port numbers are 1 to 16.
• If required, you may disable the Beeps played when Call Taping starts by clearing the Play Beep when
Call Taping/ Conversation Recording starts check box on this page.
Call Taping conversations are recorded in a single, common mailbox. These can be accessed directly by the
Mailbox Owner (user of the extension to which this common mailbox is assigned). Other extension users can also
access this common mailbox by calling the Voice Mail System.
Instructions for accessing the mailbox are provided separately for these two groups of users.
OR
If the common mailbox is password protected, make sure that you provide the password to all extension
users who are to be provided access to this mailbox.
The above instructions contain the default access codes. If these have been changed and use the current
access codes.
22. This is the default Voice mail Feature Access Code. If this has been changed and use the new code.
23. Only if the mailbox is password protected, you will be prompted to enter the password.
24. See previous note.
What's this?
Call Toggle allows you to have two simultaneous telephone conversations, talking to two persons alternately.
Call Toggle is also referred to as Hold-Consult or Call Splitting. You can toggle between:
How it works
• A, B, and C are extensions.
• A is in speech with B.
• A wants to talk to C.
• A is in speech with B.
• A wants to talk to D.
• A is in speech with D.
• A wants to talk to E.
• A gets Ring Back Tone and D is put on hold. D gets music on hold.
• The party put on hold during Call Toggle cannot hear the conversation between the other two parties.
• You can also toggle between an incoming internal/external call (indicated by call waiting tone) and an
internal/external call you are currently in speech with.
• You can also answer an incoming 'Interrupt Request' call and toggle between the interrupting extension
and the extension you were in speech.
• You can convert a Call Toggle into a three-party conference by dialing Flash-0.
• You can transfer the call you are currently in speech with to another extension.
• You can park the call you are currently in speech with.
How to configure
Call Toggle is a “Class of Service (COS)” dependant feature. By default, Call Toggle is included in the ‘Basic
Features’ allowed to all extensions of ETERNITY. You cannot disable this feature selectively in the COS of
extensions, without disabling the entire set of Basic Features.
No other configuration is required for this feature, except for programming a DSS key for Call Toggle, if required, on
the DKP and Extended IP Phone extensions. See Phone Key Settings under “DKP Extensions” and Phone Key
Settings under “Matrix Extended IP Phone Key Settings” in SIP Extensions.
OR
• Dial Hold-1.
• Speech with extension 1.
• Press DSS key assigned to Call Toggle again
OR
• Dial Hold-1.
• Speech with extension 2.
OR
• Dial Hold-1.
• Speech with extension.
• Press DSS Key assigned to Call Toggle
OR
• Dial Hold-1.
• Speech with external party.
• Dial Hold-1.
• Speech with external party 1 on trunk 1.
• Press DSS Key assigned to Call Toggle again.
OR
What is this?
Call Transfer enables you to relocate an existing call from an extension or trunk to another extension or to an
external number. Calls can be transferred after notifying the other extension/external number about the impending
transfer or calls can be transferred directly without notification.
• Call Transfer – Screened: The Operator puts the caller on hold, dials the desired party's extension, and
informs the desired party of the impending transfer. If the desired party chooses to accept the call, the call
is transferred.
• Call Transfer – While Ringing: The Operator puts the caller on hold, dials the desired party's number and
transfers the call when the desired party's extension starts ringing.
This feature is used when there are several other calls to be attended and the Operator cannot wait for the
desired party to answer.
• Call Transfer – On Busy: The Operator puts the caller on hold, dials the desired party's number and
transfers the call even when the desired party is busy in speech with another person. The busy extension
gets intrusion tone and can choose to answer the intruding (transferred) call.
• Call Transfer – Trunk-to-Trunk: An external call is transferred on to another trunk line. The Operator puts
the external caller on hold, dials the desired party's external number, and transfers the call after or without
notifying the desired party of the impending transfer.
Trunk-to-Trunk call transfer may be used to transfer incoming calls for out-of-office extension users to their
cell phones, or to connect personnel at remote or distant locations. For instance, an out-of-office executive
who does not have long distance dialing permission can call the office and request the operator to connect
him to the desired party on a trunk line.
• Blind Transfer to VMS: The Operator puts the caller on hold, dials the feature access code for Blind
Transfer to VMS, dials the desired party's number, and transfers the call. The call is transferred to the
mailbox assigned to the desired party. The caller may leave a message in the mailbox.
Call Transfer is not exclusively an Operator feature, though it is used mostly by Operators. Calls can be
transferred by any extension to another extension or external number, if Basic Features are allowed in
Class of Service of the transferring extension.
How it works
A and B are extension users.
C is an external caller.
D is an external number.
1. Screened Transfer:
• If B does not accept the call, Operator may dial Flash to retrieve the call and speak to C.
• The Operator can also abort call transfer while B's phone is ringing by dialing Flash. The Operator gets
connected to C.
3. Transfer On Busy:
• If B does not answer the intrusion beeps at the end of the Transfer on Busy Timer, the call is returned
to the Operator. C gets ring back tone.
• If the Operator too is busy at the time of call return, C gets busy tone.
OR
Transfer the call as soon as D's phone starts ringing. (transfer while ringing)
• C and D are now in speech for the duration of the Trunk-to-Trunk Inactivity Timer25.
• A warning tone is given at the end of the Trunk-to-Trunk Inactivity timer (programmable; default: 90
seconds). On expiry of this timer, the call is disconnected.
• To extend the call, either C or D must dial any digit in tone (DTMF), except '##'.
Dialing ‘##’ to extend the (Trunk to Trunk Transfer) call will result in Call Disconnection.
In the case of Trunk-to-Trunk transfer, both parties in speech on trunk lines must be informed that their call
would be disconnected at the end of the Trunk-to-Trunk Inactivity Timer and that they must dial any digit,
except ‘##” to extend the call.
• If A does not have a mailbox assigned, the Operator will get an error tone while transferring the call.
• The Operator may retrieve C's call by pressing Hold/Flash/Call Appearance key.
Feature Interactions:
• CLIP and Caller ID Presentation while Transfer: ETERNITY provides the flexibility to display either the
extension number that is transferring the call or the held party's number, i.e. the number of the party that is
about to be transferred. Refer “Calling Line Identification and Presentation (CLIP)”.
• Privacy: Call Transfer-On Busy will not work if the busy extension has Call Privacy from intrusion Tone in
its Class of Service.
• DND: Call Transfer will not work if the destination extension has set DND.
25. The process of Trunk-to-Trunk transfer takes place outside of the PBX. So, the PBX will not know which of the two trunks have
gone ON-Hook. Hence the Trunk-to-Trunk Inactivity Timer. The call is automatically disconnected when this timer expires.
• This feature must be enabled in the Class of Service of the extension(s). Call Transfer included in the
'Basic Features' allowed to all extensions of ETERNITY NE. So, all extensions of ETERNITY can use this
feature.
You cannot disable 'Call Transfer' selectively without disabling the entire set of 'Basic Features'. See
“Class of Service (COS)” and to know more.
• You may change the default values of the following timers related to Call Transfered, if required:
• Transfer While Ringing Timer: It is the time for which the system rings the extension. By default it is
set to 30 seconds. At the end of the timer the call is returned to the transferring extension.
• Transfer on Busy Timer: It is the time for which the system waits for the busy extension to respond to
the intrusion tone. By default the timer is set to 30 seconds. At the end of the timer the call is returned
to the transferring extension.
• Trunk to Trunk Inactivity Timer: This is the time duration after which the system disconnects the call
transferred from one trunk line to another. By default it is set to 90 seconds. At the end of the timer the
call is disconnected, if either party does not dial digits to extend the call. This Timer is relevant for
analog trunks, i.e. CO Trunk-to-CO Trunk calls only.
To change the values of these timers, see “System Timers and Counts” for instructions.
• The extensions must be assigned a personal mailbox in their Voice Mail Settings. The option Assign
Personal Mailbox must be enabled in the Voice Mail Settings of the extension.
• See “Basic Settings” for instructions on configuring the Voice Mail Settings of the different Extension types:
“SLT Extensions”, “DKP Extensions” and “SIP Extensions”
How to use
OR
• Press Hold.
• Dial-Trunk Access Code26- External Number and go ON-Hook.
OR
• Press Hold and dial desired party's extension number and Go-ON Hook.
OR
OR
• Dial 1078.
• Dial desired party's extension number.
• Go Idle.
26. Trunk Access Code: users worldwide may dial a code from 0, 5, 61, 62, 63, and 64. Users in USA may dial a code from 0, 9, 81, 82,
83, and 84.
Extension to Trunk:
What's this?
The ETERNITY provides the facility of detecting the caller's number and presenting it on the display of the called
extension phone. This feature is called Calling Line Identification and Presentation (CLIP).
The calling number can be presented on DKP and Extended IP Phones. The calling number can also be presented
on SLTs that support CLI protocols.
The signaling protocols for CLI supported by ETERNITY are: DTMF, FSK V.23, and FSK-Bellcore.
These protocols are supported on trunks as well as extensions. Any type of trunk line and supporting DTMF or FSK
signaling can be interfaced with the ETERNITY.
Similarly, any type of telephone instrument supporting DTMF or FSK signaling protocol can be connected to the
SLT port.
How it works
When CLIP is enabled on a trunk,
• It sends this information to the landing extension/Operator along with the ringing signal.
• In case of, Internal calls the calling extension's name and number both are presented to the called
extension.
• In the case of External calls, only the number will be displayed on the landing/Operator extension.
• When the landing extension/Operator transfers the incoming call to an extension, putting the external
caller on hold, the system sends this information to the extension to which the call is transferred.
• During the transfer, the number of the landing extension/Operator will be displayed on the transfer
destination extension.
• On successful call transfer, the caller's number will be displayed on the transfer destination extension.
In the case of Call Transfer, the system also provides the option of displaying to the transfer target extension either
the number of the Transferring Party or the number of the Party put on hold to be transferred, while the call is being
transferred. This feature is called Caller ID Presentation on Call Transfer.
It is also possible to remove and replace the '+' character received as CLI on telephones that do not support CLIP
starting with this character.
For example, the mobile network sends the calling party number with '+' as the prefix. If the telephone connected as
extension does not support this, it will not present the CLI of the caller. To overcome this, ETERNITY NE provides
you the option of replacing '+' with an appropriate number string which these telephones can display.
How to configure
The functioning of this feature is controlled by two parameters: CLIP Type and Caller ID on Call Transfer.
If you want to replace '+' characters received as CLI on telephones that do not support CLI prefixed with this
character, you may configure the Replace ‘+’ from CLI option in the “System Parameters”.
If SLTs supporting CLI are connected to the ETERNITY, you must select a signaling protocol for CLI for the SLT
extensions. TMF is selected as CLIP Type for all SLT extensions.
• Open Jeeves.
• Click the SLT Extensions link. The page of the first SLT extension opens. Click more.
• Display number of Party kept on Hold when call is transferred on this Extension
• To configure CLIP Type and Caller ID on Call Transfer on another SLT extension, click the desired
extension number tab and follow the instructions as above.
To replace '+' characters received as CLI on telephones that do not support CLI prefixed with this character,
• On this page, under System Parameters, select the Replace '+' from CLI check box.
• In the Replace '+' from CLI with the number string field, enter the desired number string that the system
should replace the ‘+’ in the string with.
What's this?
The ETERNITY allows extension users to suppress their identity, i.e. extension number and name, when they call
other extensions. This feature is called Calling Line Identity Restriction (CLIR).
Extensions that have 'CLIR Override' facility can view the CLI of those that have suppressed it with CLIR.
This is a feature of the PBX and not the PSTN. It is applicable for internal calls only.
Needless to say, this feature will work only on the proprietary digital key phone EON and SLTs that support Caller
Line Identification (CLI).
How it works
• Extension A has CLIR enabled.
• Extension B does not have CLIR Override enabled.
• Extension C has CLIR Override enabled.
• When A calls B, B cannot view the extension name and number of A.
• When A calls C, C can view A's extension name and number.
Now,
• Extension D calls extension E.
• A picks up the call.
• D will be able to view A's name and extension only if it has CLIR Override enabled and is a digital key
phone, EON.
CLIP and Caller ID Presentation while Transfer: Both these features will not work if CLIR is enabled.
How to configure
'CLIR' and 'CLIR Override' are Class-of-Service-dependant features. Extensions that are to be allowed these
features, must have them enabled in their “Class of Service (COS)”.
By default, both these features are disabled on all extensions of ETERNITY. Thus, none of the extensions of the
ETERNITY can suppress their CLI or force any other extension to display its CLI.
Decide which extensions should be allowed CLIR and which should be allowed CLIR Override and accordingly
configure their COS. See “Basic Settings” for instructions on configuring COS of the different Extension types: SLT,
DKP, SIP.
OR
To disable CLIR:
OR
• Dial 1030.
• You get confirmatory tone and message on the phone's display.
• Go idle.
• Lift handset.
• Dial 103-1.
• You get confirmation tone.
• Replace handset.
To disable CLIR:
• Lift handset.
• Dial 103-0.
• You get confirmation tone.
• Replace handset.
27. You are recommended to assign a DSS Key with LED to this feature. When the assigned DSS key is pressed, it will glow red indi-
cating that CLIR is enabled.
28. If a DSS key with LED has been assigned, when you press the key again, the LED will be turned off indicating CLIR is now
canceled.
What's this?
For each feature of the ETERNITY NE that an extension user has set/enabled on the extension, the system
provides access code for cancellation of the feature.
However, there is also a single master command for extension users with which all features that are set on an
extension can be canceled.
When the extension user dials 'Cancel All Extension Features' command, the following features, if set, are
cancelled from the extensions:
• Alarms
• Auto Answer
• Auto Call Back
• Auto Redial
• Background Music
• Call Follow-Me
• Call Forward
• Do Not Disturb
• Hot Line
• Hot Outward Dialing
• Trunk Reservation
• Walk-In Class of Service
Auto Redial, Background Music, Call Forward and Call Follow-Me, Do Not Disturb, Hotline and Trunk
Reservation are Class of Service dependent features. These features can be set/canceled from an
extension only if these are included in its “Class of Service (COS)”.
How to use
OR
• Dial 1051.
• You get confirmation tone and confirmatory message on your phone display.
• Go idle.
What's this?
Class of Service (CoS) defines the permission an extension will have on a PBX. It defines the set features of the
PBX that the extension is to be allowed access to.
Feature requirements vary among users and with time. Certain groups of extension users may have a need for
voice mail, while another group may need the ability to forward calls to a cell phone, and still others may have no
need to make calls outside the office.
Similarly, certain features that are required during working hours may not be required during non-working hours.
ETERNITY NE offers the flexibility to allow or deny extension users access to features of the PBX, on the basis of
their requirement and according to time of the day. For users, access to various features from their extensions is
their Class of Service (Cost).
How it works
Each extension port of the PBX has an associated CoS for the Day and Night, that indicates which features of the
PBX the port is allowed to access during working hours and during non-working hours.
A feature can be allowed or denied to an extension by enabling or disabling it in the CoS of that extension.
The CoS of all extensions may be uniform, or different CoS can be assigned to different extensions, according to
the time of the day. Doing so, each extension can access a different set of features during the Day (working hours)
and at Night (non-working hours).
Basic Features
A set of features including Internal Call, Call Hold, Call Toggle, Call Transfer, Department Call, Operator Access,
Redial, and Call Mute defined as Basic Features and allowed in the CoS of all extensions.
It is not possible to enable or disable selectively any of the features included in "Basic Features".
How to configure
The table below presents the list of features supported on the extensions for the Day and Night.
Default CoS
Allowed During
Feature
Day Night
Account Code
Auto Redial
Barge-In
Basic Featuresa
Call Forward
Call Park
Call Pickup
Call Recording
CLIR
CLIR Override
CO Call Waiting
Conference
Continued Dialing
DISA
Do Not Disturb
Dynamic Lock
Forced Answer
Forced Release
Hot Desk
Hotline
Interrupt Request
Paging
Raid
Room Monitor
SA Extension
SA Mode
Trunk Reservation
Trunk-Trunk Transfer
a. Basic Features includes: Internal Call, Call Hold, Call Toggle, Call Transfer, De-
partment Call, Operator Access, Redial, Mute.
• Against each extension name/number on the list, write the features needed for the Day and Night. You will
notice that the features needed by many extensions are identical.
• List the common features to be allowed to and features to be denied to all SLT, DKP, SIP extensions.
• Are there any other features, in addition to those on the common list, which you want to allow to select
extensions?
• If yes, extend the common list you prepared by adding the features to be allowed to the select extensions.
• Now, to configure the CoS features to be allowed or denied to extensions, go to “Basic Settings” of Jeeves,
• Click the extension settings link for SLT, DKP, SIP Extension.
The check boxes of features that are allowed for the Day and Night are selected.
The default CoS fulfills the requirements of most extension users. Check if the default CoS features
you want to allow are enabled and features you want to deny are disabled
• To allow a feature for the Day/Night, select its corresponding check box.
Finally, test the CoS you configured for each extension by invoking the features from each extension.
What is this?
ETERNITY offers the facility to detect the calling party's number and name. This is known as Calling Line
Identification.
On the basis of CLI, it is possible to land calls from a particular telephone number on a particular extension or group
of extensions. This is known as CLI Based Routing.
How it works
A, B, C are extensions. D and E are external callers.
Calls made by D are to be landed on A
Calls made by E are to be landed on B and C.
The CLI of D and E and their corresponding landing destinations should be programmed in the CLI Based Routing
Table.
If D's number is not programmed in the CLI Based Routing Table, the call will be routed according to the incoming
call routing configured for the trunk.
How to configure
For this feature to work, the CLI Based Routing Table must be configured with the numbers of the calling parties
and the numbers of the corresponding destination extensions.
You can program up to 400 numbers in the CLI Routing Table. Each calling party number in the CLI table is stored
against an index from 001 to 400.
: :
For each Calling Party Number, select the corresponding landing destination. The landing destination may
be an SLT, DKP, SIP or Virtual extension.
• Open Jeeves.
There are 100 entries on each page. To go to the next 100 Index, click the numbered links at the top of the
CLI table (101-200, 201-300, 301-400).
• Refer to the table you created on paper, and configure the following parameters.
• In the Name field, enter the name of the calling party. You can enter a maximum of 8 characters in this
field.
• In the Route to field, select the landing destination extension number, which may be SLT, DKP, SIP or
Virtual extension.
What’s this?
A Closed User Group is a network of PBXs to provide seamless connectivity. The PBXs connected in the network
behave as a single unit. Extension users of one PBX can reach the extension users of the other PBX without dialing
any access code, as if they were dialing extension numbers of their own PBX.
You can connect ETERNITY NE and other PBX(s) to over Analog trunks as well as over the IP Network.
How it works
Let us understand this application with the help of this illustration:
CO CO
Location A Location B
CO1 CO1
CO2 SLT1
SLT1 CO2
201
SLT2 SLT2 301
202 ETERNITY
PBX A
SLT3
SLT3 302
203 115.118.161.165
SLT4 121.124.130.110
SLT4 303
Ethernet Ethernet
IP
Menu Sat 01 05: 30
204 WAN
DKP1 WAN SLT5
FwdBus
y FwdNRCan
cel Mute Con
f erenc
e Trans
f er
304
DND VoiceMail Names Red
i al Rejec
t Hold
1 2 abc 3 def
4 ghi 5 k
jl 6 mno
CLI R
7 pqrs 8 t uv 9 wxyz
CA4
il e CA3
Hot n
* 0 #
SI P2 CA2
SI P1 CA1
Broadband Broadband
Router/Modem Router/Modem
Here,
• To create a CUG over IP Network, both ETERNITY NE and PBX A may be connected directly to the
Public network, in Peer-to-Peer mode.
• Seamless connectivity can be provided to the extension users at both locations with suitable configuration
of ETERNITY NE and PBX A.
The CUG Table stores up to 8 entries. You must configure the following parameters in the table:
• Route Code: The Route Code is the truncated number string for the extension numbers of the other
PBX. The truncated number string must start with the starting digit of the extension numbers. In this
case, all extension numbers of PBX A start with ‘3’, so you can configure the Route Code as a single
digit, ‘3’.
• Strip Digit Count: This parameter is not relevant for this application. Keep the Strip Digit Count as ‘0’.
• Self Route: This flag is not relevant for this application; keep it disabled.
• Dialed Digit Count: This is the digit length of the extension numbers of PBX A29. Here all extension
numbers of PBX A have three digits, the Dialed Digit Count should be configured as ‘3’.
• Route using Trunks: This is the number of the Tie Trunk that the system should use for routing the
dialed number string that matches with the Route Code and Dialed Digit Count.
It the CUG is connected over Analog Trunks, you must select the CO Trunk you have connected to
the SLT (FXS) port of PBX A.
If the CUG is connected over IP network, you must select the SIP Trunk which you have used for this
application.
The CUG Table you configure on ETERNITY NE would look like this:
Strip Dialed
Route Index Route Code Digit Self Route Digit LCR Route using Trunks
Count Count
• At Location B, you may do a suitable configuration of the PBX. If the PBX at Location B were ETERNITY,
the CUG Table you configure would look like this:
Strip Dialed
Route Index Route Code Digit Self Route Digit LCR Route using Trunks
Count Count
29. When digits are dialed on a trunk, the system waits for the Inter Digit Wait Timer after the last digit is dialed. The Dialed Digit Count
helps avoid the delay caused by the Inter Digit Wait Timer. When you configure the digit length, the system will match the dialed
number string with the digit length you have configured and will route the number without waiting for the Inter Digit Wait Timer, if a
match is found.
If the CUG is connected over IP network, you must also configure the SIP Trunk as Peer-to-Peer trunk
and configure the Peer-to-Peer Table. See “Peer-to-Peer Calling” for instructions.
How to configure
To configure Closed User Group,
• Open Jeeves.
• Click the Closed User Group link. The Closed User Group Table page opens.
• In the Route Code field, enter the extension numbers of the other PBX in the CUG. Instead of entire
number strings, you can configure a single digit, the starting digit of the extension numbers as Route Code.
A maximum of 16 digits can be configured as Route Code. Default: Blank.
• In the Dialed Digit Count field, enter the digit length of the extension numbers of the other PBX in the
CUG. The digit length may be up to 99. Default: 99.
• Double click the Route using Trunks field. A multiple selection box opens.
• On the left, the trunks appear with their names (if configured in “Naming Trunks”) and port numbers in a
sequence, starting with CO trunks, followed by Mobile trunks and SIP trunks.
If you have not assigned any names to the trunks, they will appear with their default names (CO, MOB,
SIP) and port numbers.
If you have enabled On-Site Configuration, only those trunks that are connected will appear in the box.
• To select a trunk, place your cursor on the desired trunk, and click the Select>> button.
Or
• Press the ctrl key and click the left mouse button to select multiple trunks.
• You may change the sequence of the trunks you selected, if required, using the Up and Down arrow
buttons on the right display box.
• You can also delete trunks from the ones you have selected.
• You may enable Rotation, if you have selected more than one trunk. Default: Disabled.
When you enable Rotation, each new outgoing call is routed through the subsequent trunk in the
group30. This ensures equal distribution of outgoing call traffic on all trunks.
30. The first call through the first trunk, the second through the second, the third through the third trunk, and so forth. Thus each new
call is routed through the trunk next to the one that routed the previous outgoing call.
• All the trunks appear in the field Route using Trunks, in the sequence you selected, separated by
commas.
What's this?
The CO Call Waiting feature gives indication to the extension user about another waiting call on a Trunk, when this
extension is in speech.
This is a “Class of Service (COS)” dependent feature. Only those extensions which have this feature enabled in the
COS allowed to them, will be given incoming call waiting indication on the trunk.
How it works
• A and B are extensions.
• CO Call Waiting feature is enabled in the Class of Service of B but not on A.
• There is an incoming call on a trunk for B.
• B is busy on a call with A.
• ETERNITY plays Beeps to B to indicate the call waiting.
• To answer the waiting call, B may dial Flash or press the HOLD key.
• B will be connected to the caller.
• A will be put on hold.
• When there is an incoming call on a trunk for A, but A is busy on another call, A will not be provided any
indication of the waiting call.
How to configure
For CO Call Waiting to work on an extension, it must be enabled in the Class of Service allowed to that extension.
By default, CO Call Waiting is disabled in the Class of Service of all Extension types: SLT, DKP, SIP. You may
enable this feature on extensions which you want to provide CO Call Waiting indication.
For instructions, see “Class of Service (COS)”, and the description for configuring the different Extension types
under “Basic Settings”.
Keep this feature disabled, if you want to provide Privacy from Trunk Call intrusion beeps to an extension type. See
“Privacy” to know more.
What’s this?
In a Dial-In, users can schedule a conference by dialing a conference number and a password and inform other
participants about the conference. The other participants can dial in to the on-going conference call to join the
conference.
Extension users can also schedule and join a Dial-In Conference from a remote location using “Direct Inward
System Access (DISA)”.
Conference Dial-In is useful for businesses to conduct client meetings or sales presentations, project meetings and
updates, regular team meetings, and communication to coworkers who operate in different locations. Besides
increased convenience, this feature allows workers to be more productive by saving time and cost of travel for out-
of-office meetings.
The ETERNITY NE supports up to 6 participants in a Dial-In Conference. At a time, two simultaneous Dial-In
conferences can be held.
How it works
A, B and C are extension users. C is at a remote location.
D, E and F are external callers.
• If C wants to schedule a conference, C must log into his extension from DISA mode.
• The Conference Number must correspond with the number of simultaneous Dial-In Conferences
supported by the ETERNITY, in this case: 1 and 2. If a user dials a conference number other than this,
system will play an Error Tone.
• When a new party joins the conference, the system plays beeps to the existing users, to inform them of
the new inclusion.
• Beeps are programmable. You can disable or enable beeps during conference.
Any extension user in a Dial-In Conference can include any other extension user or external party in the conference
by dialing the Conference Include feature code.
• If the conference has been initiated from the DISA mode, the caller (remote user) must dial the code for
withdrawing from the conference.
• If a participant goes ON-Hook, without dialing the feature code for withdrawing from a Multi-party
Conference, the participant will be included in the conference again, when the participant dials the
Conference Inclusion feature code (191). To avoid this, participants must dial the feature code for
Withdrawing from conference.
• If all participants of a Dial-In Conference have withdrawn from the conference, one-by-one, but none of
them have dialed the feature command to terminate the conference (190), the system will start the
'Release Conference if idle for more than (Minutes) Timer'. This Timer is programmable, and by default
it is set to 002 Minutes. On the expiry of this Timer, the system will free the resource occupied by the
conference on the conferencing circuit. However, any participant can join the conference before the
Conference Release Timer expires.
When an extension user cancels a conference, the system disconnects all the participants from the
conference, but does not free the resource occupied by the conference. To free the resource occupied by
the conference, you must release the conference (see Releasing a Dial-In Conference).
A Dial-In Conference can be canceled from SA mode also. This is useful when the participants forget the
password and cannot join the conference or cannot initiate the conference.
How to configure
For this feature to work,
• the feature 'Conference' must be enabled in the“Class of Service (COS)” of the extensions. By default, this
feature is enabled on all extensions, so all extensions can use this feature. See “SLT Extensions”, “DKP
Extensions”, “SIP Extensions” under “Basic Settings”.
The feature 'Conference' in the Class of Service also includes 3-Party and Multi-party Conference.
Extensions that are denied 'Conference' in their Class of Service will not be allowed Dial-In as well as 3-
Party and Multi-party Conference.
• If desired, you may also change default value of the Release Conference if Idle for more than (min) Timer.
See “System Timers and Counts”.
• If extension users are to be allowed to initiate or join the Multi-party Conference from a remote location,
“Direct Inward System Access (DISA)” must be enabled on the trunk on which they call.
How to use
To cancel a Conference:
• While you are all in speech, go ON-Hook.
• Go OFF-Hook.
• Dial 190.
You can also release a Dial-In Conference from SA Mode using Jeeves. To do this,
• Open Jeeves.
• Enter the conference number (1 or 2) which you want to cancel in the Cancel Dial-In Conference
Number field.
When you enter DISA mode, you get beeps, dial digits before the DISA Inactivity Timer elapses.
Never dial 'Flash' when in DISA mode, you will get disconnected.
To schedule a conference
• Dial 194-Conference Number-Conference Password.
• You get confirmation tone.
• Replace Handset.
• Inform all intended participants about the time, conference number and password.
What's this?
ETERNITY offers three types of conference calls: Conference-3 Party, “Conference Dial-In”, and “Conference-
Multiparty”.
Conference-3 Party (also referred to as Three-Way Calling) is a telephone call, in which the calling party can have
two other persons participate in the call.
A 3-Party Conference is initiated by dialing the number of the first person one wishes to talk to. The first person is
informed about the conference and put on hold. The number of the second person one wishes to talk to is dialed.
When the second person answers, s/he is informed about the conference. Three-way speech is established by
pressing Flash-0.
An already connected two-way speech can be converted into a conference by adding a second person, without
disconnecting the call with the first person.
Thus, a 3-Party Conference may be planned or conducted on the spur of the moment.
A 3-Party Conference can be conducted with extensions of ETERNITY and between extensions and external
numbers.
It is also possible to conduct an Unsupervised 3-Party Conference, wherein the operator connects two trunks
through the system and withdraws from the three-way speech.
How it works
A, B, C are extensions.
D and E are external numbers.
How to configure
For this feature to work, the feature 'Conference' must be enabled in the Class of Service group of the extensions
that are to be allowed this feature. By default, all extensions of ETERNITY are allowed this feature in their COS and
can make Conference calls. If you want allow this feature to certain extensions only, see “Basic Settings” for
instructions on configuring “Class of Service (COS)” for the different Extension types: SLT, DKP, SIP.
If extension users at remote locations are to be allowed to initiate the 3-party conference, “Direct Inward System
Access (DISA)” or “Direct Inward Dialing (DID)” must be enabled on the trunk on which their call lands.
The feature 'Conference' in the Class of Service also includes Dial-In and Multi-party Conference.
Extensions that are denied 'Conference' in their Class of Service will not be allowed all three types of
conferences: 3-Party, Dial-In and Multi-party Conference.
How to use
• Dial #
• Dial 0 to grab a trunk. You get Trunk dial tone.
• Dial telephone number of Party 2. You get ring back tone.
• Speech with Party 2.
Now,
• Press DSS key assigned to 'Conference'.
OR
If Party 2 is a trunk,
• Dial #
• Dial 0 to grab a trunk. You get Trunk dial tone.
• Dial telephone number of Party 2. You get ring back tone.
• Speech with Party 2.
• Dial Flash-0
• Three-way speech is established.
What’s this?
Like the Dial-In Conference, a Multi-party conference allows speech between more than three participants.
The key difference between Dial-In and Multi-party conference is that in a Dial-In conference participants can
include themselves in the conference by dialing into it without assistance, whereas in a Multi-party Conference the
party initiating the conference must include the participants by dialing their numbers and the Multi-party Conference
feature code.
Extension users can initiate multiparty conference from a remote location using “Direct Inward System Access
(DISA)”.
ETERNITY NE supports 6 participants in a Multi-party Conference. The system supports 2 simultaneous Multi-
party Conferences.
How it works
A, B, C, and D are extension users.
E and F are external numbers.
• Having included the last participant, F, A dials conference inclusion code again.
• A is included in the conference.
• All participants can now converse.
If the conference has been initiated from the DISA mode, the caller (remote user) must dial the code for
withdrawing from the conference.
If participant goes ON-Hook, without dialing the feature code for withdrawing from a Multi-party
Conference, the participant will be included in the conference again, when the participant dials the
Conference inclusion feature code (191). To avoid this, participants must dial the feature code for
Withdrawing from conference.
If all participants of a Multi-party Conference have withdrawn from the conference, one-by-one, but the
none of them have dialed the feature command to terminate the conference (190), the system will start the
'Release Conference if idle for more than (Minutes) Timer'. This Timer is programmable, and by default it is
set to 002 Minutes. On the expiry of this Timer, the system will free the resource occupied by the
conference on the conferencing circuit.
Canceling a Conference
Any participant in a conference can dial the Cancel conference code (190) to end the conference, all participants
will get Error Tone and the system resource occupied by the conference will be freed.
How to configure
For this feature to work,
• the feature 'Conference' must be enabled in the“Class of Service (COS)” of the extensions. By default, this
feature is enabled on all extensions, so all extensions can use this feature. See “SLT Extensions”, “DKP
Extensions”, “SIP Extensions” under “Basic Settings”.
The feature 'Conference' in the Class of Service also includes 3-Party and Multi-party Conference.
Extensions that are denied 'Conference' in their Class of Service will not be allowed Multi-party as well as
3-Party and Dial-In Conference.
• If desired, you may also change default value of the Release Conference if Idle for more than (min) Timer.
See “System Timers and Counts”.
How to use
What's this?
You may recall that “Access Codes” are dialed at different call phases. No two Access Codes must be the same in
the same call phase.
For example, the same access code cannot be used for two different features like Call Forward and Redial, since
both these features are invoked in the 'Dial' phase. Similarly, Station and Logical Group Codes too must be unique
and should not match with any of the features invoked in the 'Dial' phase.
However, ETERNITY allows overlaps within Feature Codes and Station Codes (Extension Numbers). One Feature
Access Code can be a part of (subset) another code, e.g. 4, 41, 412, etc.; extension numbers can be 201, 2011 etc.
So, when such overlapping access codes are dialed, the system matches the first digit. On finding more than one
Access code starting with the same digit, the system will not know how to interpret the instruction and act
accordingly.
Conflict Dialing feature resolves this. When an access code that is a subset of any other access code is dialed, the
system waits for some time for the extension user to dial the next digit. If the user does not dial any digit within that
time, the system interprets it as the smaller Access Code, and invokes the associated feature.
The time for which the system waits for the next digit to be dialed before resolving the Access Codes is called
"Conflict Dialing Timer". This timer is set to 2 seconds and is programmable.
How it works
You may set,
• If A does not dial any other digit before the Timer elapses, the system interprets the code as '41' and
invokes the Alarm feature.
• If A dials '2' before the Timer elapses,
• If such access codes exist, the system again waits for the duration of the Conflict Dialing Timer for another
digit to be dialed.
• Thus, only when the conflict in the access codes is resolved will the system respond accordingly.
How to configure
The working of this feature is controlled by the Conflict Dialing Timer, which is set by default to 2 seconds and can
be changed as desired.
If the duration of the Conflict Dialing Timer is long, it may cause delay in the system's response to the
feature. If the duration is less, the system may misinterpret the access codes. Ensure that the value of the
Timer is programmed optimally (i.e. at least the default value).
• Scroll to Other Features. Set the Conflict Dialing Timer to the desired value.
What's this?
Conversation Recording allows extension users to record their talk with other extension users or external parties,
after or without informing the opposite party.
This feature can be used to record verbal agreements, important discussions, instructions, interviews, client
requirements, take or place orders, etc.
For this feature to work, the Voice Mail System must be present in the system and extensions must have a mailbox
assigned to them for recording conversations.
• Matrix Comsec is not responsible for any mis-/abuse of this feature by users.
How it works
A and B are extensions. Both are assigned a mailbox each.
C and D are external parties.
• A calls C.
• C answers the call.
• A dials the command for Conversation Recording in mid-speech.
• C is put on hold.
• The system sends a string of digits to the Voice Mail System to initiate Conversation Recording.
• A and C are in speech again.
• The conversation recording starts in A's mailbox. The system plays beeps, if Conversation Recording
Beeps are enabled.
• A or C disconnects the call.
• Conversation recording ends.
• A can listen to the recorded conversation by invoking the voice mail feature.
The same is repeated when B calls A. As both have mailboxes assigned, both can record the conversation.
How to configure
To provide this feature to extensions,
• Conversation Recording must be enabled in the “Class of Service (COS)” of the extensions to which this
feature is to be allowed. By default, none of the extensions have this feature in their COS. Decide which
For instructions on configuring Conversation Recording related parameters different extension types, see
“SLT Extensions”, “DKP Extensions”, “SIP Extensions” under Basic Settings.
• Extensions that are to be allowed Conversation Recording must also have a mailbox. By default all
extensions are assigned a mailbox in their Voice Mail Settings.
If required, you may disable the Beeps played when Conversation Recording starts. Decide whether you want
Beeps to be played and accordingly enable/disable the Play Beep when Call Taping/ Conversation Recording
starts in the “System Parameters” under Advanced Settings.
How to use
OR
OR
Conversations are recorded as New Messages. So, follow the voice mail prompts for listening to new
messages.
31. This is the default Voice mail Feature Access Code. If this has been changed, use the new code.
What's this?
Customer Name is the name of the organization/enterprise that has deployed ETERNITY NE. As the User, you can
program the name of your company/organization in the system.
When Customer Name is programmed in the system, this name will appear as header on the various System
Reports generated and printed by the ETERNITY like SMDR Incoming, Outgoing and Internal Call Reports, Alarm
Status reports, etc.
The Customer Name may consist of a maximum of 80 alphanumeric characters, including punctuation marks. So,
you can add a contact address to the Customer Name.
How to configure
Customer Name can be programmed at the time of installation on the “Pre-requisites” page under Basic Settings,
or any time thereafter on the “System Parameters” page under Advanced Settings.
You can also change or correct the Customer Name you have assigned any time.
• Enter the name (and address, if desired) of the organization/enterprise in the Customer Name field. For
example: Prudent Investment, 701 Sunshine Boulevard, Bannerghatta, Bangalore.
What's this?
Certain features of the ETERNITY NE like Operator, Class of Service, Toll Control, Direct Inward Dialing (DID),
Direct Inward System Access (DISA), etc., require extensions and trunks to behave differently according to the
working hours and non-working hours, which are referred to as Time Zones.
These Time Zone-dependant features and facilities are operated automatically according to the Time Tables
programmed in the system. In a Time Table, the Time Zones—Working Hours, Non-Working Hours—are defined
for the entire week. Time Table is assigned to trunks, extensions and other time-zone dependant features. The
system executes the Time-Zone dependent features and facilities automatically according to the Time Table.
Day/Night Mode allows you to manually change the Time Zone of the system at any point in time, by issuing a
command. For example, the office is to be closed on account of an unplanned holiday or emergency. So, the Time
Zones of all extensions and trunks must be set to Non-working hours to route outgoing calls and land incoming
calls from/to the appropriate destination. You can set the ETERNITY to Night Mode until the office remains closed
and set it back to operate as per the Time Table, when work is resumed.
To cite another example, the office must work for extended hours. You can set the ETERNITY NE to Day Mode and
set it back to operate as per the Time Table.
When you set the system in Day/Night Mode, the system overrides the Time Tables assigned to Trunks,
Extensions and Operator. According to the mode you selected, it applies Working Hours/Non-Working Hours to run
all the Time-Zone dependent features of the system.
When the system is set to Day Mode, it applies Working Hours as the Time Zone for all extensions, trunks and time
zone dependant features and facilities. When the system is set to Night Mode, it applies Non-Working Hours as the
Time Zone on Time-Zone dependent features of the system.
Thus, Day/Night Mode forces the system to work in a particular Time Zone, until it is changed again, manually.
How to configure
Day/Night mode can be set by the System Engineer (SE mode) as well as by the System Administrator (SA Mode).
To set the system in Day/Night mode from the System Engineer mode,
• Open Jeeves.
• Dial 1072-018-Code
Where,
Code is from 1 to 3
1 is for Day Mode
2 is for Night Mode
3 is for Operate system as per Time Table.
• Exit SA mode.
What's this?
Daylight Saving Time (DST) is the practice of advancing clocks so that afternoons have more daylight and
mornings have less. Typically clocks are adjusted forward one hour near the start of spring and are adjusted
backward in autumn.
Many countries of the world use DST, though the start and end dates of DST vary with location and year. Even
within countries, uniform DST may not be observed. For example the states of Arizona and Hawaii do not observe
DST. Certain countries may observe DST in certain years, for instance Guatemala, while in most countries of Asia
and Africa, and in certain countries of South America, DST is not observed at all.
When ETERNITY is installed in a country/region where DST is used, it is necessary to synchronize the Real Time
Clock of ETERNITY NE with the local time.
So, if you are installing ETERNITY in a country where DST is used, find out the DST convention currently in use in
that country, and adjust DST accordingly.
How it works
The forward and backward adjustment of clocks can be Scheduled or Manual.
• Scheduled DST Adjustment: The Real Time Clock of the ETERNITY is advanced and set backward
automatically according to the DST convention of the country/region where the ETERNITY is installed.
Scheduled DST Adjustment is useful in countries/regions where DST Time is fixed, e.g. Europe, USA and
Canada without yearly variations.
The table below gives describes the DST conventions followed in the different countries for which
ETERNITY will automatically adjust DST.
The DST Type is to be selected according to the country/region where the system is installed.
When DST Mode is set to 'Scheduled' and the DST Type is selected, the system will automatically adjust
DST at the preset dates and time for the country/region where the system is installed.
For example, if ETERNITY is installed Spain, the DST Type 01 applicable to this country should be
programmed as Scheduled DST. The system will automatically advance the clock on the last Sunday of
March at 01.59.03:00 am every year (the start date of DST) and set the clock backward on the last Sunday
of October at 02.59.02:00 am of the same year.
• Manual DST Adjustment: The Real Time Clock of the ETERNITY is advanced and set backward
manually according to the DST convention of the country/region where the ETERNITY is installed.
Manual DST Adjustment is to be used in regions/countries that have no fixed DST Convention and where
yearly variations in DST practices are likely.
When DST Mode is set as 'Manual', you must program the start and the end time, i.e. the time at which the
clock is to be advanced and the time at which the clock is to be delayed.
1. The 'Day of Month' method, which specifies a day of the month DST will start or end. For example:
starting on the 2nd Sunday of March and ending on 1st Sunday of November.
2. The 'Date and Month' method, which specifies a date of the month that DST will start or end. For
example: starting on March 11 and ending on November 4.
DST is not applicable in certain regions/countries, like Asia and South America. In such cases, the DST
Mode is to be 'Disabled'.
How to configure
• Log in as System Engineer.
• If you do not find your region on this list, you are recommended to set DST Mode to Manual and adjust
DST manually.
• Go to the option Forward Time Adjustments to advance the time when DST starts.
• Day-Month Wise to specify the day of the month DST will start.
OR
• Date-Month Wise to specify the date of the month DST will start.
If you select Day-Month Wise option, the Date-Month Wise option will be disabled, and vice versa.
Day-Month Wise
• If you select the Day-Month Wise option, you should now select the desired options in each of the combo
boxes:
• Ordinal number: In the first combo box, select the Ordinal number of the day of the month, i.e. the 1st,
2nd, 3rd, 4th, 5th day, when DST begins.
• Day: In the second combo box, select the day of the month - Sunday, Monday, Tuesday, Wednesday,
Thursday, Friday, Saturday - when DST begins.
• Month: In the third combo box, select the month when DST beings (January-December).
• Change Time From: In this combo box, select the time when DST will begin to change. The time mode
is 24 hours, with options from 00 to 23 hours and 00 to 59 minutes.
• To: In this combo box, select the time to which the DST is advanced. The time mode is 24 hours, with
options from 00 to 23 hours and 00 to 59 minutes.
• Change Time From: The time when DST will begin to change. The time mode is 24 hours, with
options from 00 to 23 hours and 00 to 59 minutes.
• To: The time to which the DST is advanced. The time mode is 24 hours, with options from 00 to 23
hours and 00 to 59 minutes.
• Now, go to the option 'Backward Time Adjustments' to set the time back (i.e. end DST and begin standard
time).
• Follow the same steps described above to set the day/date, month, hours and minutes except, here you
must set these parameters according to the time when DST ends.
• ETERNITY NE gives you the flexibility to set the 'Forward DST Adjustment' according to Date-Month,
while the Backward DST Adjustment according to Day-Month. Similarly, the reverse is also possible,
i.e. Forward DST may be set according to Day-Month, while the Backward DST may be set as Date-
Month. This is flexibility is particularly useful for setting DST of countries where the start of DST is
defined by date and month, e.g. First of April, but the end of DST is defined by Day and Month, e.g. the
last Sunday of October (as observed in Cuba).
• Wherever time adjustments are made at 00:00 hours, use the previous date and set DST start time (i.e.
"from" time) at 23:59 hrs.
What's this?
ETERNITY is supplied with preset values for system and feature settings, as which may be altered and customized
by users to match their requirements and preferences. The factory-set values for system and feature settings that
are automatically assigned by the system are referred to as Default Settings or standard settings.
Every programmable parameter in the system has factory-set default values, which may be changed or customized
by the users to match their requirements and preferences.
How it works
The default settings are to be loaded or restored in the following situations:
So, users must select the appropriate Region for the country/region in which the system is installed.
The system will load the default settings for the country/geographical region where the system is installed.
The system is designed to work efficiently with the default settings. So, if the country/region-specific
default settings match their requirements, users may not even need to alter or customize the values of
various parameters,
They may work with default settings for the most part, customizing only some of the parameters to match
their specific requirements.
The country-specific default settings of various parameters that will be loaded on changing the 'Region' are
presented in the table below. For default values of Trunk Access Codes, Emergency Numbers, Distinctive
Rings, for various countries refer the respective topics.
Default
Default Default Default
Country Country DST Distinctive Abbr.
Time DST CPTG DKP Opr TAC
Code Name Schedule Ring Dialing
Zone Mode Language
Type
001 Afghanistan GMT+04:30 English
002 Algeria GMT+01:00 English
003 Antigua and GMT-04:00 English
Barbuda
004 Argentina GMT-03:00 04 Spanish
005 Australia GMT+08:00 Scheduled 2 05 English 9
(Perth)
006 Australia GMT+09:30 Scheduled 2 05 English 9
(Note2)
(Adelaide)
When there is a system malfunction, possibly caused by a configuration error that you are unable to
diagnose, you may restore default settings.
Whenever you restore the default settings in the system, all the programmable parameters except Network
Port Parameters32 and the Region will be set back to their default values.
You cannot default Region; you can only select a region to load the country-specific default settings and
default the system.
32. The IP Address, Subnet Mask, Primary DNS, Secondary DNS, Host Name, Domain Name, DHCP Server Address.
• The SE password you enter must be the current password. E.g.: if it is 1234, enter 4321 and click OK.
• The names programmed for the Extensions will disappear and the default extension numbers assigned to
the extensions will appear.
Without the SE Password, you cannot restore default values via the software. If you forget the SE
Password, you must resort to hardware default of the SE-Programming Password first.
What's this?
Department Call enables you to group together extensions of a particular department so that callers can reach
anyone in the department by dialing a common access code assigned to the department.
Calls made to such groups of extensions are called Department Calls and the access code used to make
department calls is called Department Number.
This feature is useful in situations where any member of a department may interact with callers, as for instance in a
information counter, a customer care cell, a technical support team, etc.
Callers can also reach individual extensions in a Department group by dialing the extension number.
ETERNITY NE supports the formation of 5 Department groups. The member extensions of a department group
may be single line telephones (SLT), digital key phones (DKP) and SIP extensions.
Each Department Group can also be assigned a mailbox for voice mail, which any member extension can access.
Each Department Group can forward its calls to an extension or to its voice mail, or to another Department Group.
How it works
Extensions A, B, C, D are grouped as a Department with the access code 391.
Internal Calls
• Extension E dials 391 to call the Department.
• The system checks E's Class of Service for the Department Call feature.
• The feature is enabled. The system checks if Rotation is enabled for the extensions in the Department.
• If the Rotation flag is enabled. The system lands the call on the extension which is set to ring first.
• Extension A, configured as the first landing destination rings for the duration of the Ring Timer
(configurable; default: 15 seconds).
• A answers the call. Speech established between A and E.
• If A does not answer, the system hunts for the next extension in the group to land the call, B.
• B starts ringing for the duration of the Ring Timer.
• If Continuous Ring is enabled on A, A will continue to ring even as B is ringing.
• If B does not answer the call at the end of the timer, the system hunts for the next extension, C.
• If B has Continuous Ring enabled, B will continue to ring even as C is ringing.
• If the call is not answered even after hunting the last extension, the system will loop back and start from
the first extension once again.
External Calls
Department Calls can be made using Direct Inward Dialing (DID). For example, a company may use DID to have
callers who want information only to dial the Information Department instead of waiting for the Operator.
Thus for each call, the system will hunt for a landing extension as per the Rotation set for the deparment extension
group. The extensions will ring for the duration of the Ring Timer, either continuously or one-by-one (as per the
Continuous Flag configured), and according to the sequence in which the extensions in the group are arranged.
Rotation ensures equal distribution of call traffic. If Rotation is disabled, the fresh call will always land on first
extension of the Department group.
Voice Mail
If your system has the Voice mail module, a Department Group can be assigned a common mailbox for Voice mail,
called the Department Group Mailbox. You can assign Department Group Mailbox to selected extensions or to all
extensions in the Department.
To take the example of Extensions A, B, C, D with the Department Access Code 391 further,
• Extensions A, B, C and D are all members of Department Group 1 with the Access Code 391.
• When there is a new message in the Group Mailbox, all four extensions - A, B, C, D - will get the Message
Wait Notification.
• The message wait notification may be a Stuttered Dial Tone or a Voice Message when the extension user
goes OFF-Hook, or blinking of the LED Lamp on the extension, or a Ring.33
• To the first extension that answers the notification call, for example, Extension A, the Voice mail System
informs about the new message(s) waiting in the Department Group Mailbox and in the Personal Mailbox.
"You have <x> new Message in your Personal Mail Box. You have <y> new Messages in your Department
Group Mail Box."
• If there is no new message in both mailboxes, the VMS will play the message: "You have Zero new
Message."
• If there is a new message in the Department Group Mailbox, but none in the Personal Mailbox, the
VMS will play the message:"You have <x> new Message in your Department Group Mailbox."
• If there is no new message in the Department Group Mailbox, but new message in the Personal
Mailbox, the VMS will play the message: "You have <x> new Message in your Personal Mailbox."
33. This will depend on the type of Message Wait Notification configured for the extension in its Voice mail Settings.
• The user of Extension A presses 2 and is taken to the Department Group Mailbox. The VMS plays the
following prompts:
• “To listen to New Messages, press 1.”
• “To listen to old messages, press 2.”
• “To send a message, press 3.”
• “To change your mailbox name, press 4.”
• “For message redirection, press 5.”
• “To go to Home Position, press 0.”
• Extension A presses 1.
• VMS plays the new messages.
• After playing the new messages, the VMS cancels Message Wait Notification set for extensions B, C and
D.
Call Forward
Just as calls can be forwarded to a Department Group, a Department Group can also forward its calls to:
• an extension.
• its own Department Group Mailbox.
• another Department Group.
ETERNITY supports the following Call Forward options for Department Groups:
• Call Forward - unconditionally: calls are forwarded to the destination number, without checking the
status or waiting for a response from the Department Group.
• Call Forward- if Busy: calls are placed on the Department Group as per the Rotation configured for it and
are forwarded to the set destination, only when all the member extensions of the Department Group are
found to be busy.
• Call Forward- if No Reply: when a call is made to the Department group, ETERNITY NE will place the
call as per the Rotation configured for the Department Group for the duration of the 'Call Forward No Reply
Timer for Department’ (30 seconds). If none of the member extensions answers the call before the expiry
of this timer, the call is forwarded to the destination.
• Call Forward - if Busy/No Reply: calls made to the Department Group will be routed to the destination, if
all members of the Department Group are busy or when none of the the member extensions answered the
call within the Call Forward No Reply Timer for the Department.
For Department Groups, ETERNITY NE does not support Call Forward to an external destination number.
• Call Forward to an external destination number, and Call Forward - Dual Ring are not supported for
Department Groups.
• Member extensions of a Department Group can set Call Forward on their extensions. However, Call
Forward set for the Department Group will have precedence over Call Forward set by individual
member extensions.
• Extensions A, B, C, D of Department Group 1 are allowed Call Forward Department Group in their Class of
Service.
• Any of them can set Call Forward for Department Group 1. Extensions A, B, C and D can also set Call
Forward on their own extensions.
• When any extension or an external caller (also using Direct Inward Dialing or Direct Inward System
Access) dials the Access Code 391 to call Department Group1, ETERNITY NE will check the Call Forward
option set for the Department Group and route the call accordingly.
• If Call Forward - unconditionally is set, the call will be routed to the destination number, regardless of Call
Forward set by any of the member extensions.
• If Call Forward - Busy is set, and the first extension in the Department Group is busy, the system will hunt
for the next free extension in the group. It will continue to hunt for a free extension. If all extensions in the
group are busy, the call will be forwarded to the destination number.
Call Forward unconditional, busy, or busy/No reply set by any member extension will not work.
• If Call Forward - No Reply is set, the system will start the Call Forward No Reply Timer Department Group
and place the call as per the Rotation set for the Department Group. If the call is not answered by any of
the extensions before the timer expires, the call will be forwarded to the destination number.
If a member extension that is offered the call has set Call Forward-Unconditional, and the Call Forward No
Reply Timer Department Group has not exipred, the call forward set by the extension will be applied. If the
timer expires, the Call Forward No Reply set for the Department Group will be applied.
If a member extension that is offered the call has set Call Forward-No Reply, or No-Reply/Busy, the Call
Forward No Reply Timer (for individual extension) will start simultaneously with the Call Forward No-Reply
Timer Department Group. If the No Reply Timer for the extension expires first, the call will be forwarded to
the destination set for the extension. If the No Reply Timer of the Department Group expires first, before
the call is answered, the call will be forwarded to the destination set for the Department Group.
How to configure
The functioning of this feature requires you to do the following:
• create Deparment groups.
• select member extensions for the Department groups
• assign appropriate access codes to the Department groups.
• enable 'Department Call' in the Class of Service extensions that are to be allowed to make Department
calls.
If you want to provide voice mail facility to the Department Group, you must
• assign a Mailbox to the Deparment Group.
• allow member extensions access to the Department Group Mailbox.
If you want to enable Call Forward to the Department Group, you must
• enable 'Department Group Call Forward' in the Class of Service (CoS) of the member extensions.
• Decide the number of department groups you want to create, for instance: 4 groups.
• Group all the extensions you want to put in each department group. You cannot group more than 16
extensions in a single department group.
• Decide in what sequence the extensions in each group should ring, i.e. which extensions should ring first,
second, third, and so forth.
• Decide the access code you want to assign to each department group.
The access codes for the department groups and extensions in this table are default access codes.
• Now, with this information ready, you may configure the department groups using Jeeves.
By default, ‘Department Group Call Forward’ is enabled for day and night time in the COS of all extensions of
ETERNITY NE. So, all extensions of ETERNITY can set Call Forward Department Group.
If you wish to allow this feature to member extensions only, retain this feature in the COS group of member
extensions only, and disable this feature in the COS group of all other extensions. For instructions see “SLT
Extensions”, “DKP Extensions”, “SIP Extensions” under Basic Settings. See the topic “Class of Service (COS)”.
• Assign an Access Code to the department group against the Department Group Number.
By default, the Access Codes assigned to Department groups are from 391 to 395.
If you decide not to use the default access codes, make sure that the access code you assign to each
department group is unique, and does not match with any SLT, DKP, SIP, Door Phone, DOP access code
or any feature access code of the Dial Phase. Refer the topic “Access Codes” to know more.
A multiple selection box opens. Select the extensions you want to include in this Department group from
the list box on the left. Configure Ring Time, Rotation, as desired.
• Configure Ring Timer(s). This timer defines the time for which the extension, on which the call lands,
should ring. Default: 15 seconds.
• Select the check box Continuous Ring, if you want the extension to ring continuously until the call is
answered. The first extension will continue to ring even as the system hunts for other extensions in the
routing group to land the call. If the call still remains unanswered, the system will return the call to the first
extension once again. This flag is of no relevance, if you have selected only one member extension in the
department group.
• Select the check box Rotation, if you want to enable rotation of calls in the department group having
multiple member extensions. When enabled, each fresh call will land on the extension which is next to the
one that received the last call. This ensures equal distribution of incoming calls to all the destinations within
the group.
Rotation has no relevance if the department group has only one member extension.
Example:
The Customer Care Department of a company has four extensions: 201, 202, 301 and 302, which needs to be
grouped for Department Calls.
Extensions 201 and 202 are SLTs. Extensions 301 and 302 are DKPs.
1. Enable the 'Rotation Flag' on routing group number 01 to distribute call traffic.
2. Enable the 'Continuous Ring Flag' for member 01 (201) and set the 'Ring Timer' to '20 seconds.
3. Set the Ring Timer of member 02 (202) to 10 seconds. Disable 'Continuous Ring' flag.
4. Retain the Ring Timer of member 03 (301) as default 15 seconds. Disable 'Continuous Ring' flag.
• Mailbox: Select this checkbox to assign a mailbox to a Department Group. By default, no mailbox is
provided.
• Mailbox Size (min): Define the size of the mailbox to any desired value from 001 to 999 minutes. The
default Mailbox Size is 5 minutes.
• Maximum Message Length (sec): Define the length of each message (in seconds) callers are to be
allowed to record in the Department Group mailbox. Change the maximum message length to any
desired value between 001 to 999 seconds.
• New Message Delivery option in Mailbox Full condition: When the Department Group mailbox is
full, you may select one of the following options for delivery of new messages:
• Do not offer to leave message: The VMS will not allow the caller to record a message by declining
delivery of the message.
• Deliver to General mailbox: Select this option if you want the VMS to record the message of the
callers in the General mailbox. The General mailbox is a shared mailbox between extension users.
Only extension users who have General Mailbox in their “Class of Service (COS)” are allowed to
access it.
When you select this option, make sure that General Mailbox is enabled in the Class of Service of
the member extensions of the Department Groups. Refer “Class of Service (COS)” for instructions.
• Overwrite old messages: Select this option if you want the VMS to overwrite the existing
messages to record the new messages. The VMS starts overwriting the oldest message message
first.
• Play message details after delivery of message: After the extension user has listened to a message
in the mailbox, you can have the VMS can play message details such as Date and Time when the
message was recorded, the caller’s number34, and the extension number dialed by the caller35 to the
extension user.
• Always: The VMS will play message details to the maibox owner after playing each message.
• On Demand:The VMS will play message details to the mailbox owner only when the mailbox owner
requests it. On completion of each message, the VMS will prompt the extension user to press a digit
34. The number of person who left the message in the mailbox.
35. The number of the extension user for whom the message is intended.
• Voice Mail Auto Attendant Features: This parameter is applicable only if you are using the VMS Auto
Attendant for “Direct Inward Dialing (DID)”.
• Voice Mail Auto Attendant Profile: Select a profile for the Department Group. The Auto Attendant
profile determines the greetings and welcome message to be played to mailbox owners when they
reach the home node. It also determines whether or not the user should be taken to the root node
directly.
• Abbreviated Name: When the VMS is used as Auto Attendant for Direct Inward Dialing, the callers
can be prompted to Dial by Name of the desired party instead of the extension number.
To allow callers to reach the Department group using Dial By Name, abbreviate the Department
Group’s name to the first three letters and enter it in this field.
• Announce Name: If you want the VMS to announce the Department group’s name when
transferring the call to the extension, select the checkbox to enable Announce Name. By default,
Announce Name is disabled.
If you enable Announce Name, make sure you record the Department group’s name on the VMS.
See “Recording Station Names” for instructions.
• Call Transfer Type: Select the desired method for transferring the call answered by the VMS Auto
Attendant to the extension. You may select any of the following methods of call transfer for the Day
(working hours) and the Night (non-working hours):
• Transfer to Mailbox: When the caller dials the extension number, the VMS Auto Attendant will
check if the extension number has a mailbox assigned and transfer the call to the mailbox of the
extension.
• Transfer immediately: When the caller dials the extension number, the VMS Auto Attendant
will transfer the call on the extension without checking its status.
• Transfer when Extension rings: When the caller dials the extension number, the VMS Auto
Attendant will wait for the extension to start ringing and then transfer the call.
• Transfer when Extension answers: When the caller dials the extension number, the VMS
Auto Attendant will transfer the call when the extension answers (goes OFF-Hook).
• Transfer when Extension permits: The VMS Auto Attendant prompts the caller to record his/
her name, puts the caller on hold and places the call on the desired extension. If the extension is
free and answers the call, the VMS announces the caller’s name to the extension user and
prompts the extension user to choose whether or not to speak to the caller. If the extension user
chooses to talk, the VMS transfers the call. If the extension user chooses not to talk, the VMS
transfers the call to the mailbox of the extension user and asks the caller to leave a message.
By default, Wait for Answer is selected as Call Transfer method for all time zones.
• From the combo box Specify Department Group, select the access code of the Department Group
whose group mailbox you want to allow access to by the extension.
How to use
OR
What's this?
Dial By Name enables extension users to call another extension or an external party by dialing the name of the
person, instead of dialing their telephone number.
This feature is accessible only to extension users of the proprietary digital key phone, EON and the Extended IP
phone.
With Dial By Name, extension users need not remember the desired party's telephone number or short codes, i.e.
“Abbreviated Dialing” codes.
For each extension, the database for names used in Dial by Name is drawn from:
• the Personal Directory, which is assigned to each extension, wherein up to 25 external party numbers
along with their names may stored. The system uses the Personal Directory to dial external parties by their
names. See “Abbreviated Dialing” to know more.
• Global Directory, which is assigned to the extension in its “Class of Service (COS)”. The Global Directory
is a system-wide list of external party numbers and names. Upto 999 numbers can be stored in this
directory, and parts of the Global Directory (Part 1, 2, 3) can be assigned to each extension in its Class of
Service. See “Abbreviated Dialing” to know more.
• Names of Extensions, which are names of users. Their names are assigned to SLT, DKP and SIP
extensions to identify the extension users. Names of Extensions are necessary for making internal calls
using the Dial By Name feature.
How it works
• Press the 'Names' key.
• The prompt <Name : > appears on the phone display.
• Enter the name of the desired party37.
• For example, you wants to call Midas Biz, enter the letter 'M' using the keypad.
• The system displays in alphabetical order, all names starting with 'M'. These numbers are drawn from the
Personal and Global Directories assigned to your extension and the Extension Names programmed in the
system.
• Scroll the list using the Up/Down navigation keys to reach the desired contact's name.
OR
Instead of scrolling the entire list, you can enter more than one initial letter of the contact's name. The
search is narrowed down to more accurate matches. The phone displays the matching entries in the
directory.
37. The process of entering the names is the same as when writing text messages (SMS) from a cell phone. The keys must be
pressed multiple times in quick succession to enter the desired alphabet.
• DSS Key: A direct station selection (DSS) key must be programmed for the Dial by Name feature. Without
the DSS Key this feature will not be accessible.
Both EON48 and SETU VP248, the Extended IP Phone, have the DSS Key labeled as 'Names'.
• Global Directory: The names of the external parties must be programmed against their respective
telephone numbers in the directory. Refer the topic “Abbreviated Dialing” for instructions on programming
the Global Directories.
• Personal Directory: The names of the external parties must be programmed against their respective
telephone numbers in the Personal Directory. Refer the topic “Abbreviated Dialing” for instructions on
programming the Personal Directories.
• Extension Names: Extensions may be SLTs, DKPs, SIP extensions. Refer “Extension Numbering Plan”
under Basic Settings for instructions on assigning names to extensions.
• Class of Service: Dial By Name is allowed to all DKP and Extended IP Phone users. However, the use of
this feature is related to the following features, which must be enabled in the Class of Service of the DKP
and SIP extension users:
• 'Internal Calls'- This is to be enabled so that the extension can call other extensions.
• Global Directory Part 1
• Global Directory Part 2
• Global Directory Part 3.
If you want the names to be drawn from Global Directory Part 2 and Part 3, provided these are
programmed, you must enable these two directories in the COS of the DKP and SIP extensions. See
“Class of Service (COS)” for instructions.
The system will display the names exactly as they have been programmed in the Personal and Global
Directories.
How to use
38. The number of matching entries that will appear at a time on your phone's display will vary according to your phone's LCD display
capacity.
• Go ON-Hook.
• Go OFF-Hook.
• Press the 'Names' key again.
• Enter the name/initial letters of the contact's name.
What's this?
ETERNITY NE provides Dial Plan to support On-hook Dialing on the Matrix Extended IP Phones.
Extended IP Phone users can make calls and access features, without going Off-hook from the handset, headset
or speaker.
How it works
When you are using the Matrix Extended IP Phone, you must configure the Dial Plan. During auto-configuration of
the Extended IP Phone, the ETERNITY NE assigns the Dial Plan to the phone.
The Dial Plan table stores up to 100 entries, which are stored against Index 001 to 100. Index001 is reserved for
'No Match Found'.
For each number string you configure in the Dial Plan, you must also define the digit length as Maximum Digits.
The Maximum Digits defines the number of digits to be dialed by the extension user for the phone to consider it as
end-of-dialing.
When the Extended IP Phone user dials digits, the phone compares the dialed number string with the entries in the
Dial Plan using the best match found logic.
When a match is found for the number, the phone waits for the duration of the Inter Digit Wait Timer to receive the
maximum digits defined for this number. When the maximum digits are received39, the phone considers it as end-
of-dialing and forwards the digits to the ETERNITY NE.
002 1#91 04
003 1#92 04
004 0 01
005 1 02
006 10 03
007 11 03
008 16 03
39. The phone will not wait for the Inter Digit Wait Timer to expire.
009 105 04
010 107 04
011 109 04
012 117 04
013 2 04
014 3 04
015 38 02
016 4 01
017 5 01
018 6 02
019 7 01
020 8 01
021 9 01
022 * 01
023 24
:: 24
:: 24
100 24
The Dial Plan is not updated automatically when Access Codes or Extension numbers are changed. You must
update the Dial Plan if you change the Access Codes or Extension Numbers.
How to configure
• Open Jeeves.
• In the Number field, enter Access Codes in the Dial state and extension numbers. A maximum of 24
digits can be entered.
• In the Maximum Digits field, define the maximum number of digits that the phone should wait to
receive before considering it as end-of-dialing and forward the dialed string to the ETERNITY NE. A
maximum of 24 digits can be entered in this field.
What's this?
Dialed Number Directory is a Digital Key Phone/Extended IP Phone feature.
It is the list of numbers dialed out from the DKP/Extended IP Phone, similar to the call history of recently dialed calls
on a cell phone.
These numbers may have been dialed out using features like Abbreviated Dialing, Quick Dial, Redial, Walk-In
Class of Service, or may be a simple outgoing call made by directly dialing the external number.
How it works
• When a DKP/Extended IP Phone extension user makes an outgoing external call, the number is stored in
the Redial Number List.
• The list is updated using the First-In First-Out logic, whereby the earliest dialed number is replaced with
the most recently dialed number.
• To use this feature, the DKP/Extended IP Phone user must invoke the “Last Number Redial” feature.
• Doing so, the Redial Number List will appear on the phone display.
• The user may now navigate the list, select the number to be dialed out.
• The system will dial out the selected number using the same Outgoing Trunk used to place this call earlier.
• If the number had been dialed earlier using Abbreviated Dialing, the system will check for Toll Control
when dialing out the number again from the dialed number directory40.
How to configure
No specific programming required.
How to use
40. Recall that the system does not check for Toll Control when Abbreviated Dialing is used.
What's this?
Digest Authentication is a challenge-based authentication service of SIP to authenticate the identity of the
originator of SIP request in the INVITE message. The recipient of the request can ascertain whether or not the
originator of the request is authorised to make the request. When the digest credentials of the originator—User
Name and Password—in the INVITE message are authenticated and accepted by the recipient, the originator and
the recipient are connected.
How it works
The Digest Authentication feature works on the basis of the Digest Authentication Table, in which the credentials,
namely the User Name and Passwords of trusted/authorised calling party SIP devices are stored. You must
configure this table. The Digest Authentication Table is common for all SIP trunks on which this feature is enabled.
When you enable this feature on a SIP trunk, for all incoming calls (SIP requests),
• ETERNITY NE will challenge the identity of the calling party, i.e. the SIP device initiating the request to
send its digest credentials.
• When the calling party sends its credentials, ETERNITY NE authenticates the credentials by matching it
with its Digest Authentication Table.
• If a match is found, the calling party will be authenticated and the call will be allowed on the SIP trunk.
• If no match is found, ETERNITY NE will consider it as invalid authentication information and reject the call.
How to configure
To use this feature,
• make a list of devices whose incoming calls (SIP requests) you want to allow after authentication.
• enable Digest Authentication on the desired SIP Trunk
• configure the Digest Authentication Table.
• Open Jeeves.
• In the User ID field, enter the User ID to be authenticated. The User ID must be within 40 characters.
• In the Password field, enter the corresponding Password. The Password must be within 16 characters.
DKP Features
• Status of other ports (Tri-colour LED indication)
• Programmable Direct Station Selection (DSS) Keys and Feature keys
• LCD notification messages
• Ringer Tune selection
• Adjustable Speech level
• Adjustable Ringer Volume
• Adjustable Backlight and Contrast levels
• Hands-free operation - Speaker key and headset connectivity.
• Call Logs - last 20 Missed, Answered and Dialed Calls.
• Message Paging
• Menu based operation of PBX features
• Multiple Language support.
PBX Features
Listed below are the features of ETERNITY that require a Digital Key Phone:
• Abbreviated Dialing
• Auto Answer
• Background Music
• Call Chaining
• Call Cost Display
• Call Duration Display
• Call Mute
• Dialed Number Directory
• Directory Dialing by Name
• Dynamic Lock
• Forced Answer
• Keypad Lock
• Live Call Screening
• Message Paging
• Off-Hook Alert
• Room Monitor
• Text Message Reply
• Time Zone Display
• User Status (Presence)
Model
Feature
EON48S EON48P EON48DS EON48DP
Total number of keys 48 48 48 48
Number of programmable keys 29 29 29 29
Capsense keys Yes Yes Yes Yes
LCD display capacity 2 lines x 24 6 lines x 24 2 lines x 24 6 lines x 24
characters characters characters characters
Touch Keys No No No No
Touch screen operation No No No No
LCD with backlight Yes Yes Yes Yes
Headset Interface Yes Yes Yes Yes
Ringer Lamp (LED) Yes Yes Yes Yes
Speaker Phone Full duplex Full duplex Full duplex Full duplex
EON48
EON48S/EON48D-S
2 lines and 24 characters LCD display, full duplex, capsense feature keys
6 lines and 24 characters LCD display, full duplex, capsense feature keys.
LCD Display
The LCD display of EON48P/48D-P/48S/48D-S is backlit and can be tilted at a convenient angle for a clear view of
the text/characters displayed.
The LCD backlight can be turned on and off as well as adjusted for contrast and brightness from the "Phone
Settings" of the DKP Phone Menu.
Ringer LED
The Ringer LED indicates incoming internal and external calls. The LED Cadence will match with the Ring
Cadence of the incoming internal/external call.
The Ringer LED changes colour according to the type of call, as described in the table below.
Priority Red
• Enter Key: To enter the Menu; when the phone is in the idle state (without any incoming or outgoing call
being made), if you tap the 'Enter' key, you will enter into the 'Menu'.
Enter key is also used to make a selection from the Menu/sub-menu options or to complete an action.
• Back Key: To move backwards when dialing a number; to go back one level in the Menu.
Refer the topic “DSS Keys” for instructions on programming these keys.
• Status of Extensions and Trunks: The LED of DSS keys assigned to Extensions/Trunks glow in three
colours to indicate status of the call event on the Extensions/Trunks and on the DKP.
Thus, the status of the DKP user's own Extension as well as that of the other Extensions (i.e. DKPs and
SLTs) and the status of Trunk lines are indicated by the LED of the DSS keys assigned to those
Extensions and Trunks on the DKP.
The following table shows the relationship between the colour of the LED and various events:
Blue The key assigned to the The key assigned to the The key assigned to the
Extension you are in speech Extension you have kept on Extension you are calling or
with. hold. from which you are being
called.
Red The key assigned to the The key assigned to the The key assigned to the
Extension that is now busy Extension which has put Extension/Trunk that is called
with another Extension/ another Extension/Trunk on or being called by another.
Trunk. hold.
Violet You are talking on a Trunk You have held a Trunk You have an incoming call on
(external call) (external call) the Trunk (external call)
• Red indicates the state of other Extensions/Trunks. For example, if the LED of the DSS key assigned
to Extension 201 is glowing Red continuously, it means Extension 201 is busy with another Extension
or Trunk.
• Violet indicates the state of the trunk you are in speech with. For example, when you are in speech on
an outgoing call on Trunk 1 the LED of the DSS Key assigned to Trunk 1 will be continuously ON.
When you put the call on hold, the LED will blink slowly.
The LEDs of DSS Keys that are designated as Call Appearance (CA) Keys will function as follows:
Blue When you are in speech with When you have put an When any Extension is calling
an extension (internal call) extension on hold (internal (internal call)
call)
Violet When you are in speech with When you have put a Trunk When any Trunk is calling
Trunk (external call) on hold (external call) (external call)
• Status of Features: The LED of a DSS key is activated when the feature assigned to this key is used.
The LED of DSS keys assigned to Extensions/Trunks glow in a single colour - Red - to indicate status
of the call event on the Extensions/Trunks and on the DKP.
• Not all features require LED indication. Hence the LED on a DSS Key is activated only if the feature
assigned to that key requires LED.
• For example, Call Pick-Up; this feature does not require an LED. So when a DSS key is assigned to
this feature, the LED of the key remains inactive, when Call Pick-Up is accessed.
• A feature like Auto Redial requires an LED to show that it has been set or canceled. So, the LED of the
DSS key to which the Auto Redial feature has been assigned will glow Red, when Auto-Redial is set,
and the LED is turned off when the feature is canceled.
• Thus the LEDs of the DSS keys function only if the LED is relevant for the feature/ function assigned to
the keys, and otherwise remain inactive for example, Raid, Interrupt Request, Barge-In, Last Caller
Recall.
Dial Pad
The dial pad consists of 12 fixed keys for the digits 0, 1-9, and the characters * and #. The dial pad is used for
dialing numbers of extensions, external parties, and for dialing the programming and feature access codes.
Speaker Key
The speaker key sets the phone in 'Speaker mode' for hands-free operation. The Speaker key is programmable,
you can program any other feature/function on this key.
Since key is programmable, the LED indication pattern will be according to the feature/function you assign to this
key. For example, if you assign an extension to this key, the LED of the key will function as a tri-colour LED to show
status of the Extension. If you assign a feature that does not require any LED activity, like Call Pick-Up, the LED of
this key will remain inactive.
Volume Keys
• "+" (plus): This is the increase key, to raise the volume of speech while talking and to decrease the Ringer
volume, when the phone is ringing.
• "-" (minus): This is the decrease key, to lower the volume of speech while talking and to decrease the
Ringer volume when the phone is ringing.
Headset Connectivity
The EON48P/48D-P/48S/48D-S provides two Headset interfaces: a 2.5mm Audio Jack and an RJ11 connector at
the bottom of the phone body.
So you can use any stereo headset of standard make with a 2.5 mm single connector or a stereo headset with an
RJ11 connector.
You can also program any of the DSS keys to function as the Headset key. Refer the topic “DSS Keys” for
instructions.
Key Maps
As EON48P/48D-P/48S/48D-S may be the extension of the Operator(s) and Executives in an enterprise to meet
the varied requirements of each user group, these key maps can be customized to match the exact requirement of
individual users. Refer the topic “DSS Keys” for instructions on customizing the Key Maps.
Phone Menu
You can access the following PBX and phone features from the Menu of EON48P/48D-P/48S/48D-S:
Call Logs To view call history of internal and external Missed, Answered and Dialed calls.
You can also edit numbers in the call logs and store them in the Personal Directory.
Call Forward To set and cancel Call Forward-Busy, Call-Forward No Reply, Call-Forward-
Unconditional, and Follow Me.
Do Not Disturb To set/cancel Do Not Disturb on the phone, i.e. block incoming internal and external
calls.
Call Cost Display To view the cost of calls made from the phone.
Change User To change User Password (required for using certain features like Call Follow Me,
Password Dynamic Lock, DISA, Walk-In Class of Service, User Absent/Present, Hot Desk) and for
customizing Phone Settings.
Phone Settings To customize settings of the phone such as Speech and Ringer Controls, LCD Display
settings (Brightness and Contrast, Backlight ON/OFF), Headset Connectivity, Call
Answering Mode (manual/auto answer).
To exit menu,
Operating EON48
Please refer the User Card for EON48 for instructions on operating the features of ETERNITY using EON.
EONSOFT
The EONSOFT is a PC-based Digital Key Phone. Based on a graphic user Interface (GUI), the EONSOFT offers all
the features of EON48, making it a substitute for the Digital Key Phone. Its integration with the ETERNITY obviates
the need for a separate telephone instrument.
The EONSOFT can be installed on any personal computer with Windows or NT operating system.
Two PC-based DSS64 Consoles are available to be used with the EONSOFT. You can use either one or both
DSS64 Consoles.
DKP Port
The DKP port connects EONSOFT to the DKP port of ETERNITY's DKP card.
Handset Port:
The Handset port connects the Receiver of the phone, to be used for speech. The EONSOFT has the provision for
attaching a Handset. A handset with spring cord is supplied by Matrix and is to be connected to the handset jack
(RJ12) on the Dongle.
Headset connectivity
EONSOFT supports headset connectivity, providing a MIC and a Speaker interface. Any stereo Headset of
standard make, with dual connectors can be connected to the MIC and the Speaker on the Dongle.
The headset connection allows speech in hands-free mode. The headset with either RJ11 or CASIO jack both are
supported by EONSOFT.
COM Port
The COM port connects EONSOFT to a computer (COM Port).
After EONSOFT has been successfully installed on a computer and the DKP parameters have been configured,
each time you open EONSOFT, the display and keypad of the phone will appear on your computer screen.
Phone Display
The EONSOFT has a 2-line and 24-character display. In the ON-Hook or idle condition, the first line displays the
Extension Number and the Extension name. The second line displays the Day, Date and Time.
When there is an incoming call, the calling party's number is displayed on Line 2 of the LCD41.
The LCD messages for various call events (dial, transfer, forward, hold, etc.), for prompts, alerts, confirmation,
errors, text messages, are displayed.
Refer the topic “DSS Keys” for instructions on assigning extensions, trunks, features to keys.
41. Only if the Extension, to which EONSOFT is connected, has been allowed CLIP facility in its Class of Service.
Green The key assigned to the The key assigned to the The key assigned to the
Extension you are in Extension you have kept on Extension you are calling or from
speech with. hold. which you are being called.
Red The key assigned to the The key assigned to the The key assigned to the
Extension that is now Extension which has put Extension/Trunk that is called or
busy with another another Extension/Trunk on being called by another.
Extension/Trunk. hold.
Orange You are talking on a You have held a Trunk You have an incoming call on the
Trunk (external call) (external call) Trunk (external call)
• Green indicates the state of the extension/trunk you access. For example, when you make a call to
another extension 203, the LED of the DSS key assigned to Extension 203 blinks Green to indicate
ringing at the extension. If you have successfully established speech with Extension 203 the LED glows
Green continuously.
• Red indicates the state of other Extensions/Trunks. For example, if the LED of the DSS key assigned
to Extension 201 is glowing Red continuously, it means Extension 201 is busy with another Extension
or Trunk.
• Orange indicates the state of the trunk you are in speech with. For example, when you are in speech
on an outgoing call on Trunk 1 the LED of the DSS Key assigned to Trunk 1 will be continuously ON.
When you put the call on hold, the LED will blink slowly.
The LEDs of DSS Keys that are designated as Call Appearance (CA) Keys will function as follows:
Green When you are in speech When you have put a When any Extension is
with a Extension (internal Extension on hold (internal calling (internal call)
call) call)
Orange When you are in speech When you have put a Trunk When any Trunk is calling
with Trunk (external call) on hold (external call) (external call)
• Status of Features: The LED of a DSS key is activated when the feature assigned to this key is used.
The LED of DSS keys assigned to Extensions/Trunks glow in a single colour - Red - to indicate status
of the call event on the Extensions/Trunks and on the DKP.
• Not all features require LED indication. Hence the LED on a DSS Key is activated only if the feature
assigned to that key requires LED.
• For example, Call Pickup; this feature does not require an LED. So when a DSS key is assigned to this
feature, the LED of the key remains inactive, when Call Pickup feature is used.
• Thus the LEDs of the DSS keys function only if the LED is relevant for the feature/ function assigned to
the keys, and otherwise remain inactive.
• The LEDs of DSS keys to which features like Raid, Interrupt Request, Barge-In, Last Caller Recall are
assigned, will not glow.
Dial Pad
The dial pad consists of 12 keys (non-programmable), which include the digit keys for 0, 1-9, and character keys for
* and #.
Function Keys
These are non-programmable keys on the keypad of EONSOFT which have fixed functions.
• Redial: This key is used for redialing the last external number.
• Func Key: This key is used for accessing the Phone menu.
• Adr: This key is used for accessing the Address Book. The EONSOFT provides the facility of an Address
Book that is integrated with the Standard Windows Address Book, for storing the numbers and addresses
of contacts. So, when a call is to be made, you can select and dial the desired number from the directory.
• Hold: This key is used for putting the caller on hold. This key is also used to make a selection in the Phone
Menu.
• OFF-Hook: This key is used for going OFF-Hook. It simulates lifting of the handset, pressing of the
speaker key to make or receive calls.
• ON-Hook: This key is used for going ON-Hook. It simulates replacing of the handset, pressing of the
speaker key to disconnect.
Navigation Keys
The following keys are used for navigating the phone menu:
• 'Func' key: This key is used for entering the Phone menu and to go back one level in the menu.
• Up and Down keys: The and keys function as the Up and Down keys to scroll the Menu and sub-
menu options. You can scroll up down the menu by clicking on and scroll up the menu by clicking on .
Speaker key
The 'Spk' key sets the phone in 'Speaker mode' for hands-free operation. The Speaker key is programmable; you
can program any other feature/function on this key.
Shortcut keys
You can use the Keyboard of the computer to operate EONSOFT, with the help of "shortcut keys'. The following
table describes the functions performed when shortcut keys on the keyboard are pressed:
F1 Help
F3 Spd
F4 Func
F6 Alt+Enter - Hold
F7 Xfr
F8 Spk
Esc ON-Hook
. (dot/period) Flash
(Up Arrow Key) Volume key. To increase volume of ringer and speech
(Down Arrow Key) Volume key. To decrease volume of ringer and speech
Tab (Tab
Backward shifting of selection
Backward)
Key Maps
EONSOFT can function as an extension for the Operator, Executive, and Hotel Attendant, also Guest (though
unlikely to be used by guests).
Phone Menu
The menu is the same as EON48.
To exit menu,
• Press the 'Func' key repeatedly to go back one level in the menu, till you reach 'Menu'.
Or
If you want to use the Keyboard, press the Shortcut key for the desired function.
Tool Tips
You can program labels and tool tips for the DSS keys, which are displayed to the user on mouse over. You can
program the function of each key as Tool Tip, to help user in intuitive operation of EONSOFT.
Call Indication
Incoming Calls are indicated by:
In order for the EONSOFT window to pop up, you must have enabled the 'PopUp When Ring' option. When
this option is enabled and the EONSOFT window is minimized a new incoming call causes the window to pop
up to its full size notifying the user about the new call. When this option is enabled and the EONSOFT window
is maximized, a new incoming call is indicated by the flashing of the Title bar of the window.
When the 'PopUp When Ring' option is disabled and the EONSOFT window is minimized, a new incoming call
is indicated by the flashing of the EONSOFT Title at the bottom bar of the computer screen.
To answer the second incoming call, you may put the current call on hold.
Operating EONSOFT
EONSOFT can be operated using the keyboard and the mouse.
Making calls
To make calls,
• A headset must be connected and 'Headset Connected?' must be set to Yes in the Phone Settings of
the DKP.
• If you are using the keyboard instead of the mouse, press the appropriate Shortcut Keys listed above
and use the Number pad on the keyboard to dial digits.
Receiving calls
• When window pops up to indicate a call,
• Click Spk key or the OFF-Hook Key.
• Talk.
• Click ON-Hook key to disconnect.
What's this?
The ETERNITY NE provides a Digital Output Port (DOP) for connecting a Door Lock release relay to be used in
conjunction with the Door Phone connected to the system.
The Door Lock connected to the DOP can be opened by dialing the related Feature Command from the System
Administrator mode.
ETERNITY NE remembers the state of DOP the during power failure. If a power failure occurs while the
Door Lock device is being operated, the system will remember the last state (in this case, ON) and switch
ON the DOP when power is restored.
Do not connect devices that do not conform to the specifications of the DOP!
How to configure
When you connect a Door Lock to the DOP, configure DOP.
• Click the Advanced Settings link, scroll to System Parameters and click the link.
• Set the Normal Contact Type for the DOP to Normally Open.
• Click Submit.
• Make sure the option Use Door Lock is enabled in the related set of “Door Phone” parameters in “Basic
Settings”.
• You may set the time for which the Door Lock should remain by configuring the option Open Door for in
the“Door Phone” parameters in “Basic Settings”.
How to use
The DOP will open and close door lock release device connected to it on the basis of the Open Door For timer or
on dialing the Feature Access Code for opening the Door. This feature command overrides the timer.
42. This is the default duration of this Timer. Your phone will display the time you have set.
What’s this?
Direct Inward Dialing (DID), an auto attendant feature, allows external callers to reach an extension directly without
the intervention of the Operator.
When DID is enabled on a trunk, whenever an external call lands on that trunk, the Built-In Auto Attendant or the
Voice Mail Auto Attendant (if Voice Mail module is installed) of ETERNITY NE, greets the caller and prompts the
the caller to dial the desired extension number. The call is then transferred to the extension number dialed by the
caller.
ETERNITY NE also offers Delayed DID, whereby incoming calls to Extensions defined as the landing destinations,
can be answered by the Built-In or the Voice Mail Auto Attendant, if none of the landing extensions answers the call
within a certain time period.
DID can be used by regular callers who know extension numbers to reach the desired extensions without Operator
assistance. Thus, DID reduces call traffic on the Operator extension, call set-up and transfer time for the callers. It
is useful especially during non-working hours and holidays, and projects a professional image of the organization.
Delayed DID works only when you have selected Operator or Extensions as the the landing destination
for incoming calls on the trunk.
DID will not work, when the dialed extension has Privacy from DID enabled in its Class of Service. So, if
you want to prevent direct access to certain extensions by external callers, enable Privacy from DID in
their Class of Service. See “Privacy”.
How it works
DID can be configured on all trunk types—CO, Mobile, SIP—for the Day (working hours) and Night (non-working
hours).
When configuring DID on a trunk, you may select either the Built-In Auto Attendant or the Voice Mail Auto Attendant
as the destination for incoming calls.
• The system waits for the period of the DID Answer Wait Timer (default: 05 seconds) to answer the call.
The caller gets Ring Back Tone from the CO network during this period.
• The system greets the caller with the pre-recorded voice message called the DID Welcome Greeting for
the current time zone. A Voice Module must be assigned for the DID Welcome Greeting.
• On the completion of the Welcome Greeting or music-on-hold at the end of the DID Music Timer, the
system plays the DID Dial Message to prompt the caller to dial the desired extension number.
The DID Dial Message is played once and the caller gets Beeps. The system waits for the DID Beeps
Timer (default: 10 seconds) to expire.
• If the caller does not dial any number before the DID Beeps Timer expires, the system plays the DID Call
Transfer to Operator message and transfers the call to the Operator.
The system waits for the duration of the DID Inactivity Timer (default: 60 seconds) for the Operator to
answer the call. If there is no answer at the end of this timer, the system releases the trunk.
If the caller fails to dial digits, you can have the call disconnected instead of routing it to the Operator. For
this, you need to enable the Disconnect DID call, when caller does not dial any digit flag in the System
Parameters. When this flag is enabled, the system will play the DID No Dial Voice message to the caller. If
the caller fails to dial a digit within the DID Beeps Timer, the system will disconnect the call.
• If the caller dials the extension number, the system checks if the number is valid.
If the dialed digits are invalid, the system plays Wrong Dial voice message to the caller. This message is
played once. The system waits for the duration for the DID Error Tone Timer (default: 5 seconds).
If Wrong Dial Voice Message is not programmed, the system plays Error Tone to the caller for the duration
of the DID Error Tone Timer, followed by the DID Dial Prompt.
• If the number dialed by the caller is valid, the system checks if the dialed extension is free.
• If the dialed extension is busy, the system plays the DID Busy Message to the caller. The message is
played once.
• If no DID Busy Message is programmed, caller will hear Busy Tone, played for duration of the DID Busy
Tone Timer (default: 15 seconds), followed by the DID Dial Prompt.
To have the call disconnected if the dialed extension is busy, you may enable the Disconnect DID Call,
when dialed number is busy flag in the System Parameters.
• The dialed extension is free. The system calls the extension and plays DID Ring Back Tone Message (if
programmed) or Ring Back Tone to the caller. This message is played until the dialed extension is ringing.
• The system waits for the period of the DID Ring Timer for the dialed extension to answer the call.
• When the dialed extension answers the call, the caller gets connected to the extension.
If the dialed extension does not answer before the expiry of the DID Ring Timer, the system prompts the
caller to dial again with the DID Dial Prompt message to the caller.
• The system diverts the call to the Operator. When the call is transferred to the Operator, the system plays
the DID Call Transfer to Operator voice message (if programmed) or plays Ring Back Tone to the caller.
When the Voice Mail Auto Attendant of ETERNITY NE is selected as the destination for incoming calls on a trunk,
this is how DID will work:
• The Voice Mail System (VMS) installed in the ETERNITY NE answers the call.
• The VMS greets the caller with the Welcome messag and the Greeting Message selected for the current
time zone (working hours and non-working hours).
• The VMS plays prompts to the caller to process the call further according to the Voice Mail Auto Attendant
Profile you assigned to the trunk.
Delayed DID
You can use Delayed DID to have incoming calls not answered by the landing destination(s)—Operator or
Extensions—within a certain time period, to be handled either by the Built-In or the Voice Mail Auto Attendant.
• As a call lands on a trunk, the system checks the Incoming Call Route configured for the current time zone
for the trunk.
• On finding Operator or Extensions as the landing destination, the system rings on the destination
extension(s) for the duration of time defined for ringing the extension (default: 10 seconds).
• If no reply is received from the extensions, the system routes the call to the auto attendant you selected:
Built-In or Voice Mail Auto Attendant.
How to configure
To use the Voice Mail Auto Attendant on trunks, do the following:
1. make a list of the trunks by their port type (“CO Trunks”, “Mobile Trunks”, “SIP Trunks”) and port number
on which you want to use the Voice Mail Auto Attendant.
2. configure Welcome and Greeting messages. You may either use the default, pre-recorded welcome
messages of the VMS, or record in WAV format, the custom welcome messages that meet your
requirements. See “Recording Voice Messages”
5. select the option Voice Mail Auto Attendant to Route Incoming Calls during the Day and the Night on
the desired trunks. For instructions see “CO Trunks”, “Mobile Trunks” and “SIP Trunks”
6. for each trunk port, configure Voice Mail Auto Attendant; select the Voice Mail Auto Attendant Profile
number.
1. make a list of the trunks by their port type (CO, Mobile, SIP) and port number on which you want to use the
Built-In Auto Attendant.
2. select the option ‘Built-In Auto Attendant’ to Route Incoming Calls during the Day and the Night on the
desired CO, SIP, Mobile trunks.
3. assign Voice Modules for DID Messages. To play to callers pre-recorded voice messages as DID
greetings and voice prompts at each stage of the DID call, you need to assign Voice Modules for the
following DID Messages:
• DID Welcome Greeting: played to callers when answering the DID call. Different welcome greetings
can be programmed for Working Hours and Non-working Hours. The DID Welcome Greeting message
is played once.
• DID Dial Prompt: played after the Welcome greeting message to prompt the caller to dial the desired
extension number. This message is played once.
• DID Ring Back Tone: played after the caller has dialed the number and the system is ringing the
dialed extension. This message is played continuously as the dialed extension rings.
• DID Wrong Dial message: played when the caller dials a wrong number or the number dialed by the
caller does not match with any extension number of ETERNITY NE. This message is played once.
• DID Destination Busy: played when the dialed extension is busy. This message is played once.
• DID Destination No Reply: played when the dialed extension does not respond. This message is
played once.
• DID No Dial: played when the caller has not dialed any number. This message is played once.
• DID Call Transfer to Operator: played to the caller when the call is being transferred to the Operator.
This message is played once.
Pre-recorded DID voice messages are provided in WAV format on the CD-ROM provided to you with
the ETERNITY NE.
Voice Module
Voice Message Application Voice Message
Number
06 DID Welcome Greeting for Night time Welcome! I am sorry, we are closed.
(Non-working and Break hours)
08 DID - No Dial message Sorry! You have not dialed any number.
11 DID - Destination Ringing message The number you have dialed is ringing.
(Ring Back Tone)
12 DID - Destination No Reply message The person you dialed is not responding.
13 DID Call Transfer to Operator message Please hold, transferring your call to the
Operator.
You may customize these DID voice messages by recording messages of your choice and assigning
them to the voice modules. For instructions on recording messages on the voice modules and
assigning voice modules, see “Voice Message Applications”.
If you do not use any of the above voice modules, the system will play the Call Progress Tone for each call
state.
• Disconnect DID call, when dialed number is not responding: when this flag is enabled the system,
if there is no reply from the landing destination extension(s), the system will disconnect the DID call
instead of routing it to the Operator. Default: disabled.
• Disconnect DID call, when caller does not dial any digit: when this flag is enabled, if the caller fails
to dial a digit within the DID Beeps Timer, the system will disconnect the DID call instead of routing it to
the Operator. Default: disabled.
What’s this?
With Direct Inward System Access (DISA) remote users can access and use the system's features and facilities
using Trunks, on which this feature is enabled.
All these can be done as if being done from a local extension of the ETERNITY NE.
DISA Variants
ETERNITY NE offers three types of DISA, each with a different method of authentication and level of access:
The callers are authenticated and allowed to use the extension on which they are logged in.
The callers must dial special digits or codes to go On-hook, Off-hook. They are allowed to make as many trunk
calls and internal calls as long as they are logged into the DISA mode.
To end the DISA login session, callers must dial the Termination code or disconnect from the remote end.
Callers can access an extension to use DISA PIN Authentication-Multiple Calls only if the extension has DISA
feature enabled in its Class of Services.
Callers are not required to dial any DISA Login Code or any password.
When the caller is authenticated on the basis of CLI, the system plays the ('internal' system) Dial Tone.
The callers must dial special digits or codes to go On-hook, Off-hook. They are allowed to make as many trunk
calls and internal calls as long as they are logged into the DISA mode.
To end the DISA login session, caller must dial the Termination code or disconnect from the remote end.
For this type of DISA, the DISA CLI Authentication Table must be configured first.
When the caller is authenticated on the basis of CLI, the system gives the caller direct access to the Outgoing
Trunks selected for dialing ‘0’. It plays the dial tone.
Callers are allowed to make a single external call. The system ends the DISA session on the completion of the call
by the caller or by the other remote party
For this type of DISA, the DISA CLI Authentication Table must be configured first.
To make another call, the caller must enter the DISA mode again, by calling the ETERNITY NE from the remote
location.
This feature allows access to system resources to remote users, and therefore has serious implications for
your system's security. Protect your system from unauthorised access and misuse.
How it works
For this feature to work, you must enable the desired DISA variant on the desired trunks: CO, Mobile, SIP.
• The system checks if a DISA variant is enabled on the trunk for the current time, i.e. Day or Night.
• If a DISA variant is enabled on the trunk, the system processes the call according to the DISA variant
enabled on the trunk.
• The caller must dial the DISA Login Code consisting of:
• the DISA Feature Access Code.
• the number of the extension the caller wants to access.
• the user password of the extension.
• On successful login, the system starts the DISA Idle State Timer (configurable; default: 20 seconds).
The system waits for the caller to go Off-hook44.
• When the caller goes Off-hook by dialing the Off-hook code #1, the system plays the internal dial tone
and waits for the caller to dial digits.
• If the caller dials an external number using a CO trunk, the system starts the DISA Inactivity Timer
(configurable; default: 2 minutes).
43. If no voice message is recorded, the system plays music-on-hold to the caller.
44. If the caller does not go Off-hook within this timer, the system releases the call.
• The system reloads this timer each time it receives digits from the caller. If the caller fails to dial any
digit within this timer, the system plays beeps for the duration of the DISA Warning Beeps Timer (fixed;
15 seconds). If no digit is received at the end of the Warning Beeps, the system terminates the DISA
session. If digits are recieved before the end of the Warning Beeps, the system reloads the DISA
Inactivity Timer.
• The caller can make as many trunk calls and internal calls.
• The caller can terminate the DISA login session either by disconnecting from the remote end or by
dialing the Termination Code #9.
• The system compares the CLI of the caller with the Calling Party Numbers configured in the CLI
Authentication Table.
• If the CLI matches with any of the Calling Party Numbers in the Table, the system provides access to
the extension configured as Auto Login extension for this Calling Party Number in the Table45.
• The caller gets logged into the Auto Login extension and gets the dial tone of ETERNITY NE.
• Now, the caller dials codes for On-hook #0, Off-hook #1 to make as many trunk and internal calls as
desired.
• If the caller dials an external number using a CO trunk, the system starts the DISA Inactivity Timer
(configurable; default: 2 minutes).
• The system waits for the caller to dial digits within the DISA Inactivity Timer.
• The system reloads this timer each time it receives digits from the caller. If the caller fails to dial any
digit within this timer, the system plays beeps for the duration of the DISA Warning Beeps Timer (fixed;
15 seconds). If no digit is received at the end of the Warning Beeps, the system terminates the DISA
session. If digits are recieved before the end of the Warning Beeps, the system reloads the DISA
Inactivity Timer.
• The system compares the CLI of the caller with the Calling Party Numbers configured in the CLI
Authentication Table.
• If the CLI matches with any of the Calling Party Numbers in the Table, the system provides access to
the extension configured as Auto Login extension for this Calling Party Number in the Table46.
• The caller gets logged into the Auto Login extension and gets dial tone of the outgoing trunks selected
for dialing ‘0’.
45. If no match is found for the CLI of the caller in the Table, the call will be routed as per the Incoming Call Route configured in ETER-
NITY NE.
46. same as previous note.
• The system waits for the caller to dial digits within the DISA Inactivity Timer. If the caller fails to dial any
digit within this timer, the system plays beeps for the duration of the DISA Warning Beeps Timer (fixed;
15 seconds). If no digit is received at the end of the Warning Beeps, the system terminates the DISA
session. If digits are recieved before the end of the Warning Beeps, the system reloads the DISA
Inactivity Timer.
• After the external call is completed, i.e. the caller disconnects from the remote end or the other remote
called party has disconnected, the caller is logged out.
• To make another external call, the caller must call the DISA enabled trunk of ETERNITY NE again.
• In all the variants of DISA, the caller can use all the features allowed in the “Class of Service (COS)” of the
extension the caller is logged into (using PIN Authentication or CLI Authentication).
• DISA Inactivity Timer is not applicable for SIP and Mobile trunks.
• DISA calls in the SMDR report are marked as "O" in the remarks column. See “Station Message Detail
Recording–Report”.
• If DISA is disabled, ETERNITY NE will route the call by DID logic, if DID is enabled. If DISA and DID
both are disabled, the incoming call will be routed as per the Incoming Call Route logic. To know more,
see “Direct Inward Dialing (DID)”.
How to configure
To provide DISA to remote users you need to do the following configuration:
• Select the DISA variant for the Trunks in the Route Incoming Calls option on the trunk port parameters.
See “CO Trunks”, “Mobile Trunks”, and “SIP Trunks” under Basic Settings for instructions.
• Enable DISA in the“Class of Service (COS)” of the extensions which you want to allow callers to access
using DISA. See “SLT Extensions”, “DKP Extensions”, “SIP Extensions” under Basic Settings for
instructions on configuring Class of Service for the different Extension types.
• Change the User Password of the DISA extensions, if you selected DISA PIN Authentication-Multiple
Calls,If you selected DISA PIN Authentication-Multiple Calls on a trunk. The default User Password (1111)
will not work. See “User Password” and “System Security” for more information and instructions.
• Configure the related timers, DISA Idle State Timer and DISA Inactivity Timer, if required. See “System
Timers and Counts” for instructions.
• If you have selected the DISA CLI Authentication-Multiple Calls or CLI Authentication-One Call on a trunk,
you must configure the CLI Authentication Table.
• Make a list of remote users and their numbers whom you want to allow DISA.
• For each remote user’s number on your list, write the Extension number of the ETERNITY NE you want to
allow this extension user to log in.
• Open Jeeves.
• Click the DISA - CLI Authentication link. The CLI Authentication Table page opens.
• You can configure as many as 999 numbers in this table, by clicking the tabs of the index on the top of the
table.
• Refer to the list of remote user numbers and the corresponding ETERNITY extension numbers you made.
• In the Calling Party’s Number column, enter the number of the remote users whom you want to allow
access to DISA using CLI Authentication. The system will match the CLI of the callers with the numbers
you store here.
How to use
If you are a Remote user, to be able to use DISA, you must know:
• the number of the Trunk on which DISA is enabled and the variant of DISA enabled on this trunk.
• the number of the extension and the user password which you want to access, if using DISA with PIN
Authentication.
• the duration of the DISA related Timers: DISA Idle State and DISA Inactivity Timer, so that you may dial
digits accordingly, without delay.
However, ETERNITY will not be able to understand the conventional way of dialing 'flash' key or going on-hook with
momentary make/break of loop current. Therefore, ETERNITY supports specific codes, which it can interpret if
these are received during DISA session.
When you are in DISA mode, use the following codes to indicate an activity:
on-hook #0
off-hook #1
Flash #2
Pause #3
A #4
B #5
C #6
D #7
+ #8
# ##
End of String #*
To use DISA,
• Dial the number of the Trunk on which DISA is enabled for the current time zone, i.e. Day or Night.
• ETERNITY answers the call. You will get music or DID Voice Message, if configured.
The DSS Console is a two-wire digital terminal to be used as an add-on module for the digital key phone, EON, as
an extension of its key map.
The DSS Console provides you quick access to Extensions, Trunks, Features/Functions of the ETERNITY NE, all
at the touch of a single key, making your call operations easier.
While the DSS Console is more commonly used by the Operator/receptionist in an organization, it meant for use by
anyone who needs to access the many features of the ETERNITY NE at a single touch of a button.
The DSS Console extends the existing key map of the DKP. When DSS64 is attached to EON48, you have in all 93
DSS keys (29 DSS keys of EON48 and 64 DSS keys of DSS64).
A DSS Console occupies a Digital Key Phone Port. So, you can use only one DKP if you want to use a DSS
Console with it.
The DSS Console can be attached with the ETERNITY NE in the same way as the DKP, EON, and is programmed
as an attachment of the DKP.
You can program Extension numbers or features/functions on the keys on the DSS Console in the same way as
you would program the DSS keys of EON, so that they can be accessed simply by pressing a single key.
LED
Each DSS Console key is equipped with an LED which glows in single (Red) or in tri-color (Green, Red, Orange)
depending on the function assigned to it.
When an extension or Trunk is assigned to a DSS Console key, the LED functions as a tri-color LED to show the
status of the extension/trunk (whether ringing, busy, in speech, on hold).
The LED color and cadence of the DSS Console keys is the same as that of the DSS keys of EON. Refer the topic
“Digital Key Phone-Operation” to know more.
Not all Features/Functions of ETERNITY require an LED. So, the LED of the DSS keys function only if the feature/
function assigned to these keys require LED indication.
What's this?
Distinctive Rings are ringing patterns used for distinguishing between different types of call events. With Distinctive
Rings, it is possible to use ring cadence of user's choice for each of these call events. For instance, Triple ring can
be set for 'Priority Internal Calls' and long rings can be set for 'Alarm Calls'.
A set of ring types is called Distinctive Ring type. The default Distinctive Ring Types are:
Call Event Ring Type set 1 (T1) Ring Type set 2 (T2) Ring Type set 3 (T3)
Auto Redial Very Long Slow Very Long Slow Very Long Slow
Programming
Continuous Continuous Continuous
Ring
Double 400-200-400-2000
400-200-400-200-400-
Triple
2000
How it works
At the time of installation, when the System Engineer selects the “Region” (as per the geographical location of the
system), ETERNITYNE loads the country-specific Distinctive Ring Type defined for the selected Region.
Refer the topic “Default Settings” for the default Distinctive Ring Type applied to your country/region.
Demonstration of rings
It is possible to demonstrate Ring Types to users by dialing the SE commands from EON/Extended IP Phone or an
SLT.
By default, the system will play each Ring Type as demonstration for 30 seconds. The duration of demonstration
can be changed by setting the 'Tone Demo Timer' to match user preference.
How to configure
The country-specific Distinctive Ring Pattern is set automatically by the system when you select the Region Code
and issue command to default the system. However, if required, you may change the default Ring Pattern loaded
by the system.
• Open Jeeves.
• Select the desired Ring Type for each call event/feature that you want to customize.
• To change Demonstration Timer, click the System Timers and Counts link under Advanced Settings.
• Set the Tone Demo Timer to the desired duration, e.g. 045 seconds.
How to use
Users of ETERNITY may be acquainted with the different Distinctive Rings played by the system so that they can
associate the terms used to describe the rings with the sound emitted by the system for each ring.
Ring Types can be demonstrated to extension users by dialing the SE commands from EON, the Matrix Extended
IP Phone or an SLT.
01 for Continuous
02 for Short Fast
• You get the prompt 'Go Idle for Ring' on your phone display.
• Go Idle.
• Exit SE mode.
01 for Continuous
02 for Short Fast
03 for Short Long
04 for Short Very Slow
05 for Long Fast
06 for Long Slow
07 for Very Long Slow
08 for Double
09 for Triple
• Exit SE mode.
What's this?
Extension users may wish to restrict calls to their extensions in order to work uninterrupted by frequent phone calls.
The feature, Do Not Disturb, enables users accomplish this.
This feature is useful to extension users who are in the middle of a meeting or any important work that requires their
undivided attention.
Doing so, calls from other extensions will be barred. However, the extension user would continue to receive:
• all the external calls transferred by the Operator/any other phone, DID Calls and DDI Calls.
• Alarm calls.
• Reminder calls.
• Auto Call Back calls.
DND has three supplementary features: “DND-Override”, “DND Text Message” and “Voice Message for DND
Notification”.
DND-Override
As the feature title suggests, 'DND-Override' allows the caller to land on the called extension, despite DND set on
the extension.
DND-Override will not work if the called extension has 'Privacy from DND-Override' enabled in its Class of Service.
When setting DND (also DND-Remote), the extension user/Operator can select an appropriate text message to be
displayed to the calling extension.
This DND text message is displayed on the calling extension, but only if the calling extension is the proprietary
digital key phone, EON.
The ETERNITY supports 9 different DND Text Messages, which can be changed as per user requirement by the
System Engineer. Users can select and set on their phones any of the DND messages programmed by the System
Engineer.
Voice Message Notification for DND is particularly useful when the extension phone on which DND is set is an SLT.
When DND is set on an extension of ETERNITY, callers who try to reach that extension are played an error tone.
Callers who are using EON are displayed the DND Text Message set by the called extension, and thus come to
know the cause of the error tone. Such a facility is not available to callers who are using SLTs, who can hear only
the error tone and have no way of knowing the cause of the error tone.
Hence, the feature Voice Message for DND Notification, whereby a pre-recorded Voice Message notifies calling
extensions of the DND set on the called extension.
To play to the callers pre-recorded voice messages as DND Notification, you must record and assign a Voice
Module.
How it works
A, B and C are extension users.
B has EON, while C has an SLT.
B has DND-Override in his Class of Service, C does not have this feature.
DND Text messages as well as Voice Message Notification for DND have been programmed by the System
Engineer.
• A has set DND on his extension with the DND Text message 'In Meeting'47.
• B calls A.
• As B has DND-Override, the Voice Message for DND Notification is played to B once, and the DND
message 'In Meeting' set by A appears on B's phone display. B gets routing Beeps.
• To exercise DND-Override, B must dial '4' the feature access code for 'DND-Override' during either during
the Voice Message or during the routing Beeps.
• B gets Ring Back Tone, if A's extension is free.
• B gets Busy Tone, if A's extension is busy.
• However, if A has Privacy from DND Override, B will get error tone and the DND message set by A
appears on B's phone.
If B fails to dial the DND-Override code before the end of the routing beeps, error tone will be played to
him.
• C calls A.
• As C has an SLT, C will get only the Error tone.
• But as Voice Message for DND Notification is programmed in the system, C will be played the pre-
recorded message once.
• Since C is not allowed 'DND-Override' in his Class of Service, he cannot exercise this feature during the
Voice Message.
• At the end of the voice message, C will be played error tone.
47. While DND and DND Text Message can be set from any phone, DND Text Message can be viewed on EON and the Extended IP
Phone only.
• DND, DND-Override and Privacy from DND-Override must be enabled in the Class of Service (COS) of
the extension(s) that are to be allowed this feature.
By default, all extension port types have DND enabled for the Day and Night in their Class of Service. So,
all extensions of ETERNITY can set and cancel DND.
DND-Override and Privacy from DND-Override are disabled in the CoS of all extensions by default. So,
none of the extensions can use DND-Override, or be exempt from DND.
While it makes sense to offer all extensions DND, providing DND-Override and Privacy from DND also to
all extensions will not serve the purpose of DND.
Decide which extensions are to be allowed 'DND', which are to be allowed 'DND-Override', and which are
to be allowed 'Privacy from DND-Override'. Generally, DND-Override is allowed to the Operator extension.
It may be allowed to extensions of persons in senior positions in the organization. Similarly, Privacy from
DND-Override may be allowed to persons in senior positions in the organization.
If you want to allow DND only to selected extensions, disable this feature in the CoS of all other extensions
other than these extensions.
Similarly, if 'DND-Override' is to be to be allowed to the Operator and a few other extensions, enable this
feature in the CoS of the Operator and other extensions.
If 'Privacy from DND-Override' is to be allowed to certain extensions only, enable this feature in the CoS of
these extensions.
For instructions on configuring DND in the “Class of Service (COS)” of the extensions which you want to
allow this feature, see “SLT Extensions”, “DKP Extensions”, “SIP Extensions” under Basic Settings.
• You may configure the DND Text Messages, see “System Parameters” for instructions.
• You may also configure the Voice Message for DND Notification, as per user requirements.
To be able to play a voice message to callers for DND notification, you must first record a Voice Module
with the desired message.
Record a Voice Module with the message "The dialed extension has activated Do Not Disturb"
(recommended).
Assign the Voice Module to the Voice Message Application defined for 'DND Notification'.
OR
• Dial 18
• Scroll to select from any of the DND messages that appears on the phone's display:
• Do Not Disturb
• Unavailable
• In Meeting
• In Conference
• Try on Mobile
• On Vacation
• On Business Trip
• Out of Office
• With a Guest
To cancel DND:
• Dial 18-0
• You get a text message 'DND Cancelled' on the phone's display and confirmation tone.
To cancel DND:
DND-Remote
• Do Not Disturb
• Unavailable
• In Meeting
• In Conference
• Try on Mobile
• On Vacation
• On Business Trip
• Out of Office
• With a Guest
To cancel DND-Remote,
DND-Override
What's this?
A Door Phone is typically used for monitoring an entrance door. It is installed in place of the Doorbell.
The door phone is similar to any ordinary phone; except it does not have a hook-switch or a dial pad. Usually, it is a
weather tight box, equipped with a button like a doorbell, which visitors press.
When visitors press the Door Phone Call Button, the phone programmed to receive the call (landing destination)
rings. The user of the called phone can answer the door phone call by simply lifting the handset and talk to the
visitor at the door. The user of the called phone can have the door opened for the visitor by either physically
appearing at the door or operating a door lock release device.
The Door Phone feature of the ETERNITY NE allows users to operate the door phone from a remote location (off-
premises) by having their calls routed to an external number.
The Door Phone feature of the ETERNITY NE is very convenient to have at:
• Delivery entrances: It is not necessary to have company personnel monitor delivery entrances. They can
just answer the Door Phone instead.
It is also possible for the doctor/pharmacist to have calls of the door phone landed on their mobile number
or any other fixed line external number, and open their clinic/pharmacy from their current (remote) location
to let the patient in.
• Residences and Apartment entrances: The identity of the visitors can be screened before letting them
in. The occupants of the house can greet their guests/relatives and let them enter the house also in their
absence by answering the door phone from their current (remote) location and opening the door for the
guests.
Matrix does not supply Door Phones. Any standard 4-wire Door Phone can be connected to the door
phone port of ETERNITY NE.
How it works
The Pre-requisites
• A four-wire Door Phone connected to the Door Phone Port of the ETERNITY NE.
• A Door Lock Release device connected to the Digital Output Port (DOP) of the ETERNITY NE, if a Door
Lock Release is to be used in conjunction with the Door Phone.
When the landing destination for the Door Phone Call is an extension of ETERNITY NE,
• The system plays a distinct Ring Type on the extension to indicate to the extension user(s) that it is a door
phone call. The Ring Type is programmable; by default Triple Ring is set as ring type.
• The visitor is played Ring Back Tone while the destination extension rings for the duration configured as
‘Ring Door Phone call for’ parameter in the Door Phone settings (default: 30 seconds)48.
If the destination extension does not answer the call within this time period, the Door Phone call is dropped
and the Door Phone goes idle.
• When the destination extension answers the call, the Door Phone circuit is activated and two-way speech
is established between the visitor and the extension user.
• The extension user may now either physically appear at the door to open it, or dial the Open Door Lock
Code (default: 1173) from the current extension.
• The Door Phone goes idle only when the extension user goes ON-Hook.
• The extension user can open the Door Lock also by dialing the feature command to operate the DOP.
For this, however, the extension must have the facility ‘DOP Turn ON/Turn OFF' in its “Class of Service
(COS)”. Refer the topic “Digital Output Port (DOP)” for operation instructions.
• It is possible for the any extension of the ETERNITY NE to establish speech with the Door Phone even
when it is idle, by dialing the unique access code assigned to the Door Phone.
When the landing destination for the Door Phone Call is an external number,
• When the visitor presses the Door Phone button, the ETERNITY NE makes a call to the pre-programmed
external number, which may be a fixed line or a mobile number.
48. The ‘Ring Door Phone Call For’ timer determines the time for which the landing destination shall ring for the door phone call. This
Timer is necessary because often visitors may press the door phone switch as they would do a door bell, for one or two seconds
only, whereas the call must remain present for a longer period of time for it to be answered.
49. This is the time for which the door will remain open.
50. An error tone will be played to the extension user for the duration of the error tone timer, if a DOP has not been assigned to the
Door phone port.
• When the external number answers the call, speech is established between the visitor and the external
number.
• The called party on the external number ascertains the identity of the visitor.
• the called party on the external number puts the visitor on hold by dialing '#2'. The system plays music-
on-hold to the visitor.
• the called party on the external number dials the Feature Access Code for 'Open the Door' (default:
1173)51.
• the Door Lock is opened for the duration of the Timer 'Open Door for’, (default: 5 seconds)52,53.
• as soon as this Access Code is dialed, speech is reestablished with the visitor.
• the visitor is invited to enter the building.
• the called party on the external number disconnects the call54; the Door Phone goes idle.
This way, it becomes possible to answer the door bell call from a remote location and open the door lock
from the remote location.
• The Door Lock can be opened by the called party on the external number also by dialing the command
for operating the DOP to which the Door Lock is connected, instead of dialing the 'Open the Door'
access code. Refer the topic “Digital Output Port (DOP)” for operation instructions.
• The called party on the external number can also call the Door Phone using Direct Inward System
Access (DISA).
• It is also possible for the called party on the external number to make multiple calls, while putting the
visitor at the Door Phone on hold. For this, the party must be logged in the “Direct Inward System
Access (DISA)” mode.
• The Door Phone feature of the ETERNITY NE offers the flexibility of selecting the Routing Mode for Door
Phone calls. The system supports two Door Phone Call Routing Modes:
• At Wish: Whenever you want to route the Door Phone Calls, you can alternately select the landing
destination, i.e., select a group of extensions at one time, an external number (the programmed fixed
line or mobile number) the next time, as required. The extension user can change the Call Routing
Mode, i.e. the landing extension or external number, as and when required.
• Scheduled: You can program the system to route Door Phone Calls automatically to the landing
destination phone according to the time of the day: Day (working hours) or Night (non-working hours).
For example, you can have Door Phone calls during the Day landed on the extension(s) of the
ETERNITY NE, and Door Phone calls during the Night on an external number. It is also possible to
assign different groups of extensions for Day and Night time.
51. This access code is to be dialed only when in speech with the visitor. If this access code is dialed when there is no speech, an error
tone will be played to the extension user for the duration of the Error Tone Timer.
52. This is the time for which the door will remain open.
53. An error tone will be played to the extension user for the duration of the error tone timer, if a DOP has not been assigned to the
Door phone port.
54. If the trunk used for placing this call is a Two-Wire Trunk, dial #0 to disconnect the call.
• The landing destination - Extension Group and the External Number - can be configured only by the
System Engineer (SE).
• The Routing Mode for Door Phone Calls, 'Scheduled' or 'At Wish' mode can be set by the System
Engineer as well as the extension users (User Mode). However, extension users must have the feature
“Door Phone Settings” enabled in their '“Class of Service (COS)”. With this feature included in their
Class of Service, the extension users can switch between At Wish and Scheduled modes by dialing the
Feature Command.
• The Time Table, the landing Extensions, and the External Number for routing the Door Phone Calls
can be programmed by the System Engineer only.
• The Outgoing Trunks selected for routing door phone calls to an external number will be common for
both Scheduled and At Wish modes.
How to configure
For the Door Phone feature to work,
• Allow 'Door Phone Settings' in the '“Class of Service (COS)” of extensions defined as the landing
destination for Door Phone calls. By default, it is enabled on all extension types, SLT, DKP and SIP
extensions.
• If a Door Lock is installed in conjunction with the Door Phone, enable the feature 'DOP Turn ON/OFF' in
the “Class of Service (COS)” of extension that are to be allowed to open the Door Lock. By default, it is
enabled on all extension types, SLT, DKP and SIP extensions.
How to use
To select a Call Routing Mode55:
55. Manually by the user of the extension configured as the landing destination for the Door Phone calls.
Users worldwide:
• Dial 1172-Access Code of the Door Phone-1 to route calls to an extension.
• Dial 1172-Access Code of the Door Phone-2 to route calls to an external number.
Users worldwide:
• Put the Door Phone call on Hold.
• Dial 1173
What's this?
Dynamic Lock allows extension users to change the Toll Control Levels (Calling Permissions) of their extensions on
their own by dialing a code.
The System Administrator/Operator can also change the Toll Control Levels of extensions using Dynamic Lock.
With this feature, extension users can prevent misuse of outgoing call facility from their extensions, especially in
their absence.
There are four types of Toll Control Levels, starting from Level 0 to Level 3 that can be set for extension phones.
For each Toll Control Level from 0 to 3, a 'Call Privilege’56 is to be assigned and corresponding number strings to
be allowed and number strings to be denied for each Call Privilege are to be configured.
• Toll Control - Level 0 is Time Zone based, wherein the Call Privilege Type must be defined for the Day
(working hours) and the Night (non-working hours). For instance, you may define 'All Calls' as Call
Privilege for the Day, and 'No Calls' as Call Privilege for the Night.
By default, Call Privilege 'All Calls' is selected for the Day and Night.
• Toll Control - Level 1 is not based on Time Zones. By default, the Call Privilege Type for this level is
'Local Calls'.
• Toll Control - Level 2 is not based on Time Zones. By default, the Call Privilege type set for this level is
'National Calls'.
• Toll Control - Level 3 is not based on Time Zones. By default, Call Privilege 'No Calls' is selected for this
level.
The Call Privilege for each of the above Toll Control Levels can be redefined according to user
requirements.
For example, Toll Control Level 3 can be redefined for allowing all types of calls by selecting 'All Calls' as
Call Privilege Type for Calls allowed for Lock Level 3. Similarly, you can configure Level 0 to allow Local
Calls and configure the Local Number strings in the list of Local Numbers.
Extension users who are allowed the Dynamic Lock feature in their Class of Service, can set the Toll
Control Level in two ways:
• Manually: the extension users change the Toll Control Level of the extension whenever they wants by
dialing the feature access code.
For example, an extension user having Toll Control Level 2 (default: National calls) can restrict long
distance dialing on the extension by setting the Toll Control Level to 1 (default: Local calls) before
56. The Call Privilege types are: No Calls, Local Calls, Regional Calls, National Calls, Limited Calls, All Calls.
Thus the extension user sets Dynamic Lock s/he manually selects the desired Toll Control Level for
his/her extension and restores the original Toll Control Level assigned to the extension.
• Automatically: the extension user changes the Toll Control Level of the extension using the Dynamic
Lock Timer. The user sets the Timer to the desired number of minutes. On the expiry of this Timer, the
system restores the original Toll Control Level assigned to the extension.
For example, an organization has defined Toll Control Level 0 as Local Calls, and Level 3 as All Calls.
An extension user of this organization is assigned Level 0. When this extension user wants to make
international calls, the user sets the Dynamic Lock Timer and selects Toll Control Level 3. At the end of
the timer, Level 3 gets locked and Toll Control Level 0 is reapplied on the extension phone.
• The changing of Toll Control level requires the extension user to dial the four-digit User Password. The
system will not accept the default User Password (1111). The extension user must first change the
default User Password.
• The Dynamic Lock Timer must be set to '00' when using Manual Dynamic Lock.
How it works
The Pre-requisites
• The Calls allowed for Lock Levels 0 (allowed for Day and Night), Level 1, Level 2, and Level 3 are
configured in the Toll Control applied on the extension.
The Process
OR
• The Operator sets Dynamic Lock manually for an extension by entering the extension number and
selecting the Toll Control Level.
OR
• Now, whenever a call is made from extension A, the system checks the Toll Control Level assigned to the
extension.
• The system then checks the associated Lists of allowed and denied numbers.
• If the Toll Control Level is 0, then Toll control is time zone based, i.e. Day and Night. The outgoing call
is allowed/denied as per the Call Privilege and the corresponding Allowed and Denied Number List
programmed for that time of the day by the System Engineer.
• If the Toll Control Level is 1, 2, 3 the outgoing call is allowed/denied as per the Call Privilege and the
corresponding number list programmed for each level.
• If Dynamic Lock - Automatic has been set by user/Operator, the system waits for the duration of the
Dynamic Lock Timer set for the extension. At the end of each outgoing call made during the period of this
Timer, the system will restart the Timer again. The system will change the Toll Control back to the pervious
Level when no outgoing call is made till the expiry of this Timer.
• If Dynamic Lock - Automatic has been set by user/Operator, and an internal call is made during the period
of the Dynamic Lock Timer, the system will check for the 'Decrement Dynamic Lock Timer for Internal
Calls' feature in the Class of Service of allowed to the extension.
If this feature is enabled, the system will start the decrement of the Dynamic Lock Timer. The system will
change the Toll Control back to the previous level on the expiry of this Timer. However, if the 'Decrement
Dynamic Lock Timer' feature is disabled in the Class of Service, the system will reset the Toll Control as
described in the previous step.
• If Dynamic Lock - Manual has been set, the extension user/Operator must set the Toll Control Level back
to the previous Level.
Feature Interactions
• Redial and Auto Redial: The system will check for Toll Control Level when an extension, on which
Dynamic Lock is set, attempts Redial or Auto Redial.
• Emergency Number Dialing: All extensions will be able to dial Emergency numbers always, regardless
of the Toll Control set on them.
ETERNITY NE provides for a separate configuration of Emergency Numbers, which remain unaffected by
Dynamic Lock set on the phones. Refer the topic “Emergency Dialing” to know more about this feature.
How to configure
For this feature to work, it must be enabled in the Class of Service of the extensions; Toll Control Level must be
configured on the extension.
By default, Dynamic Lock enabled on all extension types during the Day and the Night. So, all extensions of
ETERNITY NE can set Dynamic Lock.
The 'Dynamic Lock Timer' is disabled in the Class of Service of all extension types.
To configure Toll Control Levels on the extension, refer the topic “Toll Control” for instructions.
How to use
Extension users can set Dynamic Lock—Manual and Automatic—by themselves or have it set by the Operator or
any other extension user for their extension.
The extension user/Operator must first change the Dynamic Lock Level and set the Dynamic Lock Timer.
To set Dynamic Lock-Manual, the extension user/Operation must set the Dynamic Lock Timer to 00.
• When the Dynamic Lock-Manual is set (Timer set to 00), the extension user/Operator must dial the
feature access code to restore the previous Toll Control Level.
• When Dynamic Lock-Automatic is set (Timer set to desired number of minutes), the system will restore
the previous Toll Control Level at the end of the Timer.
The extension users must change the default User Password to be able to set the Dynamic Lock on their extension.
Refer the topic “User Password” for instructions on changing the password.
OR
• Dial 141.
OR
OR
• Dial 142.
OR
OR
• Dial 142.
OR
OR
• Dial 1072-00257.
OR
57. The feature 'SA Extension' must be enabled in the “Class of Service (COS)” of the extension from which this code is being dialed.
If the feature ‘SA Mode’ is enabled in the Class of Service of this extension, the extension user will be prompted for the SA pass-
word.
OR
• Dial 1072-002.
OR
OR
• Dial 1072-002.
OR
What’s this?
When an emergency call is made from an extension, the system dials out the number using any of the free trunks
selected for routing Emergency Numbers. Since the number is dialed out by the PBX, the Emergency Service that
attends to the call will be able to locate the PBX, but not the extension that made the call.
Similarly, the Operator too has no way of knowing which of the extensions made the call, thus, making it difficult to
quickly reach and provide help to the extension that made the emergency call.
With the Emergency Detection and Reporting feature, the Operator can know from which extension the emergency
call is being made. Whenever an Emergency call is made by an extension user, the system detects and reports it to
the Operator extension.
How it works
When an extension of ETERNITY NE makes an emergency call,
• the system hunts for the outgoing trunks selected for routing emergency numbers, and dials out the
number from a free trunk.
• simultaneously, the system informs the Operator by ringing on the Operator extension(s) for the duration of
the Alarm Ring Timer (configurable; default: 45 seconds).
• If Operator is a DKP or Extended IP Phone, it will ring continuously, and an emergency message will be
displayed on the LCD.
The emergency message shows the number of the extension which has made the emergency call, in this
case, extension 203.
What's this?
The ETERNITY NE supports dialing of Emergency number immediately without any blocking.
When an extension user dials an Emergency number, the system will hunt for a free trunk from the outgoing trunks
selected for the emergency number. See “Emergency Numbers”.
The system will not apply any of the following on the extension dialing the Emergency number:
• Toll Control (Allowed Denied Numbers, Dynamic Lock)
• Call Budget (even when call budget is consumed)
• Call Duration Control
• Automatic Number Translation
The system will allow the extension to dial the Emergency number even in the following conditions:
• the extension is in Off-Hook state.
• the extension is in Standby Mode.
• the extension has grabbed the trunk line (using Trunk access code or selective access)
• the call state is in any state: Busy, Error, Confirmation.
• SIM card is not present in the Mobile port.
• Mobile port is not registered with the network.
• SIM PIN is not valid.
• the keypad of the extension phone is locked.
Emergency Number will always be out dialed through the outgoing trunks you selected for dialing these numbers,
except when you have grabbed a trunk using Selective trunk access code/Selective Trunk Access DSS key. In
which case, the number will be dialed only from the trunk you have grabbed.
Emergency dialing will not work if Mains Power to the ETERNITY NE fails.
How to configure
The Emergency numbers are fixed as per the Region where ETERNITY NE is installed. See the topic “Emergency
Numbers” under Basic Settings for instructions on configuring Emergency Numbers.
How to use
To dial an Emergency number,
• Go Off-Hook
• Dial the Emergency Number
OR
For example:
Dial 0-112
• Wherever the Trunk Access Code conflicts with the Emergency Number, the emergency number
should be dialed after dialing the Trunk Access Code.
• Let us take the example of Australia, where the emergency number is 000 and the trunk access code is
0. Now, when an extension user of ETERNITY located in Australia dials ‘0’ of the emergency number,
the system will consider it as trunk access code and will apply the trunk access code logic.
• Therefore, in such cases, the extension user must first dial the Trunk Access Code and then the
Emergency Number. In this case, the extension user must dial 0-000 for emergency number dialing, so
that the system will not wait for the Conflict Timer to apply the Trunk access code logic.
What’s this?
Public exchanges support features like call waiting, call forward, etc. To be able to use these features, users need
to dial certain codes during speech.
When a PBX is connected between the user and the central office, the codes for dialing the features of the central
office may clash with the codes for accessing the features of the PBX, making it difficult for users to access the
features of the central office while in speech.
To overcome this, ETERNITY NE supports Flashing on Trunks, which informs the system of the dialing of codes on
trunks by extension users.
How to configure
To be able to use this feature, Flashing on Trunks must be allowed to the extension in its “Class of Service (COS)”.
For instructions see “SLT Extensions”, “DKP Extensions”, “SIP Extensions” under Basic Settings.
How to use
• Press Flash
• Dial *.
• Dial PSTN Code.
What’s this?
Using this feature, you can make your calls follow you wherever you go. You can receive your calls on another
extension, whenever you want.
How it works
• A’s extension number is 201.
• B’s extension number is 203.
• A is currently at B’s extension.
• A wants to receive calls from extension 201 on extension 203.
• A sets Call Follow Me on extension 203.
• All calls landing on A’s extension 201 will be forwarded to extension 203.
• When A returns to extension 201, A cancels Call Follow Me.
• The extensions dial tone changes to feature tone if its calls are forwarded.
• Multiple users can use ‘Follow Me’ from the same extension.
• Follow Me can be overwritten. Extension A sets Follow-Me on extension B. After a period of time; goes
to extension C. A can receive calls on extension C by setting Follow Me on extension C. Follow Me set
by A on extension B will be cancelled.
• Follow Me cannot be chained. If extension A sets Follow Me to extension B. And extension B sets
Follow Me on extension C, Follow Me of extension A is automatically cancelled.
How to use
For EON & Extended IP Phone Users
What’s this?
Extension users can force other extension users to answer their calls when there is no response from the called
extensions.
How it works
Forced Answer can be requested from an SLT, a DKP and from the Extended IP Phone (calling extension).
However, the called extension (being forced to answer) must be either a DKP or an Extended IP Phone.
Forced Answer can be used when the called extension is busy, but has free call loops (call appearances)
for calls to land. For example:
• Extension A (SLT) calls extension B (DKP).
• B is in speech with extension C, but there is a call appearance free on B’s DKP.
• Since B’s DKP has a free call appearance, A’s call lands on B.
• B’s extension starts ringing.
• A dials Forced Answer code during ring back tone.
• Extension C is put on hold.
• A is in speech with B.
• A may now talk to B.
• When A disconnects the call, B is now in speech with C.
How to configure
To be able to use Forced Answer, extension users must have this feature enabled in their “Class of Service (COS)”
for the Day and Night time, as required. See “Basic Settings” for instructions on configuring the different extension
port types.
You can also dial ‘5’, the feature code for Forced Answer, immediately after dialing the desired extension
number, instead of dialing it during Ring Back Tone. This way, you can talk to the desired extension user
without waiting for the called extension user to answer your call.
What’s this?
Forced Call Disconnection enables extension users to disconnect a busy extension or a trunk at will, and free the
system resources (access to extension and trunk) for themselves.
How it works
Forced Call Disconnection of an Extension:
• A, B and C are extensions.
• A and B are in speech.
• C calls B and finds it busy.
• C uses Forced Call Disconnection by dialing the feature command.
• C gets confirmation tone, while A and B get error tone.
To be able to use Forced Call Disconnection, the extension user must have a higher “Priority” than the
extension user whom he/she tries to forcibly disconnect.
You are advised to restrict access to this feature only to important extension users. Extension Users who
are allowed this feature are advised to use it judiciously.
• Forced Release feature enabled in their “Class of Service (COS)” for the Day and Night time, as required.
See “Basic Settings” for instructions on configuring the different extension port types.
• Press the DSS Key assigned to Forced Call Disconnection on Busy tone.
OR
What’s this?
An organization may have a Centralised Information Office which provides information related to different
departments such as HR, IT, or General information. For each department in the organization, an extension number
can be defined as a Help Desk.
How it works
• Extension 202 is defined as Help Desk for HR policies and general rules.
• Extension 216 calls the Help Desk extension 202.
• If the Help Desk extension is busy, an Auto Callback request is set automatically on the Help Desk
extension.
• As soon as the Help Desk extension is free, the system will serve the auto callback request.
• The Help Desk extension calls back extension 216.
How to configure
To enable Help Desk feature on any extension,
• Click the link of the desired extension port type: SLT, DKP, SIP to which you want apply this feature.
• Select the extension number to which you want to apply this feature, by clicking the tab.
The selected extension tab opens.
• Select the Assign Help Desk function to this Extension check box.
The selected extension will be designated as Help Desk.
• Click Submit.
What’s this?
Hot Desking enables extension users to use all the properties of their own extension from another extension.
Hot Desking is useful for people who are often away from their own desks and must work from another. Hot
Desking allows them to use all the features and facilities of their own extension from another.
How it works
This feature is supported on DKP and SLT extensions only.
The User Password of both extensions involved in Hot Desking must not be 1111.
Hot Desking can be performed only when both the extensions are idle.
• When Hot Desk is performed from the Hot Desking extension, all the properties of the Host Extension are
copied to the Hot Desk Extension.
• On the Host Extension, the user cannot perform any activity except Cancel Hot Desking.
• You must cancel Hot Desk from both the Hot Desk Extension and the Host Extension.
• After cancelling Hot Desk, the Host Extension and the Hot Desk Extension acquire their original properties.
How to configure
For this feature to work, the feature 'Hot Desk' must be enabled in the“Class of Service (COS)” of the Host
Extension and the Hot Desk Extension. By default, this feature is disabled on all extensions, so none of the
extensions can use this feature. To enable this feature in the COS of the extensions, see “SLT Extensions”, “DKP
Extensions”, “SIP Extensions” under “Basic Settings”.
How to use
The User password of both the extensions involved cannot be default password.
• On the Hot Desk Extension, press DSS Key assigned to Hot Desk.
OR
• Dial 1091
• Enter own extension number (Hot Desk Extension Number)
• Enter own extension User Password
• You get confirmation, Hot Desk cleared
• Go ON-Hook.
• Go to the SLT extension (Hot Desk Extension) with which you want to swap your SLT extension (Host
Extension) properties.
• Lift the handset of the Hot Desk extension.
• Dial 1091
• Dial Host Extension number
• Dial Host Extension User Password
• Replace handset.
What’s this?
This features eliminates repeated dialing of numbers from an extension. Hotline can be set for: Internal Extension
Numbers, Department Groups, External Numbers, Outgoing Trunks.
How it works
• Holtine can be set from any SLT, DPK or Extended IP Phone.
• Select the type of Hotline as per your requirement: Internal Extension Numbers, Department Groups,
External Numbers, Outgoing Trunks.
• Configure the Hotline Timer. To provide immediate Hotline, set Hotline Timer to 00 and for Delayed Hotline
configure a value for the Timer as per your requirement.
C, the Sales Manager has to frequently dial the number of the B, Sr. Co-ordinator-Sales. C sets Hotline for
B’s number and also configures the Hotline Timer as 5 seconds.
• C goes OFF-Hook
• ETERNITY NE give dials tone and waits for 5 seconds
• If no digit is dialed within the expiry of this time by A, B’s number will be dialed out automatically.
• B in speech with A.
• To provide immediate Hotline, set Hotline Timer to 00 seconds, then as soon a C goes OFF-Hook, B’s
number will be dialed out.
How to configure
To be able to use Hotline, extension users must have this feature enabled in their “Class of Service (COS)” for the
Day and Night time, as required. See “Basic Settings” for instructions on configuring the different extension port
types.
What’s this?
For PBX users in countries, where the Calling Line Identification (CLI) received must be suitably modified before it
can be used to dial out the number, ETERNITY offers the feature ‘Incoming CLI Modification’.
The Incoming CLI received with the Country or Area Code, or both. However, the dialing pattern of the public
network may require the received CLI to be prefixed with additional digits, to dial out the same number. Or the
dialing pattern of the public network may require the CLI to be stripped off the prefixed digits to dial out the same
number.
With the feature ‘Incoming CLI Modification’ programmed, the ETERNITY detects whether the incoming CLI is a
local number, a national, or an international number. It modifies the incoming CLI accordingly, by adding or stripping
off the prefixed digits so that the number can be dialed out as per the dialing pattern supported by the public
network.
The modified CLI is presented to the extension phones and is stored in the “Call Logs”, and SMDR (see “Station
Message Detail Recording (SMDR)”). Extension users can call a number in the Call Logs without having to modify
the CLI manually.
How it works
• Incoming CLI Modification parameters must be programmed in the system considering the dialing pattern
supported by the local public network.
• Accordingly, ETERNITY matches the CLI received with the programmed parameters.
• It detects whether it is an international, national or local number.
• It modifies the CLI according as per the Modification parameters programmed.
• It presents the modified CLI to the extension; stores the modified CLI in the SMDR and in the Call Logs of
the extension, provided it is a digital key phone.
• When the received CLI is dialed out by the extension user from Call Log, ETERNITY dials out the same
number.
How to configure
For this feature to work, you must configure Incoming CLI Nomalization in the “System Parameters” of
ETERNITY NE. To do this,
• Enable Incoming CLI Nomalization: Enable this flag if you want to use the Incoming CLI Modification
feature. By default, this flag is disabled.
If you receive CLI in dialable format, there is no need to use this feature. In such case, keep the flag
disabled. You do not need to program any of the CLI Nomalization parameters.
• Country Code: Enter the Country Code of the country where ETERNITY is installed. The Country Code
helps ETERNITY detect whether the Incoming CLI received is a national or an international number. Do
not enter any prefix for the Country Code. For example, if your ETERNITY is installed in USA, enter only ‘1’
as the Country Code. Do not enter ‘+’ or “00’ as prefix to the country code ‘1’. By default the Country Code
is ‘91’ (India).
• Area Code: Enter the Area Code of the place where the ETERNITY is installed. The Area Code helps
ETERNITY detect whether the Incoming CLI received is a local number. Do not enter any prefix for the
Area Code. For example, if you want to enter Area Code for Mumbai, enter only ‘22’. Do not enter the
prefix ‘0’ to the area code. By default, Area Code is ‘265’ (Vadodara city).
• National Prefix: Enter the digits that are required as Prefix for dialing long distance, National (within the
country) numbers. The prefix may be upto 5 digits, with numbers from 00000 to 99999. By default, ‘0’ is set
as prefix for dialing national numbers.
• Area Code required to make local calls?: Depending on the dialing pattern of your local public
telephone network, you may choose from the following options:
• No: select this option if your public telephone network does not require the dialing of Area Code for
local numbers.
• Yes: select this option if your public telephone network requires you to dial the Area Code for local
numbers.
• Yes, with Prefix Digit: select this option if you public telephone network requires you to dial Area Code
with a particular Prefix for local numbers. If you select this option, you must also program the Prefix
digits for the Area Code.
• Prefix Area Code: If you have enabled Area Code required to make local calls? with Prefix Digit, in
the previous parameter, enter the prefix digits for the area code for local calls in this field.
What’s this?
Interrupt Request allows you to break into an on-going conversation after intimating the extension user about the
interruption.
In case of an urgent trunk call the operator can put the call on hold, interrupt the busy extension user to inform
about the urgent call and then transfer the urgent call.
How it works
• A, B and C are users of the system.
• C calls A.
• C gets Ring Back tone (RBT) and A gets beeps indicating him about a new call. If A dials Flash to answer
C’s call before the expiry of the Interrupt Request Timer, A will be in speech with C. B is put on hold and
will get music on hold.
• If A does not dial Flash before expiry of the timer, C’s call will be disconnected.
• After the conversation between C and A is over and C goes on-hook, speech between B and A will be re-
established.
Feature Interactions
• Call States:
• Interrupt Request works only if the dialed extension is busy. The dialed extension may be busy with
another extension or trunk (external number).
• Interrupt Request works only if the user about to be interrupted in is in a two-way normal speech with
another user or external party.
• It will not work if the busy signal is due to the user being OFF-Hook, or in the middle of dialing, or
accessing a feature of the PBX.
• “Call Toggle”: Once A and C comes in speech with each other, A can toggle between B and C using Call
Toggle feature.
• Privacy against Interrupt Request: If the feature 'Privacy against Interrupt Request is enabled for an
extension, it cannot be interrupted.
How to configure
To be able to use Interrupt Request, extension users must have this feature enabled in their “Class of Service
(COS)” for the Day and Night time, as required.
See “Basic Settings” for instructions on configuring the different extension port types.
How to use
What’s this?
ETERNITY offers a facility—Last Caller Recall— to trace the extension that last made the call to your extension.
How it works
• When the called extension answers, speech is established between B and the called extension user.
How to use
For EON & Extended IP Phone Users
• Lift Handset
• Press DSS Key assigned to Last Caller Recall.
OR
• Go OFF-Hook
• Dial 1092
The system dials out the extension number that last called your extension.
What’s this?
This feature redials the last external number string dialed from the extension.
How it works
• Extension A dials the feature access code for ‘Redial’.
• If Extension A is an SLT, the system dials the last external number dialed from Extension A using the same
trunk access code used for dialing that number.
• If Extension A is a DKP or an Extended IP Phone, all external numbers dialed by Extension A are
displayed on the phone’s LCD.
• Extension A may select the number to be dialed out. The system will dial out this number using the same
trunk access code used for dialing this number.
• If Extension A has ‘Dynamic Lock’ set and uses Redial feature, the system will check for Toll Control as
per the Lock Level set for Extension A before dialing out the number.
How to configure
No particular configuration is required for this feature to work. Redial is included in the Basic Features allowed to all
extensions by default in their “Class of Service (COS)”. So, all extensions can use the Redial feature.
How to use
For EON & Extended IP Phone Users
• Press Redial Key.
Or
• Dial 7
A List of external numbers last dialed will appear on your phone’s display.
• Scroll to select the desired number.
• Press Enter key.
• The system dials out the external number.
Least Cost Routing (also referred to as Automatic Route Selection) is an expense control feature of ETERNITY NE.
Least Cost Routing (LCR) is useful when there are different trunk lines for making outgoing calls, and the service
providers of these trunks offer different tariffs for calls made to certain locations or numbers or during a particular
time of the day.
When a call is made from an extension of the ETERNITY NE, LCR recognizes where the call is going to. It selects
the lowest cost trunk from among all the trunks allotted to that extension to make outgoing calls, depending upon
how the LCR is configured.
The system can be configured to select the most cost effective trunk for the time of the day when the call is made
from the extension, or to select the most cost effective trunk for the destination number dialed from the extension,
or to select the most cost effective trunk considering both time of the day and destination number.
1. Time-based LCR: This type of LCR may be used when you have trunk lines of more than one service
provider, and each offers a different tariff according to the time of the day.
For example, Service Provider 1 offers a lower tariff for calls made between 9am to 8pm, while Service
Provider 2 offers a lower tariff for calls made between 8pm to 9am.
When Time-based LCR is configured, the system uses the Online-dialing logic, whereby digits dialed by
the user are directly passed on to the trunk.
2. Number-based LCR: This type of LCR may be used when you have trunk lines of more than one service
provider, and each offers different tariffs according to the area or distance, or phone numbers dialed. For
instance, Service Provider 1 provides lower calling rates for calls made from City A to City B, than Service
Provider 2 and Service Provider 3.
3. Time- and Number-based LCR: This type of LCR is a combination of number and time based LCR, i.e.
the service providers offer different tariffs according to the time of the day as well as area/distance.
For example, Service Provider 1 offers lower rates for calls made from City A to City B during peak hours
9am to 8pm, as compared to Service Provider 2, whereas Service Provider 2 offers lower rates for calls
made from City A to City B during off peak hours (8pm to 9 am).
When Time+Number-based LCR is configured, the system uses Store and Forward dialing logic, whereby
digits dialed by the user are first stored at a memory location in the system, and then dialed out on the
lowest cost trunk.
4. Service Provider-based LCR: This type of LCR may be used when the same Service Providers offer
different rates for calls made to numbers within their own network and for calls made to numbers of
another Service Provider's network. For example, Service Provider 1 offers lower rates to call a Service
Provider1 number in City A and in City B, than for calling numbers of Service Provider 2 in the same cities.
This type of LCR may also be used when the same Service Providers apply different charges for different
subscriber services provided by them. For example, Service Provider 1 offers both Fixed Line as well as
Mobile services and applies different charges for fixed line and Mobile services.
ETERNITY also supports LCR based on Carrier Pre-Selection. This type of LCR is useful where there exist
different service providers for local and long distance calls. Refer the topic “Least Cost Routing - Carrier
Pre-Selection” to know more.
Cost Factor
For LCR to work, all trunks that are allotted to extensions for making outgoing calls, must first be assigned a Cost
Factor, starting from 01 to 10.
After assigning Cost Factor to Trunks, you must configure the Type of LCR to be used on CO, Mobile and SIP
Trunks.
• Make a table of the trunk types and assign a cost factor to each trunk type, as shown below.
CO1 BSNL 01
CO2 BSNL 02
Mobile 1 Reliance 03
Mobile 2 BSNL 04
SIP 1 Pulver.com 05
SIP 2 VoipTalk.com 06
• Configure the Cost Factor number you assigned to the Trunk types in their respective trunk parameters.
• For instance, assign Cost Factor 01 to CO1 and Cost Factor 02 to CO2. Similarly, assign Cost Factor 03
to Mobile Trunk 1, Cost factor 04 to Mobile Trunk 2, Cost Factor 05 to SIP trunk 1.
• For configuration instructions, refer the topics “CO Trunks”, “Mobile Trunks” and “SIP Trunks”.
• Define the Time Zone, i.e. the start and end time, when the LCR should be applied for the outgoing calls.
• Refer to the table you prepared for assigning Cost Factor to trunks.
• For example, you want calls made during 9am to 8pm to be routed through BSNL trunks (CO1 and CO2).
If these trunks are busy, you want the system to route calls through the BSNL mobile trunk. When this line
is busy, you want the system to attempt to route calls through the line of Reliance.
• You want calls made between 8pm to 9am to be routed through BSNL CO1 trunk only.
• At Time Zone Index 1, define the Time Zone start and end time in 24 Hours:Minutes format, enter the Cost
Factor you assigned to CO1 (01) and CO2 (02) as Preference 1 and Preference 2 respectively. Enter the
cost factor you assigned to and as Preference 3 and Preference 4 respectively.
1 09:00 20:00 01 02 04 03
2 20:01 08:59 01 01 01 01
• Similarly, at Time Zone Index 2, define the Time Zone in 24 Hours:Minutes format. Enter the Cost factor
you assigned to CO1, i.e. 01 as Preference 1, 2, 3, and 4. When calls are made during this time period,
they will be routed through CO1 only.
• If you have finished defining Time Zones and the preferred trunks for the time zones, configure the Time-
based LCR using Jeeves.
• Open Jeeves.
• Enter the values of the Time-based LCR you prepared on the sheet of paper in the appropriate fields.
• Enter each of the number strings at an Index number from 01 to 99. A Number string may be a complete
telephone number, a truncated phone number or an area code.
• For each number string you enter, select the Trunk with the lowest cost as your first preference, i.e.
Preference 1. Select the trunk of your second, third and fourth preference (in order of increasing cost.
When the trunk you selected as first preference is busy, the system will route the call through the next
trunk you have set that is free.
• Refer to the table you prepared for assigning Cost Factor to trunks.
All mobile numbers start with the number '9', which is prefixed with a '0' when making long distance mobile
calls, so enter '9' and '09' as the number strings. For '9' as well as '09', select the Mobile trunk(s) through
which the calls should be made in order of preference.
Similarly, all local numbers start with 2, so enter this number in the number string column, and select the
CO trunk in the order of preference. As in this example, you have only two CO trunks, so you may keep the
same two trunks as your preference.
Cost Factor
Index Number
Preference 1 Preference 2 Preference 3 Preference 4
1 9 04 03 01 02
2 09 04 04 03 03
3 2 01 02 01 02
99
• If you have finished entering the number strings, and selecting the preferred trunks for the numbers,
configure the Number-based LCR using Jeeves.
• Enter the values of the Number-based LCR you prepared on the sheet of paper in the appropriate fields.
• Define the Time Zone(s) when the service providers offer lower tariff. You can define up to 8 time zones.
• For each Time Zone you define, specify the Number strings on which lower tariff is applied during that
Time Zone.
• For each Number string you enter for a particular time zone, assign Cost Factor. Select the trunk with the
lowest cost as your first preference. Select trunks of your second, third and fourth preference (in order of
increasing cost). Refer to the table you prepared for assigning Cost Factor.
When the trunk you selected as first preference is busy, the system will route the call through the next
trunk you set as preference if it is free.
For example, service provider of CO1 and CO2 (assigned Cost Factor 01 and 02) offers the lowest rate for
calls made to Area Code 022 between 8am to 12pm, followed by service providers of Mobile Trunk 2
(assigned cost factor 04) and Mobile Trunk 1 (assigned cost factor 03).
• Assign Cost Factor preference for the number string in this sequence: 01, 02, 04, 03
Number Index
• If you have finished defining the time zones, entering the number strings, and selecting the preferred
trunks for the number strings, configure the Number and Time-based LCR using Jeeves or a Telephone.
• Enter the values of the Time+Number-based LCR you prepared on the sheet of paper in the appropriate
fields.
01 3 080 3 05 06 01 02
02 6 022 3 01 01 02 04
: : :
99 2 03852 5 01 02 01 02
• As you can see, the Service Provider-based LCR Table is similar to the Number-based LCR table.
• You can configure as many as 99 different numbers which are stored against Index numbers from 01 to
10.
• The number strings may be the complete telephone number, a truncated phone number or the first digit of
the phone number.
• For each number string that you enter against an Index number, you must also specify the Area Code and
the Ignore Digit Count.
• The Ignore Digit Count is the number of digits in the area code that the system should ignore before
checking the Service Provider-based LCR table. For each area code that you enter, the corresponding
Ignore Digit Count will be the number of digits in the area code. For example, the area code for the number
starting with '3' is 080, which consists of 3 digits. So, the Ignore Digit Count for the number/area code 080
will be 3.
• For each number string and area code that you enter, assign the Trunk of the service provider who offers
the lowest tariff to that number/area code. Refer the table you prepared for assigning Cost Factor to trunks.
• If you have finished entering the number strings, their corresponding area codes and the Ignore Digit
Count, and the preferred trunks, configure Service Provider-based LCR using Jeeves.
• To configure Area Code and Ignore Digit Count, click the link Call Cost Calculation.
• Enter the Area Codes and the corresponding Ignore Digit Counts from the sheet you prepared for Service
Provider based-LCR. You may also enter the respective name for each area code, if desired.
• Now, click the link Least Cost Routing (LCR). The LCR options will appear below this link.
• Enter the values of the Service Provider-based LCR you prepared on the sheet of paper in the appropriate
fields.
• For the trunks you selected for Dialing the Trunk Access Codes 0, 61, 62, 63 and 64, select the desired
LCR Type from the combo box: Time-based, Number-based, Time+Number based, Service Provider-
based (Cost Factor).
Since you have already configured the LCR Type, you do not need to configure the LCR settings further on
this page.
What’s this?
This type of Least Cost Routing is used in countries where the same service provider offers local call and long
distance calling services. These service providers allow subscribers to select the service provider or Carrier for long
distance calling.
For example,
Service
Provider B
Service
Provider D
A subscriber of Service Provider A must grab trunk lines of Service Provider A to call other subscribers in the local
area.
However, when the subscriber of Service Provider A wants to make a long distance call, the subscriber must dial a
prefix to select the a carrier (trunk) of the desired long distance, Service Provider B, C and D. Thus, the subscriber
accesses a secondary service provider by dialing a short code or prefix to for long distance calling.
This feature works on the basis of “Automatic Number Translation”. Using Automatic Number Translation,
ETERNITY NE adds the code of the appropriate secondary Service Provider to the number string dialed by the
extension user to route the call to the desired secondary Service Provider.
How to configure
To use this feature, you must do the following:
• Enable Automatic Number Translation feature on the trunk—CO, Mobile or SIP—of your Primary Service
Provider.
• Select an Automatic Number Translation Table Number from 1 to 8 for the trunk.
• Configure the Automatic Number Translation Table. In the Automatic Number Translation Table, in the
Dialed Number String column, enter the long distance numbers that extension users will dial. In the
Substitute Number String column, enter the Prefixes of the Secondary Service Providers which should be
added to the dialed numbers. For example, the code ‘961’ for Service Provider B must be prefixed to the
number ‘2630555’ dialed by extension users, you must enter ‘2630555’ in the Dialed Number String
column and ‘9612630555’ in the Substitute Number String column.
See “Basic Settings” for instructions for configuring Automatic Number Translation on the different Trunk Port
types. See also, feature description for “Automatic Number Translation”.
What's this?
Certain features of ETERNITY NE require the purchase of a license. When you buy ETERNITY NE, you get a
unique license number for your product and a set of 'license-free' features are loaded in the system. Described
below are the features that require license.
SIP Extensions
ETERNITY NE supports up to 16 SIP Extensions. You can register SIP enabled devices like an IP Phone, a Soft
phone, Analog Telephone Adapter, as SIP extension of ETERNITY NE.
ETERNITY NE provides eight SIP extensions license-free. If you want to register more than eight SIP extensions,
you need to purchase a license.
You may activate your License Online. For this, keep the following items ready:
• The License Voucher containing the 16-digit PIN.
• A valid, unique User ID and Password from the Matrix License Support Centre.
• Access to Internet.
• Current License Key of the system.
• Open Jeeves.
• Keep your Current License Key and the License Voucher ready.
• Enter your User Name and Password provided by Matrix and click the Login button.
• In the field Current License Key, type the current product license key you noted from the License
Management page of Jeeves.
• Click Details. The details appear in the fields Product Family, Product Name, Product Variant.
• Click the Activate button and wait for a few seconds, as the activation is initiated.
On successful activation, the confirmation message will appear on your screen along with the activation
date and time.
You will also be sent a confirmation mail to your e-mail ID (registered with Matrix).
• Go back to the Jeeves window (or log in as System Engineer again, if your session has ended).
• Enter the new License Key generated in the field Enter License Key.
When you activate your IP8 License to enable 8 IP Users, it will appear on your Service Profile
If you are unable to use Online Activation of the License Key or have no internet access, contact the Matrix
License Support Centre for assistance in generating the new License key.
• Open Jeeves.
• Click the License Management link. The License Management page opens.
You may view the features and functions that are currently available to you under Service Profile.
• Send your Current License Key and the License PIN (on the Voucher) to the Matrix License Support
Centre.
• Enter the New License Key you obtained from Matrix in the field Enter License Key.
When the system is defaulted or upgraded, the current License key and Service Profile will be saved.
What’s this?
Live Call Screening enables extension users to screen callers before attending their calls on their extensions.
How it works
To be able to use this feature,
With the above pre-requisites fulfilled, this is how Live Call Supervision will work:
• B calls A, and is transferred to A’s mailbox (as Call Forward to Voice Mail System is set).
• The VMS Auto Attendant offers B the option to leave a message in the mailbox of extension A.
• As B starts to record a message in A’s mailbox, the speaker of the DKP/Extended IP Phone of A gets
turned ON for the duration of the Live Call Screening Timer (configurable; default: 10 seconds).
• If A wants to answer the call, A can go Off-Hook. A gets connected to the caller and the system stops
recording the message in the mailbox.
• If A does not answer the call, the speaker is turned Off automatically on the expiry of the Live Call
Screening Timer.
• Also, after listening to some part of the message, if A finds that the call is not important, A can ignore the
caller by dialing any digit. When A dials any digit, the speaker of the DKP/Extended IP Phone is turned
OFF, but the message recording in the mailbox continues.
See “Basic Settings” for instructions on configuring the Voice Mail Settings and Class of Service of different
extension port types: “SLT Extensions”, “DKP Extensions”, “SIP Extensions”.
If required, you may also change the duration of the Live Call Screening Timer. See “System Timers and Counts”
for instructions.
What’s this?
Using Live Call Supervision, any extension can know the last external number dialed by another extension, even
when that extension is in speech with an external party.
This feature is useful for supervisors who want to know where their subordinates are calling.
This feature is supported on DKP and Extended IP Phone extensions, and on SLT extension which have CLI
phone.
How it works
• A is the supervisor of B.
• When A requests Live Call Supervision for B’s extension, the system retrieves the last external number
dialed by B and presents it on the display of A’s phone.
• If the last number dialed by B is an internal number, A will get error tone, as the system supports live call
supervision of external calls only.
Live Call Supervision can be used also when the extension being supervised is in speech with an external
party.
See “Basic Settings” for instructions on configuring the different extension port types: “SLT Extensions”, “DKP
Extensions”, “SIP Extensions”.
How to use
For EON & Extended IP Phone Users
What's this?
Logical Partitioning is used to restrict the flow of call traffic between Public and Private networks and between
Public Switched networks and VoIP networks.
This feature may be used in countries where such restrictions are mandated by telecom regulations. For example,
in certain countries, calls from VoIP to Public Networks (CO, PSTN, Public Land Mobile Network) are not allowed.
Local telecom regulations may either disallow termination of lines from both networks on the same equipment or
may allow lines from both networks to be terminated on the same equipment, provided the equipment is designed
to restrict flow of call traffic from these networks. For example, the Telecom Regulatory Authority of India allows
termination of lines of the PSTN and VoIP Networks in the same equipment, only if these lines are logically
partitioned. Termination of lines from both these networks in the same equipment without a logical partition
constitutes an offence in India.
ETERNITY NE supports Logical Partitioning for this purpose. You can allow or restrict calls between networks by
applying Logical Partitioning on the system.
How to configure
• Log into Jeeves as System Engineer.
Now, allow or restrict the calls between the different Trunk Port types.
• By default for countries other than UK, USA and Italy calls within and between all types of trunks are
restricted. In other words, trunk to trunk calls are not allowed.
• In UK, USA and Italy calls within all types of trunks, including VoIP are allowed.
• When call permission is restricted between trunks and/or between a type of trunk to VoIP, the following
feature interactions will apply:
• Call Transfer: Trunk to Trunk Transfer between restricted categories of trunk will not be allowed. If
the user attempts trunk-to-trunk transfer between restricted trunks, Error Tone will be played.
• Raid: If a user using DISA attempts to Raid a conversation of an extension with a trunk to which call
permission is restricted, the Raid attempt will fail and the user will get an Error Tone.
• Conference: An extension user will not be able to include restricted trunks in a 3 party or multi-
party conference. An Error Tone will be played when s/he attempts it.
• Dial-In Conference: Participation in a Dial-In Conference from trunks with restricted call permission
is not allowed.
• External Call Forward: In the case of DID, DISA or when transferring a trunk call to an extension,
if the extension has set call forward to an external number, the system will allow the call only if the
call permission between the source and destination trunk is allowed. Otherwise, Call Forwarding
will not be applied.
• Hotline: When a user has logged into DISA and the extension being used for the DISA login has
the Hotline - Trunk or Hot outward dialing (HOD) feature enabled, the system will allow the call
between the source trunk (from where the DISA login is made) and the destination trunk (which is
used as Hotline Trunk) only if calling is permitted between them. Otherwise an Error Tone will be
played to the DISA caller on the expiry of the Hotline Timer.
What’s this?
Extension users often have to dial access codes for specific functions like dialing a feature code, making an internal
call, making an external call, etc.
ETERNITY NE supports Macros, using which, you can abbreviate long number strings for regularly used functions
in to macros and assign them to a DSS key on a DKP/Extended IP Phone extension.
You can also assign Macros on SLTs that have special keys.
How it works
• Extension 201, frequently sets Call Forward-All Calls to an external number 26550333.
• To do this, each time, Extension 201 must dial 131-Trunk Access Code-26550333-#*.
• Instead of having to dial this lengthy number string, a Macro can be created for Call Forward-All Calls to
External number.
• If Extension 201 is an Extended SIP Phone or a DKP, the Macro can be assigned to a DSS key on the
phone.
• Instead of dialing this number string, the user of Extension 201 can simply press the DSS key on which this
Macro is assigned.
• Thus when the DSS key on which a Macro is assigned is pressed, the corresponding access code is
executed.
ETERNITY also supports Macros for SLT which have special keys. When each of these keys is pressed, a special
number string, which you can program is dialed.
For example, an SLT instrument has 5 special purpose keys. When these keys are pressed, the strings *50, *51,
*52, *53, *54 programmed on these keys are dialed out.
You can create Macros for the strings dialed out using the special keys, whereby the string dialed by each of these
special keys is associated with a particular function. For example, the special key for dialing *50 is associated with
Call Forward -All Calls to an external number. So, when the extension user presses *50, the system receives this
string and takes appropriate action, i.e. interprets it as call forward to the external number, and sets call forward.
Thus, each time the extension user presses the special key *50, the system considers that the extension user has
dialed 131-Trunk Access Code-26550333-#*.
To configure Macros,
• Open Jives.
The Macros page opens. Each macro is stored against an index number.
• In the Number String field, enter the strings the system should consider as command when the DSS
Key on the DKP/Extended IP Phone.
For detailed instructions on assigning a keyboard macro to a DSS key, see “Phone Key Settings” under
“DKP Extensions” for instructions. Also see, “Matrix Extended IP Phone Key Settings” under “SIP
Extensions”
• In the Number String field, enter the strings the system should consider as command when the special
key on the SLT is pressed.
• In the Access Codes field, enter the strings sent by the SLT on pressing the special function key.
For example, if the SLT sends the string ‘*53’ to the ETERNITY NE when the function key for Alarms is
pressed, enter the string 163 (the feature access code for Voice-guided Alarms) in the Number String
field, and enter the string *53 in the corresponding Access code field.
The Access Code that you assign here in Macros must not conflict with any other Access Codes in the
Dial Phase. See “Access Codes”.
What’s this?
Using Meet Me Paging, extension users can announce the name of the person they want to talk to on the Page
zone. The called person can respond to the announcement, by dialing, from any extension, the Meet Me Paging
code and the extension number of the caller.
When making a Meet Me Paging call, extension users can announce the extension number from where they are
calling, so that the called person knows which extension number to dial.
This feature is of great use to Operators. Using this feature, the Operator can put an important call on hold and
track the extension user who is not at the desk using Meet Me Paging. Once the person is located and calls the
Operator, the Operator can transfer the call to the person at the current location.
How it works
• The Operator receives a call for B.
• Operator calls B’s extension, but B is not at the desk.
• Operator uses Paging and makes the announcement for B, asking B to call the Operator.
• B is near extension 201. B dials Meet Me Paging Code and dials the extension number of the Operator.
• B is in speech with the Operator.
How to configure
To be able to use Meet Me Paging, extension users must have the feature“Paging” in their Class of Service.
If you are the Called Party (for whom the message has been announced):
• Lift the handset.
• Press the DSS key assigned to Meet Me Paging.
OR
• Dial 1093
• Dial the number of the paging extension.
• Speak with the paging extension user.
If you are the Called Party (for whom the message has been announced):
What’s this?
The Message Wait feature of ETERNITY NE enables extension users/Operator to set Message Wait on other
extensions to deliver important messages.
If the extension user has a mailbox assigned, the Message Wait feature indicates to the extension user, the arrival
of new messages in the user’s mailbox.
Thus, Message Wait can be set by extension users as well as by the Voice Mail System.
How it works
• The Operator has an important message to communicate, so the Operator sets Message Wait on
Extension A, using the Message Wait key (if configured) or by dialing the feature access code.
• Extension B tries to reach Extension A, and sets Message Wait on Extension A, using the Message Wait
key (if configured) or by dialing the feature access code.
• Message Wait will be indicated to Extension A according to the Type of Message Wait Notification set for
Extension A. This may be in the form of a Stuttered Dial Tone, a Voice Message, Ring, or LED Lamp.
• If Extension A is a DKP or an Extended IP Phone and has DSS key assigned for Retrieve New Message,
the LED of this key will glow to indicate new message wait.
• Now, Extension A can dial the feature access code to retrieve Message Wait, or press the Retrieve
Message Wait Key, if assigned.
• The system will call the extension that first set Message Wait on Extension A. In this case, the Operator. If
the Operator is busy, the system will place the call on Extension B. The system will try to call the
extensions that set Message Wait until the call is answered.
• The extension that set Message Wait on A gets the CLI of A as Message Wait. A can now deliver the
message.
• The LED of the Retrieve Message Wait key, if assigned, on Extension A will be turned off after all message
wait set by other extensions on Extension A have been served.
• There is a new message in A’s Mailbox. The VMS indicates this to Extension A as per the Type of
Message Wait Notification set for Extension A. This may be in the form of a Stuttered Dial Tone, a Voice
Message, Ring or LED Lamp.
• If Extension A is a DKP or an Extended IP Phone, the Voice Mail key on the phone will also glow to
indicate the arrival of a new message.
• If the Retrieve Message Wait key is assigned on the DKP/Extended IP Phone of Extension A, the LED of
this key will also glow simultaneously to indicate arrival of the new voice mail.
The VMS answers the call. After Extension A has listened to the new message(s), the LED of the Voice
Mail key is turned off.
The LED of the Retrieve Message Wait key, if assigned, will also be turned off.
Voice Mail has priority over extension Message Wait set by extensions. If an extension has both
Message Wait and new Voice mail, and when the extension presses the Retrieve Message Wait key or
dials the feature access code to Retrieve Message Wait, the call will be placed first to the Voice Mail
System. When the extension user presses the Retrieve Message Wait key again, the call will be placed to
the extension that set Message Wait.
• If you want voice message to be played as message wait notification, record and assign a Voice Module.
Refer “Voice Message Applications” for instructions.
ETERNITY can play only 4 Voice Modules simultaneously. The Voice Module for Message Wait
Notification will not be played if there are already 4 being played simultaneously. In which case, Stuttered
Dial Tone will be played for Message Wait Notification, when the extension user goes OFF-Hook.
Ring
• When a new Message Wait is set on the extension, the system will play Message Wait Ring (Short, Fast)
on the extension. See “Distinctive Rings”.
• The extension will ring for the duration of the Message Wait Ring Timer (configurable; default: 30
seconds). If the call is not answered within this timer, the system will ring on the extension again for as
many times as the Message Wait Ring Count (configurable; default: 10 times), and at the interval set as
the Message Wait Ring Timer Interval (configurable; default: 30 minutes).
• When the extension user answers the call, the user gets connected to the VMS or the extension that set
Message Wait.
How to configure
To provide this feature to extensions, you must do the following configuration on the extensions:
• Enable the Message Wait feature in the “Class of Service (COS)” of extensions. This allows the extensions
to set and cancel Message Wait on other extensions. Only those extensions that have this feature in their
COS can set or cancel Message Wait on other extensions. See “Basic Settings” for instructions on
configuring the COS of different Extension types: SLT, DKP, SIP. By default, this feature is enabled in the
COS of all extension types for the Day and Night.
• If you selected Voice Message as Voice mail/Message Wait Notification Type for an extension, you must
also record the desired Voice Message in a Voice Module and assign it to the Message Wait application.
See “Voice Message Applications” for instructions.
• If you selected Ring as Voice mail/Message Wait Notification Type for an extension, you may configure
the following Ring Parameters:
• Message Wait Ring Timer (default: 30 seconds)
• Message Wait Ring Count (default: 10 attempts)
• Message Wait Ring Timer Interval (default: 30 minutes).
See “System Timers and Counts” for instructions.
How to use
For SLT
What's this?
ETERNITY NE offers mobility to its extension users whose nature of work keeps them away from their desks
frequently and for longer durations.
Using mobility extensions, the extension users of ETERNITY NE can make and receive their calls from their current
(remote) location, placing calls through the system, and can access the system just as any other normal extension
of the ETERNITY.
How it works
ETERNITY supports two types of users:
• Station Users: they are extension users of ETERNITY NE to whom a dedicated physical extension is
provided on their desks.
• Virtual Users: they are extension users of ETERNITYNE who share a physical extension, or may not
have any physical extension allotted to them.
The facility of Mobility Extension is provided to both virtual and normal extension users using the features
“Direct Inward System Access (DISA)” and “Call Forward”.
How to configure
To provide Mobility Extension to users, the follow the steps described below.
• List out the Extension Users and Virtual Users and configure them first.
• Make sure that extensions which are to be provided Mobility Extensions have the features Call Forward
and DISA enabled in their “Class of Service (COS)”.
• Make a list of External numbers to which the Mobility Extension users will forward their calls. Configure
these numbers in the 'Allowed List' of Local, Regional, National and International Numbers, as appropriate.
• The Toll Control assigned to the extension will be applied when a call is forwarded to an external number.
Make sure that the extensions which are to be provided Mobility Extension have the required “Toll Control”
level for Call Forward to the External Numbers (the numbers you have programmed in the Allowed List).
Toll Control level is to be programmed on the Mobility Extensions.
• Configure the parameter 'External Call Forward for' on the Mobility Extensions. This parameter defines the
types of calls for which the External Call Forwarding is to be applied. Select any one of the options, Internal
Calls Only, Trunk Calls Only, Internal + Trunk Calls as required.
• Program the parameter 'DISA' on the trunk lines on which Mobility Extensions users are to be provided
access to. Make sure you select the option 'CLI Auth. Multiple Calls' in the 'DISA' parameter of the trunk
port.
Program this list of numbers in the 'Calling Number' field of the Authentication Table. Program the 'Port
Type' and 'Port Number' of the Extension assigned to Mobility Extension Users in the 'Auto Login As' field
for the respective 'Calling Number' field. Refer the topic “Direct Inward System Access (DISA)” to know
more.
How to use
The Mobility Extension Users of ETERNITY NE can use its features from a remote location as described below.
Receiving calls
To receive calls, the Mobility Extension User must set Call Forward on his extension with an external number
(mobile number, landline number, etc.) as the destination number.
To make calls ring on the extension and the external number simultaneously, the Mobility Extension Users must
activate the Call Forward-Dual Ring feature on their extensions.
The Mobility Extension Users can also choose where they want to receive the calls during a particular time of the
day. For example, they can receive calls during a particular time of the day, i.e. Time Zone on their external number
and have their calls received by their Voice mail or the Operator or any other number during another Time Zone. To
do this, they must set “Call Forward-Scheduled" on their extension. Dual Ring can also be set for Call Forward-
Scheduled.
Making calls
The Mobility Extension User should make a call on the DISA enabled trunk of ETERNITY from the external number
and the system will provide the dial tone to the user after authenticating the external number with the help of the
DISA-CLI Authentication table.
On getting the dial tone, the Mobility Extension User can make internal as well as external calls as per the “Toll
Control” and “Class of Service (COS)” assigned to his Extension.
The Mobility Extension User can also dial codes of the Personal directory and Global directory numbers to use the
feature Abbreviated Dialing.
Accessing Features
Mobility Extension Users can access the system features by dialing specific codes after making calls on the DISA
enabled trunk of ETERNITY, or after answering the calls received on their external number.
On-Hook #0
Off-Hook #1
Flash #2
Pause #3
Described below are instructions for Mobility Extension users on using different call management features.
Call Hold
To put a call on hold,
Call Transfer
To conduct a screened Call Transfer,
OR
OR
OR
Call Pickup
To pick up the call of same Call Pickup-Group,
Multiparty Conference
To create a Multiparty Conference,
Call Forward-Unconditional
To set Call Forward-Unconditionally,
Call Forward-Busy
To set Call Forward-Busy,
Call Forward-Scheduled
To set/cancel Call Forward-Scheduled,
• Mobility Extension users can have Call Forward and Call Forward-Scheduled set on their extension by
the Operator or by another extension user.
• Using “Call Forward-Remote” and by setting “Call Forward-Scheduled” from the SA mode (using SA
commands or Jeeves), the extension user/Operator can set Call Forward and Call Forward-Scheduled
for any Mobility extension user. Refer the respective topics to know more.
What’s this?
The music played to extension users and external callers who are put on hold is called Music on Hold (MoH).
How it works
A Voice Module assigned the Music on Hold application serves as the source of music for this feature. By default,
Voice Module 1 is assigned for Music on Hold.
When a caller is put on hold, ETERNITY NE plays the music recorded in Voice Module 1.
You can play a voice message of your choice instead of music to the callers. The message may contain any
promotional information about your company or services provided by your organization, etc.
Music-on-Hold will also be played to an extension, when the option Route to Operator is selected as the
Alarm Notification Type for the extension, and the Operator extension is busy, when the extension goes
Off-hook to answer the alarm call.
How to configure
You can record a piece of music or a message of maximum 16 seconds duration and assign it to Voice Module 01,
which is reserved for Music-on-Hold.
Refer the topic “Voice Message Applications” for instructions on recording and assigning voice modules.
What's this?
The Multi-Stage Dialing feature of ETERNITY is typically used in applications like Calling Card, where extension
users are required to dial digits in stages when making a call using the calling card.
The Multi-Stage feature enables extension users to directly dial the number they want to call, and the system dials
out the number at different stages of the call by suitably modifying the number.
How it works
A typical example of Multi-Stage Dialing is the use of prepaid Calling Cards. Here, the person using a calling card
must dial a fixed number string before dialing the actual number. When using a calling card,
• Users must first dial the number of the Calling Card server, for example: 1602233 (7 digits).
• After the call is answered by the Calling Card server, users must dial the PIN provided by the calling card
service provider, for example 1132121234.
• After dialing the PIN number, users can dial the number they want to call, for example 0014125126508.
Thus, when using a calling card, users must dial a very lengthy number string, each time they need to make a call
using the calling card.
The use of Multi-Stage Dialing saves the time and effort of dialing out lengthy digits in stages.
The Multi-Stage Dialing makes use of the “Automatic Number Translation” table. This table must be configured on
the trunk from which extension users will use to make calls using Calling Cards.
• Extension users are to be allowed to make international calls using Calling Card from the trunk CO1.
“Automatic Number Translation” table must be configured on CO1 trunk.
• The Automatic Number Translation table consist of Dialed Number Strings and Substitute Number strings.
• In the Dialed Number String, you must configure ‘00’, the prefix for international numbers.
• In the Substitute you must configure the Calling Card server number and the PIN Number.
• As the system must wait for the Calling Card server to answer before dialing the PIN, you must configure
Wait for Answer between the Calling Card server number and the PIN number.
You must insert a delay by configuring the Pause Timer after the PIN number and the destination number,
i.e. the prefix ‘00’.
1 00 1602233W1132121234P00
32
• When the Automatic Number Translation table is configured, the Extension user can simply dial the Trunk
Access code and the destination number: 0/5/61/62/63/64 - 0014125126508.
• The system matches the dialed number with the Dialed Number String of the ANT table, the number
matches with the entry ‘00’ stored in the table.
• The system dials the Substitute Number string 1602233. It waits for the calling card server to answer the
call.
• When the call is matured, i.e. the calling card server has answered the call, the system dials the PIN
number 1132121234 and waits for the Pause Timer before dialing the destination ‘00’.
Thus, the extension user directly dials only the desired destination number, the system substitutes this
number by adding the Calling Card server number and PIN number and dials these numbers in two stages.
How to configure
To be able to use Multi-Stage Dialing, you must configure the following:
• Automatic Number Translation table on the trunk you want to use this feature. For instructions on
configuring ANT on different Trunk types, see “CO Trunks”, “Mobile Trunks”, “SIP Trunks” under Basic
Settings.
For instructions on configuring DTMF Out Dial on different Trunk types, see “CO Trunks”, “Mobile Trunks”,
“SIP Trunks” under Basic Settings.
• If required you may configure the Call Proceeding Tone for Multi-stage Dialing as Network tone,
Pseudo Tone, or Silent. For instructions, see “System Parameters”.
What’s this?
This feature helps the extension user to disconnect the speech transmission path in the middle of a conversation.
The extension user can still listen to the opposite party because the receiving path remains connected. Mute is
useful when you want to consult someone in the middle of a conversation, but do not want the opposite party to
listen to your discussion. You can Mute a call before making a call or during speech.
How it works
• A is in speech with B.
• A wants to consult to C in the room, but does not want B to hear their conversation.
• A presses the Mute Key.
• The transmit speech path from A to B is disconnected. The receive path remains connected.
• So, A will be able to hear B, but B will not be able to hear the conversation between A and C.
• When A has finished consulting C, to resume speech with B, A presses the Mute key again.
• The transmit speech path from A to B is restored. A and B are in speech again.
How to use
This parameter is related to the features “Toll Control” and the “Dynamic Lock”. Toll Control allows you to define a
particular calling permission for each extension, referred to as 'Call Privilege'. A Call Privilege allows the extension
to call certain areas and restricts it from calling others. The extension can also be restricted from the dialing of
specific telephone numbers.
If you want to use Toll Control, define the Local, Regional, National, and any other specific number strings which
the system should allow to be dialed and restrict from being dialed. For this, you need to configure the Number
Patterns, for Local, Regional, National and Limited Calls.
• The Call Privileges 'No Calls' and All Calls’ do not require any number pattern to be configured.
• The Call Privilege Type 'Limited Calls' allows the dialing of specific telephone numbers only.
• Open Jeeves.
• Define Local Calls number strings. You may enter up to 999 local numbers.
• In the column Allowed Numbers, enter the number strings to be allowed to be dialed out by the system.
You may enter only the first digit of the number string, or a part of the string, or the complete number string.
• In the column Denied Numbers, enter the number strings which you want to restrict from being dialed out.
• Click Submit.
• Enter as many local numbers as you want by clicking the Index number links 001-250, 251-500, 500-750,
751-999.
• Enter the Allowed Numbers and Denied Numbers in their respective columns.
• Click Submit.
• Enter as many Regional numbers as you want by clicking the links 001-250, 251-500, 500-750, 751-999.
National Calls
• Click the National tab.
• Enter the Allowed Numbers and Denied Numbers in their respective columns.
• Click Submit.
• Enter as many National numbers as you want by clicking the Index number links 001-250, 251-500, 500-
750, 751-999.
Limited Calls
The Call Privilege Type 'Limited Calls' allows the dialing of specific telephone numbers, which may be local,
regional, national or international numbers.
• Enter the Allowed Numbers and Denied Numbers in their respective columns.
• Enter as many Regional, National numbers as you want, by clicking the Index number links 001-250, 251-
500, 500-750, 751-999.
• To configure another Limited Number List, click the tab Limited 2, Limited 3 and follow the same
instructions as above.
What's this?
When the handset of an extension is not placed correctly, it will not be possible for the Operator or any other
extension to call the extension. Also, incoming calls will not reach the extension, Alarms and Reminders cannot be
placed on that extension.
To avoid this inconvenience, the ETERNITY NE supports the feature 'OFF-Hook Alert', whereby the system detects
and informs the Operator of the extension phone that remains OFF-Hook accidentally.
How it works
To give the Operator an OFF-Hook Alert,
• When the Operator answers the call, s/he is played a confirmation tone, the text message remains on the
display.
• The Operator can inform the extension user to place the handset of the phone correctly.
• If the extension phone is an SLT, OFF-Hook Alert will be given to the Operator phone only. (The Operator
phone must be a DKP or the Extended IP Phone).
• If the extension phone is a DKP/Extended IP Phone, OFF-Hook Alert will be give to both, the Operator
phone and the extension phone.
• If the extension phone that is OFF-Hook is DKP/Extended IP Phone, the ETERNITY NE will activate 'OFF-
Hook Alert' on the extension phone, by playing the Error Tone continuously, on speaker to draw the
attention of the extension user.
• While the Error Tone for OFF-Hook Alert is being played on the extension phone, if the user presses the
Speaker Key, the Error Tone will continue to be played on the handset until it is replaced correctly.
How to configure
For this feature to work, the OFF-Hook Alert to Operator flag must be enabled in the “System Parameters”.
To do this,
• Open Jeeves.
• Go to the parameter Give OFF-Hook Alert to Operator, select the check box to enable the flag.
What's this?
Paging allows you to make announcements to a group of extension users by simply lifting the handset of your
extension phone and dialing a code.
This feature is useful when you want to call several people at once, for example, to inform them about a meeting
you have scheduled. If the persons you want to call have Digital Key Phones (DKP) extensions or the Matrix
Extended IP Phone, or any Open SIP Phone as their extensions, you can use paging instead of calling them up
one by one.
• You can start paging from SLT, DKP or any SIP Extension. However, Paging is possible only on DKP
and SIP Extensions, which may be Extended IP Phones or Open SIP Phones. The Open SIP Phones
on which you are paging must support Call-Info or Alert-Info header for Paging.
• The DKP/SIP extensions being paged can only hear the person who is paging as the speakerphone
mic is not activated on the DKP/IP Phone.
How it works
The Pre-requisites
• Page Zones must be created. You can create 12 different Page Zones.
• The DKP and SIP extensions which are to be paged must be included in the Page Zones. You can create
12 different Page Zones. Each Page Zone accommodates up to 16 DKP and SIP extensions.
• Paging must be enabled in the Class of Service allowed to the DKP, SLT, SIP extension from which this
feature is to be used.
The Process
• A DKP/SLT/SIP extension user having Paging in the Class of Service, dials the Access code for Paging
and the Number of the Page Zone to which s/he wants to make the announcement.
• The system checks hunts DKPs and IP Phones in the Page Zone Number which are free and activates the
speakerphones of these DKPs/SIP Phones in the Page Zone number.
• All DKP and SIP extensions in the Page Zone can hear the announcement. But they cannot speak to the
calling DKP/SLT/SIP extension user, as the mic of their speakerphones is not activated.
For each Page Zone number, decide and assign the DKP and SIP extensions.
On a sheet of paper create a two-column table for each page zone, as shown below. Write the numbers of the DKP
and SIP extensions you want to include in the page zone in the member column.
Page Zone 1
Index Member
1 DKP-1 (301)
2 DKP-2 (302)
16
Page Zone 2
Index Member
1 DKP-1 (301)
16
• For each Index number, select a Member extension. Select the extension number you want to include in
this Page Zone. Default: None.
• To configure another Page Zone, click the Page Zone number tab.
• If you have finished configuring Page Zones, you may log out.
How to use
It is possible to page from a DKP, SLT, and SIP Extensions.
What’s this?
Making an IP call without the intervention of a proxy server is called Peer-to-Peer Calling. As Peer-to-Peer calling
does not require a proxy server, voice communication using this application can be done virtually free of cost. The
major cost savings offered by this application makes it a very attractive mode of inter-branch or intra-office voice
communication.
How it works
Let us understand how to use Peer-to-Peer Calling with the following illustration:
Location A Location B
201
SIP1
301
SIP2
202
SIP3 115.118.161.165 121.124.130.110 302
203 Ethernet
WAN IP WAN
303
Broadband Broadband
204 Router/Modem Router/Modem
304
ETERNITY PBX A
205
305
Menu Sat 01 05: 30
206 FwdBus
y FwdNR Can
cel Mute Con
f erenc
e Trans
f er
306
DND VoiceMail Names Red
i al Rejetc Hold
1 2 abc 3 def
4 ghi 5 k
jl 6 mno
SI P2 CA2
SI P1 CA1
• Peer-to-Peer calls can be made between the two locations with suitable configuration of ETERNITY and
the PBX.
• Select a SIP trunk to be used for this application and enable it. For example, SIP1.
• Set the Treat Incoming Peer-to-Peer call as option on the SIP trunk to Station.
• Set the Send CLI option on the SIP trunk as Calling Party Wise.
• You may also configure the Closed User Group (CUG) Table to avoid dialing the Trunk Access Code
for the outgoing SIP Trunk, i.e. SIP 1.
• In the Route Code field, enter the extension numbers of the PBX at Location B. Instead of entire
number strings, you can configure a single digit, the starting digit of the extension numbers as
Route Code. In this case, you may configure Route Code as ‘3’, as all extensions at Location B start
with ‘3’.
• In the Dialed Digit Count field, enter the digit length of the extension numbers at Location B. In this
case, ‘3’.
• In the Route using Trunks field, select the SIP trunk number to be used for routing the Peer-to-
Peer calls. In this case, SIP 1.
The CUG Table you configure on ETERNITY would look like this:
Strip Dialed
Route Index Route Code Digit Self Route Digit LCR Route using Trunks
Count Count
The Peer-to-Peer table stores up to 1000 entries. Each entry consists of the parameters Number,
Domain Address and Name.
• In the Number column of the Peer-to-Peer table, enter the digit you configured as Route Code in
the CUG Table for calling the extension numbers of the PBX at Location B. In this case ‘3’.
• For the numbers you entered, in the Domain Address field in the table, enter the IP Address of the
WAN Port of the PBX at Location B.
• For the numbers you entered, in the Name field in the table, enter any desired name you want to
assign as an identification or tag.
Prudent
002 3 121.124.130.110
Mumbai
003
• With the necessary configuration done at both locations, when an extension user 201 of ETERNITY at
Location A dials 301, the number of an extension of PBX A, the system checks the CUG table to match the
dialed digits with the Route Code and the Dialed Digit Count. As a match is found, it selects the SIP trunk
defined for routing the Route Code, i.e. SIP 1.
• As SIP 1 is set to Peer-to-Peer mode, the system checks the Peer-to-Peer Table configured. It finds a
match for the digit ‘3’ and places the call to the IP Address 121.124.130.1 configured for this number.
How to configure
For instructions on configuring the SIP Trunk parameters for the Peer-to-Peer application—SIP Trunk Mode, Peer-
to-Peer Table, SIP ID, Treat incoming Peer-to-Peer calls as Station, Send CLI—see “SIP Trunks”.
For instructions on configuring the CUG Table, see “Closed User Group (CUG)”.
• Open Jeeves.
• In the Number field, enter the number string—prefix or entire number—that will be dialed. In this case,
it would be the Route Code in the CUG Table for calling the remote extension numbers. The number
string must not exceed 24 characters. Default: Blank.
• In the Domain Address field, enter the domain name or IP Address to where the call is to be placed.
The Address may consists of up 40 characters (maximum). Default: Blank.
The Domain Address can also be in the form of Address: Port number.
• In the Name field, enter a name to identify the number string you configured. It may be the name of
your contact or any name you wish to assign to the number string. The name may consist of 24
characters (maximum). Default: Blank.
The name you configure here will not be used in SIP signaling.
What's this?
Extension users may want to indicate their availability to callers from other extensions.
For example, an extension user may want to leave his desk for an indefinite period, but does not want to use Call
Forward or set Do Not Disturb. He wants to indicate to callers about his absence. Similarly, extension users who
are present at their desk may want to hide their presence from other users; or they may want to show their current
activity to the other extension users like they are Busy, or are away from their desks, or on the phone with someone
on another call, etc.
With the Presence feature of ETERNITY NE, extension users, including the Operator, can 'publish' their presence
to callers from other extensions. By doing so, they can indicate to the other extensions about their availability.
In the same way, the Presence feature allows extension users to view the 'Presence' status (availability) of the
extensions that they want to call, before making the call or when their call is not answered.
How it works
Publishing Presence
Any SLT, DKP, SIP Extension User can 'publish' their presence by setting any of the messages listed in the
following on their phone, by dialing the access code for this feature.
SIP Extension users who want to publish their presence have two options:
• Using the PUBLISH feature supported by the SIP Client.
• Using the feature access code for Publish Presence supported by ETERNITY NE.
The first option requires the parameter 'PUBLISH' to be enabled in the SIP Extension Settings. Refer
“Configuring SIP Extensions”. By default, this parameter is disabled.
When any other DKP extension user calls this extension, the text message 'User Absent' will appear on
the caller's phone display.
If the extension phone that has set 'Absent' is a DKP, the letter 'A' appears on the phone's display to
indicate absence.
The letter 'A' disappears when the extension user sets a presence message other than 'Absent'.
As 'Absent blocks incoming external as well as internal calls, it can be used as an extension of the “Do Not
Disturb (DND)” feature, which blocks only internal incoming calls.
• External callers who call the extension, on which 'Absent' is set, will get an error tone only.
• Outgoing calls can be made from the extension which has set 'Absent'. Only incoming calls are
restricted.
• If more than one extension is configured as "Operator" (routing group), incoming calls will be blocked
only on the Operator extension which has set User Absent.
1. Present:
When an extension user sets 'Present', all incoming calls will be received as normal on this extension.
If previously set as 'Absent', when a DKP extension user sets 'Present' the letter 'A' will disappear from the
phone's display.
When any other DKP extension user calls this extension, the name of the extension user will be displayed
on the caller's phone display, when the called extension is ringing.
2. Auto Detect: When an extension user sets 'Auto Detect', the system will detect the state of the phone;
depending on the call state, it will publish the presence message to the other extensions. Three types
Publish Presence messages are possible, with Auto Detect:
a. Idle: When the system detects the extension phone to be ON-Hook, it indicates the status of the phone
to other extensions 'idle'.
b. On the Phone: When the system detects the phone to be OFF-Hook, or in speech with another party
or if it detects an incoming call placed on the phone, it will indicate to the other extensions that this
extension user is 'On the Phone' with another party.
c. DND Text message: When the system detects that the extension phone has Do Not Disturb (DND) set
on it with a DND Text message, it will display to the calling extension, the DND message set by the
called party (this may be the default DND message or the DND Text message set by the called
extension).
3. Away: When an extension user sets 'Away', the system will display this message to the other extensions.
4. On the Phone: When an extension user sets 'On the Phone', the system will display this message to the
other extensions.
5. Do Not Disturb: The extension user can set this message to be published to other extensions, if s/he
wants to work uninterrupted.
Unlike the DND Feature, the extension user who has set this message will continue to receive calls both
internal as well as external calls, as the system considers this extension as 'present'.
6. I am Mobile: The extension user can set this message to be displayed to other extensions, when s/he is
not at the desk.
8. Out for Meal: The extension user can set this message to be displayed to other extensions when going on
a lunch break.
9. Out of Office: The extension user can set this message to be displayed to the callers when s/he leaves
the office temporarily.
When an extension user sets any Publish Presence message other than 'Absent', the system will consider
the user as 'Present'. All incoming and outgoing calls will be allowed on this extension.
It is possible to program another message in place of Publish Presence messages listed from 6 to 9: I am
Mobile, In a Meeting, Out for Meal, Out of Office.
Publish Presence messages can be set or changed for any extension from the System Administrator (SA)
mode.
Viewing Presence
• Extension users can know the status of another extension user before calling or when the extension user
does not answer the call.
• Generally, when DKP extension users call another extension, the name of the called extension is
displayed on the calling DKP extension. Now, if the flag 'Display User Status during Call' is enabled in the
System Parameters, when DKP extension users call another extension, the calling DKP extensions will be
displayed the 'presence' status message published by the called extension58.
• SLT extension users, whose phone is equipped with a CLI display, can see the status of another extension
by dialing a feature access code, then going ON-Hook. The system will ring back the SLT and send the
Presence status of the desired extension as CLI.
• SIP extension users can use the Presence feature of ETERNITY to view the presence status of other
extensions. For this, they must dial the feature access code and the number of the desired extension.
• SIP extension users who want to view the status of other extensions using the feature supported by their
SIP Client, must have 'Presence Subscription' enabled in their SIP Extension Settings. Refer “Configuring
SIP Extensions”.
How to configure
This feature involves the configuration of the following parameters:
• 'Display User Status during Call' flag: DKP extension users will be able to view the presence status for
the called extension only if this flag is enabled in the System Parameters.
• PUBLISH: SIP extension users who want to publish their presence using the feature supported by their
SIP client will be able to publish their presence status only if this feature is enabled in their SIP Extension
Settings. This parameter is not necessary, if they want to publish presence using the feature of ETERNITY.
58. DKP users can also dial a feature access code and the number of the extension to see the status of that extension on their DKP.
But this would not be required, if the 'Display User Status during Call' flag is enabled in the System Parameters.
• Publish Messages: It is possible to customize the Publish Messages listed above from 6 to 9 viz.: 'I am
Mobile', 'In Meeting', 'Out for Meal', 'Out of Office'.
• Go to 'Display Presence Status during Call on DKP'. Click to enable the flag.
• You can change message number 6 to 9 as desired. The string may consist of a maximum of 16
characters.
By default both the features PUBLISH and Presence Subscription are disabled. To enable these features,
select the Allow PUBLISH check box and the Allow Presence Subscription check box.
How to use
This feature requires you to dial your User Password. The default User Password 1111 is not accepted. Please
change the User Password first.
Publish Presence can be set for an extension also from the System Administrator mode.
OR
• Dial 104
• Enter User Password on the prompt.
• Scroll to the desired Publish message from the menu:
• Absent
• Present
• Auto Detect
• Away
• On the Phone
• Do Not Disturb
• I am Mobile
• In Meeting
• Out for Meal
• Out of Office
• Press Enter key to select message.
• You get the confirmatory tone.
OR
• Dial 1072-014
• Enter Destination Number, i.e. the number of the extension Publish Presence is to be set.
• Scroll to the desired Publish message from the menu:
OR
• Dial 1097.
• Enter Extension number
• The status of the extension number you dialed will be displayed on your phone's LCD.
• Go ON-Hook.
0 Absent
1 Present
2 Auto Detect
3 Away
4 On the Phone
5 Do Not Disturb
6 I am Mobile
7 In Meeting
9 Out of Office
• Replace handset.
0 Absent
1 Present
2 Auto Detect
3 Away
4 On the Phone
5 Do Not Disturb
6 I am Mobile
7 In Meeting
9 Out of Office
• Replace handset.
• Lift handset.
• Dial 1097-Extension Number.
• You get confirmation tone.
• Go ON-Hook during confirmation tone.
• Your phone will ring and the status of the extension number you dialed will be displayed on your phone as
CLI.
What’s this?
When ‘Priority’ is assigned to trunks, whenever there are incoming calls on multiple trunks at the same time, the call
on the trunk with higher priority will be answered by the landing destination extension/Operator first. When Priority
is assigned to Extensions, calls from extensions with higher priority will have precedence in landing on the
destination extension.
Higher Priority can be assigned to Extensions of important or higher ranking persons in an organization; for
example, calls from senior managers or top executives in an organization can be allowed to be answered first by
the destination extension. Higher Priority can be assigned to particular Trunks, so that when there are incoming
calls on different trunks at the same time, the call on the trunk with the higher priority gets answered first by the
destination extension. For instance, special or private trunk lines, trunk lines dedicated as help lines or emergency
trunks, or trunks designated as hotlines can be assigned greater priority.
How it works
Priority can be assigned to all Trunk types (Analog, Mobile, SIP) and Extension types (SLT, DKP, SIP).
M e nu S at 0 1 0 5: 3 0
Priority: 9
Fwd Busy Fwd NR Cancel Mute Conf ee
rn c
e Tr ansfer
DN D Voi e
c Ma li Names Redi al Rej ect Hol d
1 2 3 de
ETERNITY
ab c f
4 gh i 5 j kl
6 m no
C LIR 7 pq r s 8 t uv
9 w xyz
C A4
H otli ne C A3
* 0 #
S IP 2 C A2
S IP 1 C A1
Incoming Call
at 10:00:12 Mobile 1
GSM
Fwd Busy Fwd NR Cancel Mute Conf eren c
e Transfer
DN D Voi c
e Ma li Names
Redi al Rej ect Hol d
1 2 ab c 3 de f
4 gh i 5 j kl 6 m no
C LI R 7 pq r s 8 t uv 9 w xyz
C A4
H ot il ne C A3 * 0 #
S IP 2 C A2
S IP 1 C A1
Here,
• There are two incoming calls, one on the Analog Trunk 01 and the Mobile Trunk 1 at the same time.
• Analog Trunk 1 has priority, ‘9’, the Mobile Trunk 1 is assigned priority ‘7’.
• Three extensions, 201, 202 and 302 are calling the Operator. Extension 301 has priority ‘7’, while
extensions 201 and 202 have the same priority, ‘5’.
Analog
10:00:10 9
Trunk 1
Mobile
10:00:12 7
Trunk 1
• These incoming calls, however, will appear on the Display of Operator’s phone (DKP or Extended IP
Phone) in the order of priority:
• Analog Trunk 1
• Mobile Trunk 1
• DKP 301
• SLT 201
• SLT 202
• Now, when the Operator goes Off-hook (pressing speaker key or picking up the handset), the call on
Analog Trunk 1 will be answered first, as Analog Trunk 1 has the highest priority.
• the Operator goes On-hook and then Off-hook, the call on Mobile Trunk 1 will be answered. Though
Mobile Trunk 1 and DKP 301 have the same priority, ‘7’, Mobile Trunk 1 will be answered first, following the
chronological order.
• When the Operator goes On-hook after answering the call on Mobile Trunk 1, the call from DKP 301 will be
placed on the Operator phone with a Priority Ring (configurable; default: Triple Ring).
• When the Operator goes Off-hook, the call from DKP 301 is answered.
• When the Operator goes On-hook and then Off-hook after answering the call from DKP 301, the call from
SLT 201 will get answered first, though both 201 and 202 have the same priority, ‘5’. In this case, Priority
Ring will not be played.
• Thus, calls from trunks and extensions are answered by the landing destination in the order of priority.
Where priority is the same, calls are answered in chronological order. Calls from extensions with higher
priority will be indicated by Priority Ring on the landing destination.
Priority is relevant only when there is more than one call on the destination.
You can assign Priority to SLT extensions. However, Priority is not relevant when the SLT is a landing
destination, because there cannot be more than one call ringing on an SLT extension at a time.
You can assign Priority to Trunks and Extension by configuring this parameter on the desired Trunk or Extension
port type under “Basic Settings”.
You can also configure, Priority Ring, if required. See “Distinctive Rings”
• Open Jeeves.
• When the page of the respective Trunk type opens, select the Trunk number on which you want to set
Priority.
• On this page, click the More link to view additional parameters on this page.
• When the page of the respective Extension type opens, click the tab of the Extension number on which
you want to set Priority.
• On this page, click the More link to view additional parameters on this page.
• Select a Priority level from 1 to 9 for the Extension. Default: 5 for all Extensions. Extensions designated as
Operator can be assigned higher priority than other extensions.
What’s this?
Extensions of ETERNITY NE can be protected from the intrusions by other extensions or from trunk calls by
activating Privacy.
How it works
Intrusions can occur on an extension when another extension invokes the following features:
• “DND-Override”
• “Interrupt Request (IR)”
• “Barge-In”
• “Raid”
To prevent such intrusions, ETERNITY NE enables you to set the following types of Privacy:
• Privacy from Interrupt Request, Barge-In, DND Override: This type of Privacy protects an extension
from intrusions by other extensions using Interrupt Request, Barge-In or DND Override.
For example: Extension A has Privacy from Interrupt Request, Barge-In and DND Override.
Extension A and B are in speech, Extension C attempts to intrude the conversation Interrupt Request or
Barge-In. Extension C’s call will be blocked and C will get error tone.
Now, Extension A has set DND and Extension B attempts to override it using DND Override. Since A has
Privacy from DND Override, B’s call will be blocked and B will get error tone.
• Privacy from Raid: This type of Privacy protects an extension from intrusions by other extensions using
Raid.
For example: This type of Privacy is set on Extension A. Extension A and B are in speech, Extension C
uses Raid to intrude the conversation. Extension C’s call will be blocked and C will get error tone.
• Privacy from Trunk call intrusion: This type of Privacy prevents the trunk landing destinations that are
busy from being intruded by another waiting call. For this type of Privacy to work, the feature “CO Call
Waiting”must be disabled on the extension.
For example: Extension A is the first landing destination for calls on Trunk 1. Extension A and B are in
speech. A new call lands on Trunk 1. If A has CO Call Waiting beeps disabled, A will not hear the intrusion
beeps. The system will land the call on the next landing destination for calls on Trunk 1.
For example: This type of Privacy is set on Extension A. Extension A and B are in speech, external caller
C uses DID to call extension A. C’s call will be blocked and C will get error tone.
How to configure
To provide Privacy to extensions, you must enable this feature in their “Class of Service (COS)”, for the Day and
Night.
By default, Privacy from Raid is enabled in the Class of Service of all Extension types: SLT, DKP, SIP. So, none of
the extensions can raid the other. You may disable this feature on extension, which you want to protect from Raid.
By default, Privacy from Interrupt Request, Barge-In and DND Override are disabled in the Class of Service of all
Extension types. You may enable this feature on extensions which you want to protect from intrusions using any of
these features.
By default, Privacy from DID is disabled on all Extension types. You may enable this feature on extensions which
you do not want external callers to reach.
By default, Privacy from DID is disabled on all Extension types. You may enable this feature on extensions which
you do not want external callers to reach.
By default, CO Call Waiting is on all Extension types. You may keep this feature disabled on extensions which you
want to provide Privacy from Trunk Call intrusion beeps.
For instructions, see “Class of Service (COS)”, and the description for configuring the different Extension
types—SLT, DKP, SIP— under “Basic Settings”.
What’s this?
Quick Dial provides DKP and Extended IP phone users the facility of ‘One-touch’ dialing of numbers stored in their
Personal Directory and the Global Directory.
How it works
Quick Dial is based on “Abbreviated Dialing”.
• the number must exist in the Personal or Global Directory assigned to the extension.
• Personal and Global Directory dialing must be allowed in the Class of Service of the extension.
• On the DKP and Extended IP Phones, DSS keys must be configured with the Short Codes or Abbreviated
Numbers that are to be dialed out. These short codes are derived from the Index numbers of the Personal
Directory and the Memory Location Index of the Global Directory.
• You can Quick Dial a number simply by pressing the DSS key.
• The system locates the number to be dialed out in the Personal/Global Directory on the basis of the Index
Number/Memory Location Index configured on the DSS Key.
How to configure
See “Abbreviated Dialing” for instructions on configuring and assigning the Personal and Global Directories.
To assign the Short Codes or Abbreviated Numbers to be used for Quick Dial on DSS keys, for each DKP/
Extended IP Phone extension,
• List down the numbers from the Personal Directory and Global Directory to be used for Quick Dial.
• If the number is from the Personal Directory assigned to the extension, note the Index number at which it is
stored in the Personal Directory: 001 to 025.
• If the number is from the Global Directory assigned to the extension, note the Memory Location Index at
which it is stored in the Global Directory: 100 to 999.
• Now, configure the Quick Dial numbers on the DSS keys of the DKP and Extended IP Phone.
For detailed instructions on configuring DSS Keys on a Digital Key phone, see “Phone Key Settings” under
DKP Extensions.
For instructions on configuring Quick Dial on the Matrix Extended IP Phone, see “Matrix Extended IP
Phone Key Settings” under SIP Extensions.
• As Offset, select the Index Number against which the number is stored in the Personal/Global Directory.
How to use
What’s this?
Raid allows you to interrupt a telephone conversation between two extension users, turning the conversation into a
three-way call.
You can use Raid to land in a conversation between two extension users, and between an extension user and an
external caller, with a warning—a beep—to the extension user. The extension user will hear a beep when you raid
and you will enter in to three-way speech with both parties.
You may also Raid a conversation without any warning by disabling the beep.
How it works
• A, B and C are extension users.
• C calls A.
• If any of these three parties disconnects, two-way speech is established between the remaining parties.
Feature Interactions
• Raid works only if the dialed extension is busy in two-way speech. The two-way speech may be with
another extension or with an external number on a trunk.
• You cannot Raid on Trunks, i.e. the external number which is in two-way speech with an extension. In this
case, C can raid the conversation between A and B, but not between A and another external number.
• Raid will not work if Privacy against Raid is enabled in the Class of Service of the extension being raided.
In this case, if Extension A has Privacy against Raid in its Class of Service, C will not be able to Raid the
conversation between A and B.
• The extension using Raid must have higher Priority assigned to it than the extension being raided. In this
case, C must have higher Priority than A to be able to invoke Raid.
Raid is a sensitive feature. You are advised to restrict access to this feature to select extension users.
By default, beep is played as a warning to the extension being raided. If required, you may disable the beep played
during Raid, by clearing the Play Beep when Raid/Conference/Dial-In Conference begins check box in the
System Parameters. See “System Parameters” for instructions.
How to use
For EON & Extended IP Phone Users
What’s this?
Generally, extensions users of the PBX are given a trunk access to make outgoing calls from their phones. It is also
common for a group of extensions to share the same trunks to make outgoing calls.
When an extension user of the PBX makes an outgoing call and the called party does not answer the call or is busy
on another line, it is possible for the called party to return the call (made by the extension user) on the basis of the
CLI number received.
However, when the called party returns the call, this incoming call mostly lands on the Operator extension, as
incoming calls are usually routed to the Operator.
Now, the Operator has no way of knowing which extension made the call. So the Operator cannot directly transfer
the call to that extension. Instead, the Operator must either ask the called party whom they wish to speak to and
transfer the call or put the called party on hold and find out the extension that made the call. This is an unwieldy
process for all concerned - the Operator, the called party and the extension user who originally made the call.
This can be overcome if the PBX is able to route the returned call to the original caller's extension.
ETERNITY makes this possible with the Return Call to Original Caller feature.
RCOC is not supported when calls are made from Analog trunks due to the signaling limitations of Analog
trunks.
How it works
The Prerequisites
• RCOC must be enabled on the desired Trunk(s): Mobile and SIP.
• RCOC must be enabled in the Class of Service assigned to the (original caller) extension where the call is
to be returned.
The Process
• When an extension having RCOC feature in its Class of Service makes an out going call, the system
checks if RCOC is enabled on the trunk through which the outgoing call is routed.
• If RCOC is enabled on the trunk, the system stores the record of the outgoing call in an internal database
referred to as the RCOC Table.
• The system sets RCOC for the outgoing call in the following conditions, according to the Destination
Port59:
• Whenever there is an incoming call on any trunk, the system matches the CLI of the incoming all with the
RCOC Table.
• If a matching record entry is found, the system routes the call to the original caller and clears the record
entry from the RCOC Table.
• The return call rings on the original caller's extension for the period of the Ring Back Tone Timer
(programmable; default 45 seconds). If the original caller does not answer the call within this Timer, the call
is routed to the Trunk Landing Group programmed for that trunk.
• If no match is found in the RCOC Table or the extension or the original caller is busy, the call will be routed
according to the incoming call logic programmed in the system.
• If a matching entry is found in the DISA CLI Authentication table, the system will give dial tone to the caller.
• The caller can now invoke RCOC feature by dialing ** (pressing Star key twice).
OR
• The caller can make calls to an extension or an external number or use a feature as required.
• If the caller invokes RCOC feature by dialing ** (pressing Star key twice), the system will check the RCOC
Table.
• If a matching record entry is found, the system routes the call to the original caller and clears the record
entry from the RCOC Table.
As RCOC is a “Class of Service (COS)” dependent feature, extensions that are not allowed this feature in
their COS cannot have their calls returned; even if this feature is enabled on the Trunk they used to make
the call.
• Each entry is kept for the duration of the RCOC Record Delete Timer (programmable; default: 999
minutes). Whenever a record is stored in the RCOC database, the Record Delete Timer for that entry is
activated. On the expiry of the Timer, the entry is deleted by the system.
• Each record is deleted from the database either after the call is returned or on expiry of the Record Delete
Timer.
• In case of Call Transfer, RCOC will be set for the extension which made the call on the trunk.
• The Ring Back Tone Timer is common to all internal calls; calls made from one extension will ring on the
destination extension till the end of this timer. Change in the Ring Back Tone Timer for RCOC returned
calls on original caller's extension will also be applied on Ring Back Tone Timer for all internal calls. So,
program this Timer taking this into consideration.
• Persons using DISA must be informed about RCOC feature access code ** and how to use this feature
when in DISA mode.
How to configure
For this feature to work, it must be enabled on the Trunk and in the Class of Service of the extensions. If desired,
the related Timers, i.e., the RCOC Record Delete Timer and the Ring Back Tone Timer may also be changed.
• Click the trunk parameters of the trunk type on which you want to enable this feature, namely:
• SIP
• Select the check box of 'Return Call to Original Caller (RCOC)' on the page to enable this feature on the
desired trunk port.
What’s this?
Certain features of ETERNITY NE like Alarms, SMDR, depend on the Date and Time of the system to function
accurately.
ETERNITY NE has its own Real Time Clock (RTC) to store date and time. When you select the Region, The RTC
parameters will be set automatically.
However, the RTC can drift over a long period. So, you may check and reset the RTC values at regular intervals to
correct this drift.
How to configure
1. Log in to Jeeves, refer
3. Click Real Time Clock sub-link. The Real Time Clock page opens.
• Date Format: Select the format to display the Date as DD-MM-YYYY (Day Month Year) or MM-DD-
YYYY (Month Day Year).
What's this?
Reminders are a variation of the “Alarms” feature, requiring the Date and Time to be set for each Reminder call.
Reminder calls are useful for extension users who wish to be reminded of important tasks or appointments.
For Reminder calls, date and time are set in the following format:
Date is set, according to Date Format you selected in the “Real Time Clock (RTC)” parameters, as:
• Day-Month-Year (DD:MM:YYYY)
Or
• Month-Date-Year (MM:DD:YYYY).
• Multiple Reminders can be set for an extension by the Operator and/or by the extension user.
• The mechanism for serving Reminders calls can be configured as 'Personalized' or 'Automated'.
• Reminders can be voice-guided, if the ETERNITY NE has a Voice Mail System (VMS) module installed in
it.
• ETERNITY NE can register as many as 48 Reminders set by the Operator and extension users.
How it works
Personalized Reminder
When the Reminder call serving mechanism is configured as 'Personalized',
• The Operator Phone rings first60, displaying the number of the extension to which the reminder call is to be
served.
• When the Operator answers this call, a call is placed on the extension on which the reminder call is set.
• The extension phone rings for the duration of the Alarm Ring Timer.
• When the extension user answers the call, the Operator greets the extension user with the reminder
message.
60. The Operator phone rings for the duration of the Alarm Ring Timer. If the Operator does not answer the call, the ETERNITY will
make two more Alarm Attempts at an Alarm Attempt Interval of 5 minutes to call the Operator.
• If the extension is busy61, the Operator phone will display a text message notifying that the extension
number is 'Busy'.
• inform the extension user about the Reminder in person or send someone to do it.
OR
OR
Personal Reminders will work even if the extension user has set DND or Call Forward.
Automated Reminder
When the Alarm serving mechanism is configured as 'Automated',
• The extension phone rings at the set time till the end of the Alarm Ring Timer. If the extension phone is a
DKP or the Matrix Extended IP Phone, Reminder message will appear on its display.
• When the extension user answers the call, s/he may be played music-on-hold, or a pre-recorded voice
message, or be connected to the Voice Mail, or routed to the Operator, depending upon the Alarm
Notification Type you have configured for the extension.
• If the extension user does not answer the reminder call, the ETERNITY NE makes two more attempts (in
all, 3 attempts) at an interval of 5 minutes between each attempt, to call the extension.
• If all Reminder call attempts go unanswered, the ETERNITY NE places the call on the Operator Phone.
The Operator Phone rings till the end of the Alarm Ring Timer. The Operator Phone displays the number of
the extension with the message 'No Reply'. The Reminder call is now considered as served.
• If the extension phone is busy, the ETERNITY NE will continue to make the Reminder call Attempts at the
Alarm Interval programmed. When all Alarm Attempts go unanswered, ETERNITY NE will place a call on
the Operator phone. The Operator Phone will display the number of the extension phone with the message
'Busy'.
61. An improperly placed receiver may also be the cause for the busy tone on the extension phone. In that case, the system will notify
the Operator Phone with the 'OFF-Hook Alert'.
• The extension phone rings for the Number of Alarm Attempt configured, at the set Alarm Attempt
Interval.
• The extension stops ringing when the user answers the call and dials 0 to acknowledge the Reminder
call. This reminder call Acknowledgement Code 0 is non-configurable.
• Reminder settings will be retained in the system during power down and system upgrades.
• When multiple reminder requests have been set by an extension user, the extension user cannot
selectively cancel a particular reminder request. Only the Operator can selectively cancel Reminders
set for an extension user from the System Administrator pages of Jeeves.
• It is not possible to modify—change the date and time—of a reminder request. So, you may cancel the
Reminder request and set a new one.
The status of Reminders set by Operator as well as extension users appears on this page, with details of time
(hours and minutes), and serving mechanism (personalized, automated).
The Operator can view the Reminder report whenever required and can also print this report.
How to configure
The configuration of Reminders is the same as Alarms.
• Configure, as required, the Alarm Call related parameters: Alarm Ring Timer, Number of Attempts,
Alarm Attempt Interval, Configurable Alarm Type and Configurable Alarm Category, and Snooze.
• Configure Macros, if the SLT extension has special function keys, and you want to a function key for the
Reminder feature.
If the Voice Mail System (VMS) is installed in the ETERNITY NE, it can offer voice-guided Reminders to extension
users and the Operator.
Voice-guided reminders lead users through a menu, helping them set the alarm in a step-by-step manner.
For SLT
If the SLT of the extension user has a special Reminder function key, s/he can set the alarm using this key.
• Press 'Reminders' key. (The label on the SLT key may differ from model to model)
• Follow the Voice Mail System prompts to set/cancel reminders.
• Without the Voice Mail System installed, the extension user having SLT with the special Reminder
function key will not be able to set/cancel Reminders. This extension user can set/cancel Reminders
only by dialing the feature access code for voice-guided Reminders.
OR
• Lift handset.
• Dial 1072-033
• Dial Extension Number.
• Dial Date and Time in the format:
DDMMYYYYHHMM
OR
• MMDDYYYYHHMM (users in USA)
• Dial 1 for Personalized, Dial 2 for Automated.
• You get confirmation tone.
• Replace handset.
To cancel reminder calls selectively, go to 'Reminder Status' page from the System Administrator of
Jeeves.
To set Reminder:
To cancel Reminder:
• Press 'Reminder' Key again.
OR
• Dial 162
• Select 'Cancel All'.
• Press Enter Key.
To set Reminder,
• Lift handset.
• Dial 162
• Dial Date and Time in the format
DDMMYYYYHHMM
OR
MMDDYYYYHHMM (users in USA)
• You get confirmation tone.
• Replace handset.
To cancel Reminder,
• Open Jeeves.
• Select the Cancel Reminder check box of the extension number for which you want to cancel the
reminder.
• click the Cancel Selected Reminders button at the bottom of the page.
What’s this?
This feature enables the DKP Extension users to listen to the conversations taking place in another location where
a DKP Extension is present.
Room Monitor can be used to monitor activities on the Shop Floor / Manufacturing area, without physically being
present there.
• Matrix Comsec is not responsible for any mis-/abuse of this feature by users.
How it works
• B is a supervisor in a Chemical Manufacturing Company.
• B’s room in on the second floor. The manufacturing area is on the ground floor.
• To keep track of the activities in the plant on the ground floor, B can activate Room Monitor. For this,
another DKP must be present where the activities are to be monitored.
• When B activates Room Monitor, the speaker of the DKP on the ground floor goes OFF-Hook. B can now
hear to all the sounds taking place in the ground floor.
Feature Interaction
• Call States:
• Room Monitor can be enabled from both-SLT and DKP, but the destination extension in the cabin for
Room Monitor must be a DKP.
• Microphone of the destination DKP is activated when the caller uses Room Monitor. But, the speaker is
not activated.
• One-way speech is established. The caller is able to the conversation taking place in room where
Room Monitor has been activated, but no one in the room will know that the monitoring is being done.
• Room Monitoring is terminated when the caller disconnects or the called party lifts the handset to dial
or the called party gets a call from any other extension.
• “Priority”: Room Monitor feature can be used by an extension whose priority is higher than the extension,
which has to be monitored.
How to use
For EON & Extended IP Phone Users
What’s this?
ETERNITY NE supports different extension and trunk port types. In the Selective Port Access feature, each port
type is assigned a Port Access Code. Extension users can access a particular port by dialing the Port Access Code
assigned to the Port and its Port Number.
How it works
• Extension user A wants to access a particular Mobile port, Mobile Port 1 to make a call. Extension A must
dial the Selective Port Access Feature Code, followed by the Port Type Code for Mobile ports and then dial
the Port Number.
• By default, the following access codes are assigned to each Port Type:
Port Type
Port Type Port Number
Code
CO 03 001 to 006
Mobile 25 01 to 02
SIP Trunk 26 01 to 04
Here, Extension A must dial 69-25-01, where 69 is the feature code for Selective Trunk Access, 25 is the
port access code for the Mobile Port, and 01 is the number of the Mobile Port which A wants to access.
Similarly, if Extension A wants to access SIP Extension 10, A must dial 69-34-010.
How to use
For EON & Extended IP Phone Users
What's this?
The ETERNITY supports Balance Inquiry and Recharging of the SIM Card installed in its Mobile Ports62.
How to use
To be able to use this feature, first get information about the following from your Network Operator:
• Balance Inquiry Number: This is the number provided by the Network Operator to the subscribers to
check Balance. Different Network Operators have different numbers. For example, the Balance Inquiry
number of Vodafone is *141#.
• Recharging Service Number: This is the number provided by the Network Operators to their subscribers
for Recharging Service. Different Network Operators have different numbers for Recharging Service. For
example, the Recharging Service Number of Vodafone is *140*.
• Click the Mobile Trunks link under Advanced Settings, click SIM Balance Recharge link.
62. ETERNITY NE supports Unstructured Supplementary Service Data (USSD), the standard for transmitting information over CSM
signaling channels and a commonly used method to query the available balance and other similar information in pre-paid GSM
services.
If you have not assigned any name to the mobile port, the Name field for that port will appear blank.
Balance Inquiry
• To make Balance Inquiry,
• Click the radio button Request of Balance Inquiry, for all those Mobile Ports for which you want to request
Balance Inquiry.
• Enter the Balance Inquiry Number provided by the Network Operator whose SIM Card you have installed
in the Mobile Port.
A maximum of 8 digits are allowed. The valid digits for Balance Inquiry number are any digits from 0 to 9
and the characters * and #
• Enter the following information in the appropriate fields under 'Recharge' for the Mobile Ports:
• Number: Enter the Recharging Service Number provided by the Network Operator in this field.
A maximum of 8 digits are allowed. The valid digits for Recharging Service number are any digits from
0 to 9 and the characters * and #.
• PIN: Enter the PIN number which is printed on the Recharge Voucher/Coupon. Your Recharge PIN
number may consist of a maximum of 20 digits.
The valid digits for Balance Inquiry number are any digits from 0 to 9 and the characters * and #.
Make sure you enter the digits and characters of the Recharge PIN number exactly as given on the
Recharge Voucher/Coupon.
• For each port that you send a Balance Inquiry/Recharge Request, you will get this USSD-Reply: "Please
wait, processing the request. Refresh the page to see the current status."
• The response received from the GSM network (including possible error messages) will be displayed under
'USSD-Reply'. When the USSD Reply is received from the network, it will appear with the Date and Time
stamp of ETERNITY in this field.
However, you can send Balance Inquiry/Recharge request for both the SIM cards present in the
system.
• During Balance Inquiry/Recharge-Request, the status of the Mobile port will be 'busy'. It will become
idle only after the USSD response is received from the Mobile network.
• The ETERNITY NE will clear the USSD Reply after system restart. So, each time you open the 'SIM
Balance and Recharge' page after system restart, the USSD Reply box will be blank.
What’s this?
You can use Self Ring Test to check the functioning of your own extension phone. Self Ring Test allows you to call
your own extension. You can check the ringing volume of your extension phone.
How to use
What's this?
Static Routing Table is required when you have more than one router (gateway) in your network and you want
ETERNITY NE to send packets to multiple routers/gateways for different types of calls.
Static Routing Table helps route calls between point to point sites (connected through Multi Protocol Label
Switching-MPLS, Frame Relay, etc.) and to public internet at the same time.
How it works
For example, two Local Area Networks, Network A and Network B, are connected through Frame Relay/ Multi
Protocol Label Switching (MPLS) network to give access to local resources and also to make Peer-to-Peer calls.
59.162.252.82
SIP Proxy
A B
192.168.1.0/24 192.168.2.0/24
Public
IP
Frame Relay/MPLS
192.168.1.1
ETERNITY ETERNITY
The Static Routing Table makes it possible to route different types of outgoing calls—Peer to Peer or Proxy—made
to different subnets through different Gateways.
The Static Routing Table defines the appropriate Gateway Address (or Router’s LAN Address) where the IP
packets are to be sent.
When ETERNITY NE sends packets, if the final destination IP Address and ETERNITY NE are not in the same
Subnet, the system will check the Static Routing Table.
If a perfect match is found, ETERNITY NE will start sending the IP packets to the corresponding Gateway Address
configured in the table.
If no match is found, ETERNITY NE will send the IP Packets to the Default Gateway Address (Network
Connection Type) you configured in “Network Parameters” the page.
How to configure
The Static Routing Table must be configured at each location where ETERNITY NE is installed.
• Open Jeeves.
• Click the Static Routing Table link. The Static Routing Table page opens.
• Destination Address: This is the address of the final destination where the call is to be made. This
can be a device IP Address or Network Address.
• Gateway Address: This is the IP address of the node where the IP packets are to be sent. Generally,
it is the IP address of the LAN interface of the Router.
To take the above example further, the Static Routing Table of ETERNITY NE at Location A should be
configured as:
• The Gateway Address 192.168.1.1 specifies the LAN address of the Router A which connects location
A and location B.
The IP address of the LAN interface of the router which connects Location A to the public internet
should be configured as Default Gateway in the Network Parameters of ETERNITY NE in location A.
With the Static Routing Table configured thus, all calls made by ETERNITY NE to 192.168.2.0/ 24 will
be routed through the router which connects Location A to Location B. Whereas, all calls made by
ETERNITY NE to address other than 192.168.2.0/ 24 will be routed through the Default Gateway.
Similarly, configure the Static Routing Table in ETERNITY NE at location B to enable calling from
Location B to Location A.
What’s this?
The ETERNITY NE can record the details of Internal, Incoming and Outgoing calls made from/to all the extensions.
This feature is called Station Message Detail Recording (SMDR).
You can store SMDR; obtain SMDR as a Report whenever you want or obtain it Online, immediately after the call
has been made or received. You can also use SMDR to calculate the cost of the calls.
• SMDR - Storage: These parameters enable the storing of the Incoming Calls Outgoing Calls and
Internal calls using filters. See “Station Message Detail Recording–Storage”.
• SMDR - Report: These parameters enable you to obtain offline Report of the records of Incoming,
Outgoing, and Internal calls. See “Station Message Detail Recording–Report”.
• SMDR - Online: These parameters enable you to obtain Online report of Incoming, Outgoing and
Internal calls. With Online SMDR you can obtain details of each call immediately after the call has been
made or received, also set the call record format you want for Incoming calls.
• SMDR - Posting: These parameters enable you to interface third-party call accounting software (CAS)
with ETERNITY for call cost calculation. You can select the protocol supported by the call accounting
software and further customize the handshaking parameters and call record formats. To know more
see “Station Message Detail Recording–Posting”.
What's this?
ETERNITY stores SMDR of Incoming calls, Outgoing Calls and Internal Calls. The call records are stored in the
SMDR buffer. For this, SMDR storage for these types of calls must be enable, and if required further filters can be
set.
ETERNITY NE can store 6000 outgoing calls, 5000 incoming calls, and 1000 internal calls in the SMDR buffer.
Once the SMDR buffer is full, the next call is stored in place of the oldest call in the SMDR buffer, using the First In
First Out (FIFO) logic.
The buffer can be cleared at any time from the System Administrator mode.
The SMDR buffer data is maintained even during power failures. However it is advisable to take frequent printouts
of the calls to avoid accidental loss of the data. See “Station Message Detail Recording–Backup”.
How to configure
To enable storage of SMDR of Outgoing, Incoming and Internal Calls, you must enable this feature in the system
and set the storage filters as per your requirement. To do this,
• Open Jeeves.
• To configure storage filters for outgoing calls, click SMDR Storage - Outgoing Calls to expand,
• Select the Store Outgoing Calls check box to enable storage of outgoing calls as per the filters you
set. If outgoing call storage is disabled, no outgoing call will be stored.
• If you want outgoing calls that exceed as certain duration to be stored, set the filter Store Calls with
speech duration more than (sec) to the desired duration. All outgoing calls with duration greater than
this value, will be stored.
• If you want outgoing calls exceeding certain metering units to be stored, set the filter Store Calls with
metering units more than (units) to the desired value. All outgoing calls that have metering units
greater than this value will be stored.
• Outgoing calls made by an extension user can be transferred to another extension. In such cases, you
may enable Call Splitting if you want to charge the amount to each extension according to the duration
of speech that each extension was involved in the call.
• If Call Splitting is disabled, you have the option of charging the call amount either to the extension that
originally made the call, i.e. the Originating Extension, or to the extension that was last in speech on the
call, i.e. the Terminating Extension.
• As the SMDR Outgoing call storage buffer is limited, you may limit the storage of calls to a certain
numbers only by configuring Store Calls made to numbers matching with.
You may configure as many as 250 numbers. The system will store records of outgoing calls made to
these numbers only.
ETERNITY uses the best match logic for storing these numbers. For example, if you configure 0 here,
the system will store all outgoing calls made to numbers starting with 0.
• To configure storage filters for incoming calls, click SMDR Storage - Incoming Calls to expand,
• Select the Store Incoming Calls check box to enable storage of incoming calls as per the filters you
set. If incoming call storage is disabled, no incoming call will be stored.
• If you want incoming calls that exceed as certain duration to be stored, set the filter Store Calls with
speech duration more than (sec) to the desired duration. All calls with duration greater than this
value, will be stored.
• If you want incoming calls that remain unanswered for certain duration to be stored, set the filter Store
Calls remaining un-answered for more than (sec) to the desired duration. All calls with duration
greater than this value, will be stored.
• If you want all calls, except calls received using DID to be stored, select the Store Normal (non-DID)
Calls check box.
• If you want DID calls to be stored, select the Store DID Calls check box.
• If you want all calls, except calls received using DID, that remained unanswered to be stored, select the
Store Normal (non-DID) Unanswered Calls check box.
• If you want all calls received using DID that remained unanswered to be stored, select the Store
Unanswered DID Calls check box.
• If you want calls made using DISA, select the Store DISA Calls check box.
• To configure storage filters for internal calls, click SMDR Storage - Internal Calls to expand,
• Select the Store Internal Calls check box to enable storage of internal calls as per the filters you set.
• To store internal calls that exceed as certain duration, set the filter Store Calls with speech duration
more than (sec) to the desired duration. All internal calls with duration greater than this value, will be
stored.
The SMDR stored in the buffer can be cleared at any time from the System Administrator mode, using Jeeves or by
dialing SA commands from an extension phone.
• Open Jeeves.
• To delete all records in the internal SMDR buffer, select Delete all Internal Calls check box.
• To delete all records in the Incoming SMDR buffer, select the Delete All Incoming Calls check box.
• If you want to delete all records in the Outgoing SMDR buffer, select the radio button Delete All OG Calls.
• You can also delete records of outgoing calls selectively, i.e. delete only records of outgoing calls made by
a particular extension or a range of extensions, or calls made between a certain period.
• To delete records of outgoing calls selectively, select the radio button Delete Selective OG Calls.
• To delete calls made by a particular extension or a range of extensions, select the Delete OG Calls
made by Stations button.
• To delete the records of outgoing calls made on a particular date or during a certain period, select the
Delete calls made between radio button. Select the start and end Date, Month and Year for this
period. If you want to delete the records of a particular date, enter the same date as start and end.
• The SMDR buffer will be cleared according to the settings you enabled on this page.
• 1072-132-DD-MM-YYYY-DD-MM-YYYY (The format of the date depends on the Date Format of the
system)
• Exit SA mode.
What's this?
The ETERNITY NE can generate report for the calls as and when the call is made, and send the report to the
printer or a computer.
Matrix ETERNITY NE supports Syslog Client for SMDR. The Syslog Client enables the system to send call records
in syslog format to the remote ‘Syslog Server’. You can view the call records on the remote server. ETERNITY NE
supports SMDR only on TCP.
SMDR generated as and when calls are made or received is called SMDR Online report. In the SMDR Online
report
How to configure
To get the Online report you must do the following:
• Enable SMDR Storage in the SMDR buffer. See “Station Message Detail Recording–Storage”.
• Assign the Syslog Server IP address as Destination address for Incoming, Outgoing and Internal calls. The
Online report is sent to this address as soon as the incoming call is completed.
• Open Jeeves.
• Log in as System Engineer.
• In the Destination IP Address: Port field, enter the IP Address and the port of the remote Syslog
Server.
• In the Destination IP Address: Port field, enter the IP Address and the port of the remote Syslog
Server.
• To configure Call Record Format for Incoming Calls, click SMDR Incoming Online Call Record Format
to expand.
• Serial Number: This is the serial number generated for each call record. Serial numbers are generated
from 000 to 999. When serial number '999' is reached, the numbers roll over to 000.
• Increment Counter: It increments when the serial number counter rolls over. The Increment counter
starts from A, ending at Z, and then roll over back to A.
• Property Code: This is the property code, if required. This is may be an abbreviation of the property
name.
• Extension Number: This is the extension number that answered the call. You can define the column
position and the field length for the extension number.
• Trunk Number: This is the number of the trunk on which the call was received.
The First Character in the Check-Inn Format is X (Fixed). The remaining three characters show the
software port number.
• Date: The date on which the call was received. The date fill flag is to be enabled.
• Filler Character field is applicable for Date, Month and Year, i.e. whether the single digit date is to be
printed as space-X or 0-X. For example, date = 1 is to be displayed as '1' or '01'.
• Where leading zeroes are not required, the date, month and year sub-fields are right aligned and the
spaces are filled with character 'space'.
• The Date field is not linked to the global flag of Date Format. The global Flag of Date format is used,
while using features or in configuration reports but not for SMDR Online. This is because the date
format used by the CAS is not the same as used by the users of the system.
• Time: The time when the call was received. The format of the time field and the time fill flag are to be
programmed.
• Filler Character field is applicable for Hours, Minutes and Seconds i.e. whether the single digit hour is
to be printed as space-X or 0-X. For example, hour = 1 is to be displayed as '1' or '01'.
• In case leading zeroes are not required, Date, Month and Year sub-fields are right aligned and the
spaces are filled with character 'space'.
• Answer Duration: The time after which the call was answered. Program the duration unit and the
duration fill flag.
• Speech Duration: The time for which the call was in speech with the extension.
When Duration Unit = Minutes, the rounding off to the nearest whole number is done. For seconds <= 30,
Minute is not incremented. For seconds > 30, minute is incremented.
• Called Number: This is applicable only for calls received on SIP trunks. The number dialed by the
caller is referred to as Called Number.
• DID Digits: This is the number dialed by the caller using DID.
• Remarks: You may use this for indicating the Type of Call, for example, DID.
• Reset Serial Number to 001: The Serial number counter can be reset to 001 after 24 hours (from
00:00 HH:MM) or every 6 hours. By default, 'No Compulsory Reset' is selected, which means the serial
number counter will not be automatically reset.
• Reset Increment Counter: The Increment Counter can be reset to 001 after 24 hours (from 00:00
HH:MM) or every 6 hours. By default, 'No Compulsory Reset' is selected, which means the serial
number counter will not be automatically reset.
How to use
You can start and stop SMDR Online report from the System Administrator mode using Jeeves or dialing SA
Commands from an extension phone.
• Open Jeeves.
• To start SMDR Online for Outgoing Calls, Incoming Calls and Internal Calls, click the Start button.
• Exit SA mode.
You may print the Online report captured on the Syslog Server after suitable modification of the format.
What's this?
The ETERNITY can generate SMDR reports in two modes:
• Online: as and when a call is made or received (see “Station Message Detail Recording–Online”)
Or
• Offline: whenever required, the records of calls stored in the buffer can be printed.
ETERNITY allows you to set a variety of filters for printing SMDR Reports.
ETERNITY NE supports Syslog63 Client for SMDR. The Syslog Client enables the system to send call records in
syslog format to the remote ‘Syslog Server’. You can view the call records on the remote server and print.
How to configure
To be able to generate SMDR -Report, you must do the following:
• Enable SMDR Storage in the SMDR buffer. See “Station Message Detail Recording–Storage”.
• Assign the Syslog Server IP address as Destination address for Incoming, Outgoing and Internal calls.
• Open Jeeves.
63. Syslog is one of the protocols used extensively for sending debug messages, and is defined in RFC 3164.
• In the Destination IP Address: Port field, enter the IP Address and the port of the remote Syslog
Server.
• In the Destination IP Address: Port field, enter the IP Address and the port of the remote Syslog
Server.
• In the Destination IP Address: Port field, enter the IP Address and the port of the remote Syslog
Server.
How to use
You can print SMDR Report whenever you want or schedule printing of the report from the System Administrator
mode using Jeeves or dialing SA Commands from an extension phone.
• Open Jeeves.
You can also print outgoing calls made using Account Code, and Calls for Department Billing Groups.
To print outgoing calls made to certain numbers, enter the desired numbers in the Print Calls made to
numbers matching with fields.
• To print outgoing calls made at a particular time, set the Hours and Minutes in 24-hour format calI in the
filter Calls made between 00: 00 and 23:59.
• If you want outgoing calls that exceed as certain duration to be printed, set the filter Calls with
duration more than (sec) to the desired duration. All outgoing calls with duration greater than this
value, will be printed.
• If you want outgoing calls exceeding certain metering units to be printed, set the filter Calls with units
more than (units) to the desired value. All outgoing calls that have metering units greater than this
value will be stored.
• Set the filters as desired. You can print records of incoming calls received on a specific extension or a
range of extensions and trunk ports: CO, Mobile, SIP.
You can also print incoming calls of different Call Types: Normal calls, calls received using DID, calls
that remained Unanswered, Unanswered DID calls, and calls made using DISA.
• To print incoming calls received from certain numbers, enter the CLI of these numbers in the Print
Calls received with numbers matching with fields.
• To print incoming calls received on a certain date or between a certain time period, set the filter Calls
received between. To print calls made on a particular date, select the same Date, Month and Year in
both fields.
• To print incoming calls received at a particular time, set the Hours and Minutes in 24-hour format calI in
the filter Calls received between 00: 00 and 23:59.
• To print calls that were kept on hold for more than a certain duration, set the filter Calls kept on hold
with duration more than (seconds) to the desired value.
• To print calls with speech duration of a certain duration, set the filter Calls with speech duration more
than (seconds)
Internal Calls
• To print Internal call Report with filters, click Internal Call - Print Filters link.
If you want to print calls made by a particular extension only, enter the same extension number in both
From and To fields.
You can also print calls made and calls received by these extensions by selecting the Call Type.
• Select Both as Call Type, to print calls made and received by the extensions.
• Select Caller as Call Type, to print only those calls that were made by the extension.
• Select Receiver as Call Type, to print only those calls that were received by the extension.
• Select None, if you do not want to use the Call Type filter.
You may print the Online report captured on the Syslog Server after suitable modification of the format.
• Exit SA mode.
• Exit SA mode.
Calls originated on Stations 1072-102-Extension No.-Extension No. Extension No.: 201 - 318
Calls originated on SIP 1072 - 188 - SIP- SIP SIP: 303 - 318
Calls made using Account Code 1072-115-Account Code-Account Code Account Code: 0000 - 9999.
Calls for Department Billing 1072-109-Dept Group No.-Dept. Group Dept. Group: 01 - 05
Group No.
• Exit SA mode.
• Exit SA mode.
Print Calls made by Stations 1072-137-Extension No.-Extension Extension No.: 201 - 318
Number No.-Call Type
Call Type: 1 - 3
1 = Receiver
2 = Caller
3 = Both
• Exit SA mode.
The Station Message Detail Record (SMDR)-Posting feature of ETERNITY NE is used for interfacing the system
with a third party Call Accounting Software (CAS).
When ETERNITY NE is interfaced with a third party Call Accounting Software (CAS) to determine the cost of the
call(s) made by the extension users, the system uses SMDR-Posting to send to CAS call record details, like
number to which the call was made by the extension user, number of the extension from which the call was made,
the date and time when the call was made, the duration of the call, metering pulses incurred for the call, etc. On
receipt of this information, the CAS calculates the cost of the call for billing.
As different CAS interfaces support different protocols, the ETERNITY NE offers the flexibility to send call detail
records using the protocol supported by CAS. ETERNITY NE supports as many as 15 different widely-used CAS
protocols such as, Holidex, Hobic, Micros A, Micros B, Comm One, Call-Inn, Bell-HOBIC, XIOX, RSI and others.
Each of these are described below.
Each posting protocol has its own handshaking protocol and call record format. You may configure any one of
these, depending upon the protocol supported by the CAS you have interfaced with ETERNITY NE. It is also
possible to customize the posting protocol to match the settings required by the CAS you have interfaced.
ETERNITY NE supports SMDR-Posting is on TCP/IP. The CAS is interfaced on the Ethernet port of the ETERNITY
NE. For every outgoing call, call detail record is posted on the Ethernet port.
SMDR-Posting Protocols
The ETERNITY NE supports as many as 15 different posting protocols from the system to CAS. The flow of
messages between the ETERNITY NE and various protocols of CAS Interface is described in the following.
1. Holidex/HOBIS A
This protocol is used by Amstar, CLS, Compass, Compu-solve, Dehan, Encore, Fabco, HIS, Holidex, HRGAS,
InnSolutions, Inn-Star, Lodgemate, Logistix, Omron, Otto Clerk, Reserve 5, Resort Computer, RDP, Springer-
Miller Systems, Star and Stuart.
HOBIS B is used by EECO and New Systems Protocol for transfer of messages from the ETERNITY to CAS.
ENQ >
< ACK
STX-(tex)-ETX-BCC >
< ACK
ENQ >
The ETERNITY will retransmit an ENQ after 5 seconds until the CAS accepts the message or until 4 NAK
responses are received.
No Response
The ETERNITY will retransmit an ENQ after 5 seconds until CAS responds or until 4 Unsuccessful ENQ
responses have been sent.
ENQ >
< ACK
STX-(tex)-ETX-BCC >
< NAK
< ACK
The ETERNITY will make a maximum of 3 attempts to send the message. If the message is still not
transmitted successfully, it will drop the message and proceed to the transmission of the next message.
ENQ >
< ACK
No Response
ENQ >
< ACK
Character
Control
Meaning ASCII
Char.
Value
ACK Positive Acknowledgement by the CAS (Indicates successful reception of data by HEX 06
the CAS)
NAK Negative Acknowledgement by the CAS (Unsuccessful reception of data by the HEX 15
CAS)
STX This marks the beginning of the data transfer. It also starts the accumulation of the HEX 02
BCC
ETX This is the last data character. Marks the end of the data. It is immediately followed HEX 03
by the BCC.
BCC This is a 'block check character' used to verify the successful transfer of data Depends on
between the systems. BCC is calculated by processing through an accumulatory the data
by an Exclusive OR operation. The BCC process should start with the character
after the STX character.
2. HOBIS B
Handshaking Parameters for HOBIS B are as below:
<LF> -message-<CR>
ACK
<LF>-message-<CR>
NAK
<LF>-message-<CR>
NAK
<LF>-message-<CR>
NAK
<LF>-message-<CR>
(no response)
After sending the message the ETERNITY NE will wait for the response to Data Timeout time. If no response
is received from CAS for the sent message, the ETERNITY NE will log this message in the System Fault Log
and look for new message to be sent to CAS.
3. Hobic
Protocol for transfer of messages from ETERNITY to CAS.
SOM-(tex)-EOM >
< ACK
< NAK
< NAK
The ETERNITY will make a maximum of 3 attempts to send the message. If the message is still not
transmitted successfully, it will drop the message and proceed to the transmission of the next message.
No Response
The ETERNITY will wait for 5 seconds to receive response from the CAS. If no response is received from the
CAS during this time period, it will drop the message and proceed to the transmission of the next message.
SOM This marks the beginning of the message. Start of Message. HEX 0A (Line Feed)
4. Micros A
Handshaking protocol for Micros A shall be as shown below:
ETERNITY CAS
<y
The first field of Text is always 'ac01' which marks the start of text. However, this field is a part of message and
not a link control character.
ETERNITY CAS
<n
The ETERNITY will make a maximum of 2 attempts to send the message. If the message is still not
transmitted successfully, it will drop the message and proceed for the transmission of the next message.
ETERNITY CAS
No Response
The ETERNITY will wait for 5 seconds and if no response is received from the CAS, it will drop the message
and proceed to the transmission of the next message.
5. Micros B
Handshaking protocol for Micros B is as shown below:
ETERNITY CAS
< ACK
The first field of Text is always 'ac01' which marks the start of text. This field, however, is a part of the
message and not a link control character.
ETERNITY CAS
< NAK
ETERNITY CAS
No Response
The ETERNITY will wait for 5 seconds and if no response is received from the CAS, it will drop the message
and proceed to transmit the next message.
7. RSI-CMS
ETERNITY sends call detail record in the following format to the RSI-CMS call accounting interface.
STX-Message-EXT
Call accounting interface will not send any response, ACK/NAK, for the messages received.
HOLIDEX:
Start
Field Filler/
Parameter Column Alignment Remarks
Length Character
Number
Extension 16 05 RA Space --
Number
Currency 33 01 NA NA $
Symbol
Trunk Number 00 NA NA NA --
Units 00 NA NA NA --
Location 00 NA NA NA --
Account Code 00 NA NA NA --
Prefix String 00 NA NA NA --
(ac01)
Remarks 00 NA NA NA --
• If the actual dialed number is less than the specified field width, the number will be sent as per the
programmed alignment.
• If the dialed number length is greater than the width of the dialed number field, the trailing number digits
will be removed. For example: if the number dialed is 15134036508 (11 digits) and the width of the
called number field is 8, then the first 8 digits will be sent.
HOBIS A:
Increment 04 01 NA NA --
Counter
Currency Symbol 33 01 NA NA $
Trunk Number 00 NA NA NA --
Units 00 NA NA NA --
Location 00 NA NA NA --
Account Code 00 NA NA NA --
Prefix String 00 NA NA NA --
(ac01)
Remarks 00 NA NA NA --
HOBIS B:
Same as HOBIS A, except the handshaking parameters.
HOBIC:
Currency Symbol 33 01 NA NA $
Trunk Number 00 NA NA NA --
Units 00 NA NA NA --
Location 00 NA NA NA --
Account Code 00 NA NA NA --
Prefix String 00 NA NA NA --
(ac01)
Remarks 00 NA NA NA --
By default the link control character EOM will be Carriage Return (CR - HEX 0D), Line Feed (LF - HEX 0A)
and Form Feed (FF - HEX 0C)
BELL HOBIC:
Currency Symbol 33 01 NA NA $
No Area Code is
provided.
Trunk Number 00 NA NA NA --
Units 00 NA NA NA --
Location 00 NA NA NA --
Account Code 00 NA NA NA --
Remarks 00 NA NA NA --
MICROS A:
Start Column
Parameter Field Length Alignment Fill Char. Remarks
Number
Serial Number 00 NA NA NA --
Increment Counter 00 NA NA NA --
Property Code 00 NA NA NA --
Trunk Number 00 NA NA NA --
Date 00 NA NA NA --
Time 00 NA NA NA --
Duration 00 NA NA NA --
Units 00 NA NA NA --
Currency Symbol 00 NA NA NA --
Location 00 NA NA NA --
Account Code 00 NA NA NA --
Remarks 00 NA NA NA --
MICROS B:
Same as MICROS A, except the handshaking parameters.
HILTON:
Currency Symbol 29 01 NA NA $
Location 00 NA NA NA --
Trunk Number 00 NA NA NA --
Units 00 NA NA NA --
Account Code 00 NA NA NA --
Prefix String 00 NA NA NA --
(ac01)
Remarks 00 NA NA NA --
In this protocol, the ETX is at the column position 61; hence the ETERNITY will send blanks from column position
50 to 60.
XIOX:
Currency Symbol 32 01 NA NA $
Location 00 NA NA NA --
Units 00 NA NA NA --
Trunk Number 00 NA NA NA --
Account Code 00 NA NA NA --
Prefix String 00 NA NA NA --
(ac01)
Remarks 00 NA NA NA --
Comm One:
Filler Char.
Start Column Field Filler Char.
Parameter Format Alignment (Decimal
No. Length Required?
Value)
Serial Number 00 00 X X X X
Increment
00 01 X X X X
Counter
Property Code 00 00 X X X X
Units 00 00 X X X X
Amount 00 00 X X X X
Currency 00 00 X X X X
Call Type
00 00 X X X X
Indicator
Location 00 00 X X X X
Remarks 00 00 X X X X
Serial Number 00 00 X X X X
Increment
00 01 X X X X
Counter
Property Code 00 00 X X X X
Extension
06 04 Fixed LA Yes 032
Number
Trunk Number 00 NA X X X X
Units 00 00 X X X X
Amount 00 00 X X X X
Currency 00 00 X X X X
Call Type
00 00 X X X X
Indicator
Location 00 00 X X X X
Account Code 00 00 X X X X
Remarks 00 00 X X X X
RSI-CMS:
Increment Counter 00 00 X X X X
Property Code 00 00 X X X X
DD/MM/
Date 38 10 RA Yes 032
YYYY
Units 00 00 X X X X
Amount 00 00 X X X X
Currency 00 00 X X X X
Location 00 00 X X X X
Remarks 00 00 X X X X
When the Call Detail Record format is customized, if there is a gap between two fields, these fields will be 'space'
(ASCII-32).
It is also possible to customize the posting protocol to match the settings required by the CAS you have interfaced.
• A computer with a spare Ethernet port (not supplied by Matrix) Or any free Ethernet Port of the LAN Switch
on which the CAS server application software is running.
Now, connect the Ethernet port ETERNITY NE to the computer (on which CAS server application is running) or to
one of the Ethernet ports of the LAN Switch, if the CAS server is in the same LAN.
How to configure
Configuring the SMDR-Posting feature involves the following steps:
• Enabling storage of SMDR for Outgoing (OG) Calls. By default, OG SMDR storage is enabled. Refer
“Station Message Detail Recording–Storage”.
• Open Jeeves.
• In the Destination Port, select Ethernet. This is Ethernet Port of ETERNITY NE on which SMDR Posting
is set up. Default: None.
• In the Destination IP Address: Port field, enter the IP Address and port of the PC on which the CAS
server application software is running, i.e. where ETERNITY should post SMDR.
• In the Listening Port (of ETERNITY) for Posting field, enter the Port number at which ETERNITY must
listen for SMDR Posting. Default: 6000.
• Response to ENQ Timeout: The time for which the sender waits for a response to ENQ from the
receiver. Default: 03 seconds.
• ENQ Retry Count - on No Response (to ENQ): The number of times the sender should send ENQ
before dropping the process, in case response is not received for the last message sent.
• ENQ Retry Time - on No Response (to ENQ): The time after which the sender should sent the ENQ
again, in case the response is not received for the last message sent.
• ENQ Retry Count - on Negative Response (to ENQ): The number of times the sender should send
ENQ before dropping the process, in case of a negative response received for the last message sent.
• ENQ Retry Time - on Negative Response (to ENQ): The time after which the sender should sent the
ENQ again.
• Data Transfer Retry Count - on No Response (to Data Transfer): The number of times the sender
should send ENQ before dropping the process. This parameter is used when ACK is received against
ENQ and there is some problem while sending the data.
• Data Transfer Retry Time - on No Response (to Data Transfer): The time after which the sender
should send the ENQ again before dropping the process. This parameter is used when ACK is
received against ENQ and there is some problem in sending the data.
• Data Transfer Retry Count - on Negative Response (to Data Transfer): The number of times the
sender should send ENQ before dropping the process. This parameter is used when ACK is received
against ENQ and there is some problem in sending the data.
• Data Transfer Retry Time - on Negative Response (to Data Transfer): The time after which the
sender should sent the ENQ again before dropping the process. This parameter is used when ACK is
received against ENQ and there is some problem in sending the data.
• Use ENQ Character: This flag is to be enabled if the protocol uses ENQUIRE (ENQ) Signal.
• ENQ Character: The ASCII character (Single Character) used to send ENQUIRE (ENQ) signal to the
receiver.
• Acknowledgement (ACK) Character: The ASCII character (Single Character) used by the receiver to
acknowledge the receipt of the Link Control Character/Message Data.
• No Acknowledgement (NAK) Character: This parameter signifies the ASCII character (Single
Character) used by the receiver to dis-acknowledge the receipt of the Link Control Character/Message
Data.
• Start of Packet Character: A string of four ASCII characters used by the receiver to indicate Start of
Packet. Each ASCII character is from 000 to 252. Start of Packet may be of one character only, in
which case the string should be completed by programming remaining three characters with ASCII Null
Character (000).
• End of Packet Character: A string of four ASCII characters used by the receiver to indicate End of
Packet. Each ASCII character is from 000 to 252. End of Packet may be of one character only, in which
case, the string should be completed by programming the remaining three characters should be
programmed as ASCII Null (000).
• Use Byte Code Check (BCC): This flag is to be enabled when the protocol uses BCC Signal.
This may be required if you selected a 'customized' protocol. To refine Call Record Format,
Refine the following format parameters according to the type of posting protocol you have selected and the
requirement of the CAS being used by the organization.
• Serial Number: This is the serial number generated for each call record. Serial numbers are generated
from 000 to 999. When serial number '999' is reached, the numbers roll over to 000.
• Increment Counter: It increments when the serial number counter rolls over. The Increment counter
starts from A, ending at Z, and then roll over back to A.
You must program this string keeping in mind the field length used by the selected/customized posting
protocol.
The default value of the default Property Code String has been set as 'AAA', as at least two known
protocols use this field. You can set a different value here and the new value will appear in the CDR
record, irrespective of the protocol type selected.
If Bell Hobic or Hilton has been selected, you should program this field as 'AAA'. If Xiox protocol has been
selected, you should program this field as HTL. These values are not protocol dependent, but can be
configured by you.
• Extension Number: This is the extension number from which the call was made. You can define the
column position and the field length of the Extension number in the Call Detail Record.
• Trunk Number: This is the number of the trunk from which the call was made.
The First Character in the Check-Inn Format is X (Fixed). The remaining three characters show the
software port number.
• Date: The date on which the call was made. The date fill flag is to be enabled.
• Filler Character field is applicable for Date, Month and Year, i.e. whether the single digit date is to be
printed as space-X or 0-X. For example, date = 1 is to be displayed as '1' or '01'.
• Where leading zeroes are not required, the date, month and year sub-fields are right aligned and the
spaces are filled with character 'space'.
• The Date field is not linked to the global flag of Date Format. The global Flag of Date format is used,
while using features or in configuration reports but not in CAS. This is because the date format used by
the CAS is not the same as used by the users of the system.
• Time: The time when the call was made. The format of the time field and the time fill flag are to be
programmed.
• Filler Character field is applicable for Hours, Minutes and Seconds i.e. whether the single digit hour is
to be printed as space-X or 0-X. For example, hour = 1 is to be displayed as '1' or '01'.
• In case leading zeroes are not required, Date, Month and Year sub-fields are right aligned and the
spaces are filled with character 'space'.
• Duration: The duration of each call. Program the duration unit and the duration fill flag.
When Duration Unit = Minutes, the rounding off to the nearest whole number is done. For seconds <= 30,
Minute is not incremented. For seconds > 30, minute is incremented.
• Amount: This is the Amount of the call. Program the amount format and the fill flag.
• Filler Character field is applicable for both the sub fields of Amount viz. Rupees/Paisa i.e. whether the
single digit Rupee is to be printed as space-X or 0-X. For example, Rupee = 1 is to be displayed as '1'
or '01'. Where leading zeroes are not required, the Rupee and Paisa are right aligned and the spaces
are filled with character 'space'.
• When Amount Format = Higher Currency, rounding to nearest whole number is done. For Lower
Currency <= 50, Higher Currency is not incremented and for Lower currency > 50, Higher Currency is
incremented.
• Currency: This is the symbol of the currency in which the Amount is charged. A maximum of 8 ASCII
Characters are allowed.
• Generally, Currency Symbol field prefixes to Amount field. Hence, to comply with various CDR formats,
it is recommended that the column position of Currency Symbol and Amount field should be
programmed properly.
• You can change the Currency Symbol used in the OG-SMDR Format.
• Call Type Indicator: This indicates the type of call made, i.e. whether local, national or international.
Define the Call Types in the “Call Type Indicator”.
• Location: This column indicates the location of the external number to which the call was made.
The system detects the location from the called location programmed in the Area and Country Code
Tables.
Called Location is programmed as one of the parameters of the Area Code Table and Country Code Table.
Depending upon the prefix dialed, the Location string is picked up from either Country Code table or Area
Code table.
The Called Location parameter in the Country Code table and Area Code table is of 8 Characters.
If the number of characters in the field Called Location is more than Field length then the remaining
characters will not be printed (overlapped by next field).
If the number of characters in the field Called Location is less than Field length then the remaining
characters in the field Called Location will be filled by spaces.
• Called Number: This is the external number to which the call was made.
In the Closed numbering system, the Area Code, Exchange Code and the Subscriber number are of fixed
length. In such case, including '-' in the called party number is not difficult. Hence, '-' is put in the called
party number. The called party number is assumed to be of 10 digits. The first '-' is placed after four digits,
counting from the right. The second '-'is placed after seven digits, counting from the right. If the dialed
number is a local number of 7 digits then the second '-'is not placed. Also, the remaining three digits are
not placed, but filled with character 'space'.
In this case, even if the call is made to a geographical area where open numbering system is followed, '-' is
placed in the same way.
• Account Code: This is the Account Code (Refer Note4) using which the call was made.
• Remarks: This column indicates the details of the call; whether it was a DISA call, DOSA call, Auto
Redial Call, type of call maturity.
Fixed Characters are used to indicate the type of call, call details, etc. The notations for the Remarks
field are:
D DISA Call
C CPD
K 12KHz/16KHz
R Reversal
D Delay
I Connect
• Reset Serial Number to 001: The Serial number counter can be reset to 001 after 24 hours (from
00:00 HH:MM) or every 6 hours. By default, 'No Compulsory Reset' is selected, which means the serial
number counter will not be automatically reset.
• Starting Character - Increment Counter: Specify the starting character of the increment counter as
the serial number rolls over, in this field.
• Reset Increment Counter: The Increment Counter can be reset to 001 after 24 hours (from 00:00
HH:MM) or every 6 hours. By default, 'No Compulsory Reset' is selected, which means the serial
number counter will not be automatically reset.
• Prefix String Required: This flag is to be programmed if the prefix string 0ac1 is to be sent when
interfacing with OG-SMDR Posting Protocol.
• In the Dialed Number String column, enter the number strings for each Call Type. You can enter the
prefix, e.g. 0 for long distance calls, 2 for local numbers, etc.
• For each Dialed Number String, define as Call Type Indicator, this is an abbreviation of the Call Type,
e.g.: LD for long distance, INTL for International, etc.
Call Type
Dialed Indicator
Number Index Meaning
Number String (Text
String)
01 0 LD Long Distance
02 95 IC Inter Circle
04 00 INTL International
: : : :
36 2 L Local
You may enter as many Call Types as supported by the Posting Protocol you have selected.
What's this?
Station Message Detail Records (SMDR), i.e. records of internal, incoming and outgoing calls made to/from
extensions of the ETERNITY are stored64 by the system in the 'SMDR Buffer'. The SMDR Incoming Call buffer has
a capacity of storing a maximum of 5000 incoming call records. The SMDR Internal Call buffer can store upto 1000
internal call records, while a maximum of 6000 outgoing call records can be stored in the SMDR Outgoing Call
buffer.
SMDR buffer data can be cleared by the System Engineer or System Administrator manually or the system clears
the data automatically when the SMDR buffer is full, by replacing the oldest call record with the latest (First In First
Out logic).
While the SMDR buffer data is maintained even during power failures, accidental data loss is not an uncommon
occurrence.
Therefore it is advisable to back up SMDR records to restore accidentally deleted, lost or corrupted files.
Back-up of SMDR records can be stored on a computer for retrieval later. The ETERNITY NE provides an
embedded FTP server to transfer SMDR call records on a computer.
You can copy the SMDR records folder using either Windows FTP or Mozilla Fire Fox.
• Type the current IP Address of the Ethernet Port of ETERNITY NE in the Address bar as ftp://
192.168.1.101
64. ETERNITY will store SMDR only if the SMDR-Storage flag has been enabled. The call records are stored according to the Storage
filters set.
• In the Password field, enter the SE password (default 1234) and click the Log On button.
• Click the Backup SMDR Data link under the SMDR link.
• Follow the same steps as described for accessing the FTP server in the topic “Upload configuration files
using Mozilla Firefox” under “Configuration Upload” (see Maintenance and Troubleshooting).
• Copy the smdr folder on the FTP to another location on your computer.
While uploading SMDR files on to ETERNITY, first, remove the current files in the system. Copy the new
files from computer (backup source) on to the system.
You can tag the back-up folders on the computer by date to store the records as archives.
What’s this?
The ETERNITY NE monitors all its activities and maintains records of these activities in the System Activity Log.
The System Activity Log has a buffer capacity of 250 records. The Activity Log stores records using the FIFO
method.
This log can be printed on a printer or downloaded on a computer in form of a report. The report can be printed or
downloaded in two modes:
• Report (Offline): The activity report is printed/downloaded whenever desired. In the Offline mode, the last
250 activities recorded by the system are printed/downloaded.
The System Administrator can print/download System Activity Log, online or offline.
Matrix ETERNITY NE supports Syslog Client for System Activity Logs. The Syslog Client enables the system to
send activity logs in syslog format to the remote ‘Syslog Server’. You can view the logs on the remote server.
How it works
• The Syslog Server address must be assigned to which the system can send the activity log.
• If the System Administrator extension is a DKP or Extended IP Phone, a DSS Key can be assigned for
System Activity Log.
• Whenever an activity is recorded by the system, the DSS key, if assigned for this feature on the System
Administrator’s DKP/Extended IP Phone extension, is turned ON.
• The System Administrator can view the activity log by pressing the DSS key (if assigned). The DKP/
Extended IP Phone of the System Administrator will display the activity in this format:
Format of Date will be DD-MM or MM-DD as per Date Format selected in the Real Time Clock settings of
the system.
Index Activity
3 DD-MM-YYYY HH:MM:SS Card Present: Slot=SS, Type: <ZZZZZ > Card VvRr
SS-Slot Number, PP -Port Number, TT -Trunk Number, XXXX -Station Number, v-Version for VvRr, r-
Revision for VvRr, ZZZZ = Card Type
How to configure
The two functional parts of system activity log are:Storage and Report Generation in the Online or Report
modes.To be able to use this feature, you must enable storage of Activity Logs, and assign the Syslog Server
address as Destination Port for the logs.
If the System Administrator phone is a DKP or an Extended IP Phone, you may assign a DSS key for System
Activity Log. For instructions, see Phone Key Settings under “DKP Extensions” and Phone Key Settings under
“Matrix Extended IP Phone Settings – Location 1” in SIP Extensions.
• Open Jeeves
• Scroll to the System Activity Log link under Advanced Settings and click this link.
• To generate System Activity Log - Online, i.e. as and when the activity is recorded, select Destination
Port. Default: None.
• In the Destination IP Address: Port field, enter the IP Address and the port of the remote Syslog
Server.
• To generate System Activity Log - Report, i.e. offline, whenever desired, select Destination Port.
Default: None.
• In the Destination IP Address: Port field, enter the IP Address and the port of the remote Syslog
Server.
• Open Jeeves.
• To clear System Activity Logs from the buffer, click the Clear SAL button.
• Exit SA mode.
You may print the logs captured on the Syslog Server after suitable modification of the format.
The Online System Activity Log report would look like this:
The Offline System Activity Log report would look like this:
-------------------------------------------------------------------------------
Eternity V10R04 Page : 1
What’s this?
The ETERNITY provides a facility to display the last activity monitored by the system on the System Administrator’s
extension phone.
How to use
To be able to use this feature optimally, the System Administrator extension phone must be a DKP or an Extended
IP Phone, and a DSS Key must be assigned on the phone to System Activity Log Display. For instructions, see
Phone Key Settings under “DKP Extensions” and Phone Key Settings under “Matrix Extended IP Phone Settings
– Location 1” in SIP Extensions.
• Go Off-hook.
• Press the DSS key assigned to System Activity Log Display.
OR
• Dial 1072-009
• The last recorded Activity log appears on your phone’s display in this following format: Date-Time-Activity
Index
The Date and Time are in <DD-MM-YYYY HH:MM:> format
The Activity Index is a two digit number from 01 to 27.
See System Activity Log Activity Index table in “System Activity Log”.
Format of either DD-MM or MM-DD will be as per date format set in the system. Refer chapter “Real Time
Clock (RTC)” for setting date format.
What’s this?
The ETERNITY maintains a log of all system faults. The system Fault Log has a buffer capacity of 100 records. The
Fault Log stores records using the FIFO method.
The System Fault log can be printed on a printer or downloaded on a computer in form of a report. The report can
be printed or downloaded by the System Administrator in two modes:
• Report (Offline): The faulty report is printed/downloaded whenever desired. In the Report (Offline) mode,
the last 100 faults recorded by the system are printed/downloaded.
Matrix ETERNITY NE supports Syslog Client for System Fault Logs. The Syslog Client enables the system to send
fault logs in syslog format to the remote ‘Syslog Server’. You can view the logs on the remote server.
How it works
• A destination port for sending the report must be selected to which the system can send the log.
• If the System Administrator extension is a DKP or Extended IP Phone, a DSS Key can be assigned for
System Fault Log.
• Whenever a fault is detected, the LED of the Fault Log DSS key, if assigned, is turned ON.
• If more than one DKP/Extended IP extension is assigned Fault Log DSS Key, the LED of all keys will be
turned ON.
• The System Administrator must acknowledge the Fault indication by pressing the Fault Log key or by
dialing the Fault Log access code. The LED of the Fault Log key is turned OFF.
The following table summarizes the different activities that are logged into system fault log.
2 DKP Absent, Flexible No. of the port, Slot No. <>, Port No<>
3 SLT Short, Flexible No. of the port, Slot No. <>, Port No <>
4 SLT Open, Flexible No. of the port, Slot No. <>, Port No. <>
9 RTC Failure
10 VoIP LAN Lost, Slot No. <>, Port No. <> on DD-MM-YYYY at HH:MM:SS
Registration Timer Fail: The system may fail to load either the Re-registration Timer or the Registration
Retry Timer. In such a case the Proxy SIP trunk will remain un-registered and will not be functional.
The system will decode the registration status message received from ETERNITY VoIP module and, if it is
found to be a problem caused by Registration Timer Failure, this will be logged in the System Fault Log.
This can happen to one or more SIP trunks, while the other SIP Trunks functioning normally. You need to
restart the system to resolve the problem.
How to configure
To be able to use this feature, you must enable storage of Fault Logs, and assign a Destination Port for the Fault
Logs.
If the System Administrator phone is a DKP or an Extended IP Phone, you may assign a DSS key for System Fault
Logs. For instructions, see Phone Key Settings under “DKP Extensions” and Phone Key Settings under “Matrix
Extended IP Phone Settings – Location 1” in SIP Extensions.
• Open Jeeves
• Scroll to the System Fault Log link under Advanced Settings and click this link.
• To generate System Fault Log - Online, i.e. as and when the fault occurs, select Destination Port.
Default: None.
• In the Destination IP Address: Port field, enter the IP Address and the port of the remote Syslog
Server.
• To generate System Fault Log - Report, i.e. offline, whenever desired, select Destination Port. Default:
None.
• In the Destination IP Address: Port field, enter the IP Address and the port of the remote Syslog
Server.
• Open Jeeves.
• To clear System Fault Logs from the buffer, click the Clear SFL button.
• Exit SA mode.
You may print the logs captured on the Syslog Server after suitable modification of the format.
Blank 00
Time (HH:MM:SS) 12
Event Description 24
What’s this?
The ETERNITY provides a facility to display the last fault monitored on the system on the System Administrator’s
extension phone.
How it works
To be able to use this feature optimally, the System Administrator extension phone must be a DKP or an Extended
IP Phone, and a DSS Key must be assigned on the phone to System Fault Log.
• When a fault occurs, the LED of the DSS Key assigned for the System Fault Log, glows.
• The System Administrator may press the DSS key or dial the System Fault Log feature access code to
acknowledge it.
• On pressing the DSS Key or dialing of the acknowledgment command, the LED of the Fault Log key is
turned OFF.
How to use
• Go Off-hook.
• Press the DSS key assigned to System Fault Log Display.
OR
• Dial 1072-010
• The Fault log appears on your phone’s display in this following format:Date-Time-Fault Index
The Date and Time are in <DD-MM-YYYY HH:MM:> format
The Activity Index is a two digit number from 01 to 12.
See System Fault Log Activity Index table in “System Fault Log”.
What’s this?
System Parameters are general parameters, related to features and facilities that are applied system-wide, such as
customer name, Day-Night mode, storage of call logs, end of dialing, alarms, DID call disconnect options,
Presence, and DND messages. Each of these is described briefly here.
Customer Name
You can assign the name of the enterprise/organization that is using ETERNITY NE as the Customer Name. The
Customer Name may contain up to 80 characters. You may enter the address of organization/enterprise along with
the name.
The Customer Name you assign will appear on the various System Reports generated and printed by the
ETERNITY NE.
System Parameters
The System Parameters include the feature options:
• Global Hold: This parameter is related to the “Call Hold” feature of ETERNITY NE. Global Hold enables
you pick up a call that has been put on hold by any extension from a DKP extension of ETERNITY NE.
• Store Internal Calls in Missed Call Log: ETERNITY NE stores call logs of external calls only. You may
instruct the system to store internal calls in the Missed Call log by enabling this parameter. See “Call
Logs”.
• Store Internal Calls in Dialed Call Log:You may instruct the system to store internal calls in the Dialed
Call log by enabling this parameter. See “Call Logs”.
• Store Internal Calls in Answered Call Log: You may instruct the system to store internal calls in the
Answered Call log by enabling this parameter. See “Call Logs”.
• End of Dialing Digit: End of Dialing Digit is a single digit, on receipt of which, the system considers the
number string dialed by the extension users as the complete string. It does not wait for further digits to be
dialed, and dials out the number. The digits * (star) or # (hash/pound) are used to indicate end of dialing to
the system, as these are unique and distinguishable from the digits generally dialed by extensions (0, 1,
2... to 9). Default: #
• Give Off-hook Alert to Operator: When this flag is enabled, the system will detect extensions that are off-
hook and ring on the Operator extension to alert the Operator about the state of the phone. This alert is
useful for detecting whether the handset of extension phones are placed correctly. Read the feature
description for “OFF-Hook Alert” to know more.
• Day/Night Mode: Certain features of the ETERNITY NE require extensions and trunks to behave
differently according to the working hours and non-working hours, which are referred to as Time Zones.
The Time Zones, i.e. working hours, non-working hours, are defined for the entire week in a Time Table.
Time Table is assigned to trunks, extensions and other time-zone dependant features.
• Emergency Dialing Reporting: When this flag is enabled, the system detects the extension that has
made the emergency call and reports it to the Operator extension. Thus the Operator can know which
extension has made an emergency call. See “Emergency Detection and Reporting” to know more.
• Replace '+' from CLI: The GSM network presents the calling party number with prefix '+' to the called
party. ETERNITY NE allows you to remove '+' and replace it with an appropriate number string as required.
To do this, you must enable Replace '+' from CLI by selecting this check box.
You may also program the number string with which ‘+’ is to be replaced in the CLI in the Replace '+' from
CLI with the number string field. In this field, enter the number string with which you want to replace '+'
received as prefix of calling party number.
If you keep the number string field blank, ETERNITY NE will remove '+' sign from the CLI of calling party
and present the remaining digits on the CLI of the Called Party.
For example:
The number string +919327237228 is received as CLI.
If ‘00’ is configured as the replace string, the CLI number would become 00919327237228
If no replacement string is configured (i.e. left blank), the CLI number would be presented as
919327237228.
• Disconnect DID Call, when Dialed Number is Busy: when this flag is enabled, the DID call will be
disconnected if the landing extension(s) is busy. The DID call will not be routed to the Operator. Default:
disabled.
• Disconnect DID call, when Dialed Number is not Responding: when this flag is enabled the system
disconnects the DID call if there is no reply from the landing destination extension(s). The DID call will not
be routed to the Operator. Default: disabled.
• Disconnect DID call, when caller does not Dial any Digit: when this flag is enabled, the system will
disconnect the DID call if the caller fails to dial a digit within the First Digit Wait Timer. The DID call will not
be routed to the Operator. Default: disabled.
• Play Beep when Raid/Conference/Dial-in Conference begins: This is a common flag for three features
of ETERNITY NE: “Raid”, “Conference-Multiparty” and“Conference Dial-In”. When this flag is enabled,
• the system plays a warning beep to the extension which is being raided by another extension, before
establishing three-way speech.
• the system plays beeps to the other participants in a Dial-In Conference when a new participant joins in
(i.e. dials in to an on-going Dial-In Conference)
• the system plays beeps to the other participants connected in a Multi-Party Conference, when a new
participant is included.
If you disable this flag, no warning Beep will be played in Raid, the existing participants in a Dial-In or Multi-
party conference will not hear any beep tone indicating the addition of a new participant.
When this flag is disabled, no indication will be given to the opposite party when the call is being taped/
conversation is being recorded. Default: Enabled.
• Play Feature Tone in place of Dial Tone when Call Forward is Set: You can select whether the you
want the system to play Feature Tone instead of Dial Tone to the extensions when Call Forward is set on
these extensions. When this flag is disabled, the system will play dial tone to the extension on which Call
Forward is set, whenever the extension goes Off-hook. Default: Enabled.
• Call Proceeding Tone–Multistage Dialing: This flag is used in “Multi-Stage Dialing” where you need to
configure Pause and Wait for Answer in the Substitute Number string for the number string dialed by the
extension users.
When an extension user makes a call using a Calling Card, and the system dials out the number using the
Multi-Stage, the extension user will get Ring Back Tone twice: first after the system has dialed the Calling
Card Number, and again after the system has dialed out the destination number (called party number).
Thus the extension user will get Ring Back Tone, twice. To avoid this, you may configure the 'Call
Proceeding Tone' to be played by the system when using Multi-Stage Dialing.
You must configure the type of 'Call Proceeding Tone', according to your requirement; whether the
extension user should be connected to the speech path when the Calling Card number is out dialed or
when the called party number is out dialed. You can select any of the following Call Proceeding Tone
options, as per your requirement:
• Network Tone: If this option is selected, the extension user will get Ring Back Tone after dialing the
calling card number and again, after the system has dialed the called party number (when the system is
dialing out the number with Pause and Wait for Answer configured in the substitute number string).
• Pseudo Tone: If this option is selected, the extension user will get Feature Tone when the user has
completed dialing all the digits. At the end of the tone, the extension user gets connected to the called
party (destination number).
• Silent: If this option is selected, the extension user will get Silence (no tone), after the extension user
has completed dialing all digits. After dialing out the called party number in DTMF, the system will
connect the caller to the called party number (destination number).
• Companding Algorithm: The companding Algorithm —A law or law—is automatically selected when
you select “Region” for ETERNITY NE. However, if necessary, you may change the default companding
Algorithm that appears in this field. Select the companding Algorithm according to the Regulatory
Requirement of the country where ETERNITY NE is installed.
• Language of SE, SA Web Interface: The GUI of ETERNITY NE supports the languages English, Italian,
Spanish, French, German, and Portuguese. When you select “Region” for ETERNITY, one of these
languages will be applied as appropriate for the region you selected. For instance, if you selected India,
English will be applied. If you selected Spain, Spanish will be applied. If you selected a country where
none of these languages are the local language, English will be applied.
The language set by the system on Region selection will be applied on the pages of the GUI for every login
session. You can change the default language set on Region selection, by configuring this parameter.
• Display Presence Status during Call on DKP: If you want to allow DKP extension and Extended IP
extension users to view the presence status of the extension they are calling, you must enable this flag.
Default: Disabled.
• Country Code: This is the Country Code of the country where ETERNITY is installed. The Country Code
helps ETERNITY detect whether the Incoming CLI received is a national or an international number. Do
not enter any prefix for the Country Code. For example, if your ETERNITY is installed in USA, enter only ‘1’
as the Country Code. Do not enter ‘+’ or “00’ as prefix to the country code ‘1’. By default the Country Code
is ‘91’ (India).
• Area Code: This is the Area Code of the place where the ETERNITY is installed. The Area Code helps
ETERNITY detect whether the Incoming CLI received is a local number. Do not enter any prefix for the
Area Code. For example, if you want to enter Area Code for Mumbai, enter only ‘22’. Do not enter the
prefix ‘0’ to the area code. By default, Area Code is ‘265’ (Vadodara city).
• International Prefix: These are digits required as Prefix for dialing International Numbers. The prefix may
be up to 5 digits, with numbers from 00000 to 99999. By default, ‘00’ is set as the prefix for dialing
International numbers.
• National Prefix: These are digits required as Prefix for dialing long distance, National (within the country)
numbers. The prefix may be pot 5 digits, with numbers from 00000 to 99999. By default, ‘0’ is set as prefix
for dialing national numbers.
• Area Code required to make local calls?: Depending on the dialing pattern of your local public
telephone network, you may choose:
• No (Area Code not required), if your public telephone network does not require the dialing of Area Code
for local numbers.
• Yes (Area Code is required), if your public telephone network requires you to dial the Area Code for
local numbers.
• Yes, with Prefix, if your public telephone network requires you to dial Area Code with a particular Prefix
for local numbers. If you select this option, you must also enter the prefix digits for the area code for
local calls in the Prefix Area Code field.
By default the following messages are configured as DND Text messages, you may customize these messages,
according to your requirement.
1 Do Not Disturb
2 Unavailable
3 In Meeting
4 In Conference
5 Try on Mobile
6 On Vacation
7 On Business Trip
8 Out of Office
9 With a Guest
Publish Message
ETERNITY offers 10 different text Messages to Publish Presence, as listed in the table below. You can customize
message 6 to 9 to match your requirement.
0 Absent
1 Present
2 Auto Detect
3 Away
4 On the Phone
5 Do Not Disturb
6 I am Mobile
7 In Meeting
9 Out of Office
When you connect a Door Lock release device, you must set the Normal Contact Type of the DOP to Normally
Open. Default: Normally Open. See “Digital Output Port (DOP)”.
Alarms
When using the features “Alarms” and “Reminder”, you may need to configure the following parameters, related to
these features.
• Use Alarm with Snooze: When you use Snooze functionality with an Alarm, the system does not consider
an alarm call as answered until the extension user has dialed the acknowledgement code ‘0’ after going
Off-Hook. By default, Snooze is disabled.
• Voice guided Alarm Verification: When this flag is enabled, the system offers the extension users the
option of verifying the alarm they have set using voice-guidance. See “Alarms and Reminders” under
Voice Mail Features.
• Configurable Alarm Type (Once Only / Daily): With this feature, the system gives the Operator and
extension user the choice of setting 'Once Only' or 'Daily' Alarm calls, when they set an Alarm call request.
Clear this check box if you do not want to give a choice. If this option is disabled, the system will allow the
Operator and Users to set Once Only Alarms only. Default: Disabled.
• Configurable Alarm Category (Personalized / Automated): When this feature is enabled, the system
gives the Operator and extension user the choice of setting 'Personalized' or 'Automated' Alarm calls, after
they have configured the Alarm Type. Clear the check box if you do not want to give a choice. If this flag is
disabled, the users can set only Automated alarms.
• Alarm ring Timer (Sec): This is the duration for which the system rings the extension to serve an Alarm
call. By default, the Alarm Ring Timer is set to 45 seconds. The range of this timer is 001 to 255 seconds.
This timer also signifies the duration for which the Operator phone rings to notify that an Alarm call has not
been answered or the extension phone is busy.
• Number of Alarm Attempts: This is number of times the system attempts to place an Alarm call on the
extension phone before notifying the Operator that the call is not answered or the phone is busy. The
Number of Alarm Attempts can be set between 1 and 9. By default, the Number of Alarm Attempts is set to
'3'.
• Alarm Attempt Interval (min): This is the time period between each Alarm Call attempt. The Alarm
Attempt Interval can be set between 1 and 9. By default, the Alarm Attempt Interval is set to 5 minutes.
Distinctive Rings
Distinctive Rings are ringing patterns used for distinguishing between different types of call events, like Internal
Calls, Trunk Calls, Auto Call Back, Auto Redial, Alarm, Emergency call, Priority, etc. If you want to customise the
Ringing pattern, for a call event, select the desired Ring Type. For more details see “Distinctive Rings”.
• Open Jeeves.
• Scroll to System Parameters under Advanced Settings and click the link.
What’s this?
Access to the Eternity is secured at three levels by way of a password:
• at the System Engineer Level with the System Engineer (SE) password.
• at System Administrator Level with the System Administrator (SA) password.
• at the User Level with the User Password.
The System Engineer and the System Administrator passwords secure the system settings from access and
alteration by unauthorized persons (anyone other than the System Engineer and the System Administrator), thus
preventing possible misuse of the features and facilities.
• The SE password is stored in the CPU. If you forget the SE password, the only way to restore the
default SE password is to change the Jumper settings of the CPU card.
• You are advised to record and store the SE password at a safe place, where it can be accessed by you
(the System Engineer) to avoid the inconvenience of restoring the default SE password.
• Enter the New Password. It may be any combination of 4 digits. Valid digits: 0 to 9.
The default SE password will be restored to 1234. You can now enter the programming mode by dialing 1#91-
1234 (the default password). You can also change the SE password again using Jeeves or by dialing a
command as described above.
If you change the default SE password (1234) again after you have reset SE Password (by changing the
Jumper on the CPU card to AB position), the system will not store the new password, until you change the
Jumper back to the default BC position. So, make sure that you have recorded the new SE password in a
safe place, from where it can be retrieved. Change the Jumper back to the default BC position.
• Enter Current Password. If you have not changed the default SA password 1111, enter this code.
• Click SA Password.
User Password
Extension Users can secure their respective extensions from unauthorized use with a password unique to each
extension. The User password too is a combination of any four digits, from 0 to 9. The default User Password is
1111, which each user can change from their respective extensions. Refer the topic “User Password” to know more.
If an extension user forgets the User Password, a new password can be issued to the extension user by the
System Administrator using Jeeves.
• Now, click the desired Extension number on the top bar. To scroll to the desired extension type and
number, click >>.
• Enter the new User Password in the field Change User Password to. The User Password may be a
combination of 4 digits. Valid digits: 0 to 9.
a. locking the Keypad the extension phone. This is possible only on DKP and Extended IP Phones.
b. setting the User Status for the extension as “Absent”; possible on DKP, Extended IP Phones and SLT
extensions. Read the feature description User Absent/Present to know more.
The System Administrator can lock the keypad of DKP and Extended IP Phones, and set users of all extension
types as ‘Absent’ from Jeeves.
• Now, click the desired Extension number on the top bar. To scroll to the desired extension type and
number, click >>.
• Set the user status for the parameter Presence by selecting the option Absent from the combo box. When
you want to change user status again to present, select Present.
• Select the option Lock for the parameter Keypad. This parameter will be configurable only if the selected
extension is a DKP. When you want to remove keypad lock, select Unlock.
• Extension users can also set their status as 'Absent' or 'Present' from their respective extension
phones. Refer “User Absent/Present”.
• DKP extension users can also lock the keypad of their phones from the DKP Phone Menu. Refer
“Digital Key Phone-Operation” for instructions on navigating the phone menu.
• It is also possible to lock/unlock the DKP keypad and set the user extension status as 'Absent'/'Present'
from a remote location using “Direct Inward System Access (DISA)”.
What’s this?
Several features of the ETERNITY NE are based on Timers and Counts. For example, how long and how many
times an extension should ring when Message Wait is set, or how long the Busy Tone, the Ring Back Tone, or the
Error Tone should be played to an extension. ETERNITY NE allows you to configure most of these Timers and
Counts to suit your requirement. Listed below are the Timers and Counts related to the various features and
facilities, along with a brief description and default value of each.
Auto Redial
Auto Redial - Dial Tone The time for which the system waits to sense the 0 to 255 3 seconds
Wait Timer (sec.) dial tone from the CO Network.
Auto Redial - Ring Back The time for which system waits to sense the 0 to 255 60 seconds
Tone Wait Timer (sec.) Ring Back Tone from the CO Network after
dialing the requested number.
Auto Redial - Ring The time for which the extension that has 0 to 255 45 seconds
Timer (sec.) requested Auto Redial should ring.
Auto Redial - Normal The time interval between auto redial attempts 0 to 255 45 seconds
Timer (sec.) when Auto Redial ‘Normal’ is set.
Auto Redial - Normal The number of auto redial attempts the system will 0 to 255 5 tries
Count make when Auto Redial ‘Normal’ is set.
Auto Redial - Priority The time interval between auto redial attempts 0 to 255 10 seconds
Timer (sec.) when Auto Redial ‘Priority’ is set.
Auto Redial - Priority The number of auto redial attempts the system will 0 to 255 20 attempts
Count make when an extension having the feature Auto
Redial Priority in its Class of Service uses Auto
Redial ‘Priority’.
Dial Tone Timer (sec.) The time for which the system plays the Dial tone. 2 to 255 7 seconds
Ring Back Tone Timer The time for which the system plays the Ring Back 1 to 255 45 seconds
(sec.) Tone.
Busy Tone Timer (sec.) The time for which the system plays the Busy Tone. 1 to 255 7 seconds
Error Tone Timer (sec.) The time for which the system plays the Error Tone. 1 to 255 30 seconds
Feature Confirmation The time for which the system plays the 1 to 255 7 seconds
Tone Timer (sec.) Confirmation Tone when a feature is set or
canceled.
Programming Tone The time for which the system plays the 2 to 255 15 seconds
Timer (sec.) Programming Tone when you successfully enter the
SE mode from a phone.
Programming Error The time for which the system plays the Error Tone 1 to 255 3 seconds
Tone Timer (sec.) when you have entered an invalid command string
while configuring a feature from a phone.
Programming The time for which the system plays the 1 to 255 3 seconds
Confirmation Tone Confirmation Tone when a system command is
Timer (sec.) successfully executed when configuring the system
from a phone.
Tone Demo Timer (sec.) The time for which the system plays Call Progress 1 to 255 30 seconds
Tone when you are demonstrating the tone.
DID Inactivity Timer The time after which the system releases the 0 to 255 60 seconds
(sec.) trunk, if the caller has not dialed any digit, or
when a DID or Trunk Auto Answer call is not
answered by the landing destination.
DID Answer Wait Timer The time for which the system waits before 0 to 255 5 seconds
(sec.) answering a DID call.
DID Music Timer (sec.) The time for which the system plays music after 0 to 255 5seconds
answering the DID call.
DID Beeps Timer (sec.) The time for which the system plays beeps to the 0 to 255 10 seconds
caller to prompt the caller to dial the desired
extension number.
DID Ring Timer (sec.) The time for which the system rings on the 0 to 255 30 seconds
landing destination extension in a DID call.
DID Busy Tone Timer In a DID call, the time for which the system plays 0 to 255 15 seconds
(sec.) Busy Tone, if the dialed extension is busy.
DID Error Tone Timer In a DID call, the time for which the system plays 0 to 255 5 seconds
(sec.) Error Tone to the caller, if the caller has dialed an
invalid code.
DISA Idle State Timer In a DISA PIN Authentication call, the time for which 0 to 255 20 seconds
(sec.) the system waits for the caller to go Off-hook after
entering DISA. If the caller does not go Off-hook
within this timer, the system releases the call.
DISA Inactivity Timer In a DISA call, the time for which the system waits 0 to 255 2 minutes
(min.) for the caller to dial digits. If the caller does not dial
any digit within this timer, the system disconnects
the call. This timer is applicable only for Analog
Trunks.
Other Features
Auto Call Back Ring The time for which the extension requesting the 1 to 255 30 seconds
Timer (sec.) Auto Call Back and the destination extension will
ring.
Interrupt Request Timer The time for which the extension on which the 1 to 255 45 seconds
(sec.) Interrupt Request is made will get the beeps.
Barge-In Timer (sec.) The time after which the extension that has 1 to 255 10 seconds
activated Barge-In gets connected to the
extension which is barged in.
Trunk Reservation The time for which a trunk remains reserved for 1 to 255 10 minutes
Timer (min.) the extension that has reserved the trunk.
Transfer while Ringing When an extension transfers a call to another 1 to 255 30 seconds
Timer (sec.) extension after it starts ringing, this is the time for
which the system will wait for the transfer target
extension to answer the call. If the transfer target
does not answer the call within this timer, the call
is returned to the transferror.
Transfer on Busy Timer When a call is transferred to a Busy extension, 1 to 255 30 seconds
(sec.) this is the time for which beeps are played on the
transfer target extension.
Trunk to Trunk Inactivity In a Trunk-to-Trunk call, this is the time for which 1 to 255 2 seconds
Timer (sec.) the system waits after call maturity for any digit to
be dialed. If no digit is dialed within this timer, the
system drops the call.
Call Park Timer (sec.) The time after which the call comes back to the 2 to 255 45 seconds
extension that has parked the call.
Call Park Release Timer The time after which the parked call gets 1 to 255 3 minutes
(min.) disconnected.
Live Call Screening The time for which the speaker of the DKP/ 1 to 255 10 seconds
(sec.) Extended IP Phone extension remains ON while
the message from the caller is being recorded.
Message Wait Ring It is the Number of times the extension should 0 to 255 10 attempts
Count ring after the Message Wait is set on an
extension. This count is applicable only when
‘Ring’ is selected as the Message Wait
Notification type for the extension.
Message Wait Ring The time for which the extension rings to indicate 1 to 255 30 seconds
Timer (sec.) that Message Wait is set for the extension. This
timer is applicable only when ‘Ring’ is selected as
the Message Wait Notification type for the
extension.
Message Wait Ring The time after which the extension should ring 1 to 255 30 minutes
Interval Timer (min.) again to indicate Message Wait is set. This timer
is applicable only when ‘Ring’ is selected as the
Message Wait Notification type for the extension.
Conflict Dialing Timer The time for which the system waits for the 1 to 255 2 seconds
(sec.) extension user to dial the next digit to resolve
conflicting access codes dialed by the extension
user.
Extension - Inter Digit The time for which the system waits for the 2 to 255 7 seconds
Wait Timer (sec.) extension user to dial the next digit. On the expiry of
this timer, the system considers it as the end of
number dialing.
Trunk - First Digit Wait the time for which the system waits for the 1 to 255 25 seconds
Timer (sec.) extension user to dial the first digit, after grabbing
the trunk.
Trunk - Inter Digit Wait When an extension user has grabbed the trunk and 1 to 255 3 seconds
Timer (sec.) is dialing a number, the system waits for the Trunk-
Inter-Digit wait timer for the extension user to dial
the next digit. On the expiry of this timer, the system
considers it as end of number dialing and proceeds
with the call.
Call Hold Retrieval This is the time for which a call put on Global 1 to 999 60 seconds
Timer (sec.) Hold remains connected in the system. If the call
put on Global Hold is not retrieved within this
timer, the call is returned to the DKP/Extended IP
Phone which put it on hold.
RCOC Record Delete This is the time for which the record of the 1 to 999 999 seconds
Timer (min.) outgoing call is stored in the RCOC Table. The
timer is activated whenever a record is stored in
the RCOC table. At the end of this timer, the
system deletes this record from the table.
Release Conference if This is the time for which the system will wait for 1 to 255 2 minutes
Idle for more than (min.) participants of a Dial-In Conference to withdraw or
release themselves from the conference, one-by-
one. On the expiry of this timer, the system releases
the Dial-In Conference and frees the resource
occupied by this conference in the conferencing
circuit.
• Open Jeeves.
• Click the System Timers and Counts link to open the page.
The Timers and Counts on this page are arranged by the name of the feature or function these are related
to.
• Change the value of the Timer or Count by entering the desired duration or count in the respective fields.
What’s this?
The current time zone—Working Hours or Non-working Hours—is displayed on the LCD of the DKP and the
Extended IP Phone.
During Non-working hours the letter ‘N’ is displayed on the LCD of the DKP and the Extended IP Phone in the idle
state.
During working hours, in the idle state, the phone display will look like this:
During Non-working hours, in the idle state, the phone display will look like this:
During Non-working hours, in the idle state, if the extension user has set User Absent and activated Keypad Lock
on the phone, the phone display will look like this:
What's this?
Toll Control (or Toll Restriction) is an expense control feature of ETERNITY NE. It enables you to program the
system so that each extension has a designated calling permission referred to as 'Call Privilege'.
Each type Call Privilege allows the extension to call certain areas and restricts it from calling others. The extension
can also be restricted from the dialing of specific telephone numbers.
• No Calls: Dialing of all external numbers is restricted. Only internal (extension-to-extension) calls are
allowed.
Only the numbers programmed in Global Directory Part I will be allowed to be dialed out, if the directory is
allowed in the “Class of Service (COS)” of the extension.
• Local Calls: Dialing of outgoing calls to Local area numbers, in addition to internal calls, is allowed. It is
possible to restrict calls to certain local numbers. To apply this Call Privilege, you must configure the list of
Local numbers.
• Regional Calls: Dialing of outgoing calls to regional numbers is allowed, in addition to internal and local
calls. It is possible to restrict calls to certain regions. To apply this Call Privilege type, you must configure
the list of Regional numbers.
• National Calls: Dialing of domestic, long-distance numbers within the country is allowed, in addition to
internal and regional calls. You can also restrict calls to certain parts of the country. To apply this Call
Privilege type, you must configure the list of National numbers.
• All Calls: Dialing of all types of numbers—local, regional, national, international—is allowed, without any
restriction.
• Limited Calls: Dialing of only specific Telephone numbers (regional, national or international) is allowed.
By applying this Call Privilege type, you can allow and restrict dialing of telephone numbers starting with a
particular digit, or a particular area code, or certain telephone numbers only. To apply this Call Privilege
type, you must program a list of the Limited numbers that are to be allowed and numbers that are to be
restricted. You can configure three such Limited number lists
Toll Control forms the basis of the features “Dynamic Lock” and “Call Budget”.
Using “Dynamic Lock”, extension users can change the Toll Control (Call Privilege) of their extensions on their own.
The Operator or System Administrator can also change the Toll Control of the extension using Dynamic Lock. To
support this feature, ETERNITY offers fours levels of Toll Control, from 0 to 3.
When the “Call Budget” feature is used on extensions, it becomes necessary to define the calling permission for
extensions that have consumed their allotted budget. To support this feature, ETERNITY offers Toll Control-Call
Budget Consumed.
• Toll Control - Level 0 is Time Zone based, wherein you must define the Call Privilege Type for the Day
(working hours) and the Night (non-working hours). For instance, you may define 'International Calls' as
Call Privilege for the Day and 'No Calls' as Call Privilege for the Night.
By default, Call Privilege ‘All Calls’ is selected for the Day and Night.
• Toll Control - Level 1 is not based on Time Zones. By default, the Call Privilege Type for this level is
'Local Calls'.
• Toll Control - Level 2 is not based on Time Zones. By default, the Call Privilege type set for this level is
'National Calls'.
• Toll Control - Level 3 is not based on Time Zones. By default, Call Privilege 'No Calls' is selected for this
level.
• Toll Control - Call Budget Consumed is applied only if the “Call Budget” feature is enabled on the
extension.
ETERNITY NE offers you the flexibility to redefine the Call Privilege for each of the above Toll Control Levels
according to the requirements of the extension users.
How it works
• When a call is made, the ETERNITY NE checks the Toll Control Level assigned to the extension making
the call.
• The system checks the 'Call Privilege' programmed in the Toll Control Level of the extension.
• For each call privilege type detected, the system will check the following to determine if call is to be
allowed or denied, as summarized in the table below:
Limited Calls Limited Calls Number List (List 1, 2 or 3) assigned to the extension
• The Local, Regional, National, and Limited Calls Number Lists consist of Allowed Numbers and Denied
Numbers.
• matches with Allowed Number list and the Denied Number list.
• matches with Allowed Number list, but not with the Denied Number list.
• matches with neither the Allowed List nor the Denied List.
• The call is restricted, if the dialed number matches with the Denied Number list, but not with the Allowed
Number list.
How to configure
Decide the type of Call Privilege you wish to assign to each extension port type: SLT, DKP, SIP.
For Toll Control to work, you need to configure the lists of Local Numbers, Regional Numbers, National Numbers
and Limited Calls, according to the type of Call Privilege you to assign to the extensions. To do this,
• Make a two-column tables each for Local, Regional, National and Limited Call numbers on paper or using
a computer.
• On one column of each list, write down the numbers you want to permit as Allowed Numbers. On the other
column write down the numbers you want to restrict as Denied Numbers. Your table may look like these:
Allowed Denied
Sr. No.
Numbers Numbers
999
Allowed Denied
Sr. No.
Numbers Numbers
999
Allowed Denied
Sr. No.
Numbers Numbers
999
• Configure these lists on the “Number Patterns” page of Jeeves under Advanced Settings.
• Configure Toll Control levels for each extension port of ETERNITY NE. If you have not already assigned a
Toll Control level to the extension ports at the time of configuring them, you may do so now.
• To select an extension number on the page, select the Extension Number tab.
• For the type of call privilege you selected, the number list configured in the “Number Patterns” page will be
automatically assigned.
If you have not configured number lists already, you may do so now:
• Click the arrow icon. The number list of the call privilege type you selected opens in a window.
• In the Allowed Numbers column, enter the long distance numbers within the country that are to be
permitted.
You may enter only the first digit of the number string, or a part of the string, or the complete number string.
In the Denied Numbers column, enter the long distance numbers that are to be restricted. The number
strings may be only the first digit of the string, a prefix, or the complete string, not exceeding 16 characters.
• Enter as many numbers as you want by clicking the Index number links 001-250, 251-500, 500-750, 751-
999.
• In the Calls allowed during Night box, select the call privilege type for the Night time.
• To configure the number list for the selected call privilege type, click the settings icon. The number list of
the call privilege type you selected opens in a window.
• Follow the same steps to select the call privilege type and configure the number list for
• Calls allowed for Lock Level 1
• Calls allowed for Lock Level 2
• Calls allowed for Lock Level 3
What’s this?
Trunk Auto Answer enables calls landing on a trunk to be answered automatically by greeting the caller with a voice
message before the call is actually handled.
Trunk Auto Answer is useful when you want callers to remain connected until one of the landing destinations
selected for incoming trunk calls becomes free to attend to the caller.
Trunk Auto Answer is useful in call centres, railway enquiry, banks, where callers need to be notified that they
would be attended shortly, so that they do not disconnect the call.
• For all Calls: the system answers all incoming calls landing on the trunk line.
• When Busy: the system answers incoming calls on the trunk, only if the landing destinations are busy.
How it works
Trunk Auto Answer on a trunk works only when Operator or Extension/s is selected as the landing destination for
incoming calls for the Day and Night.
So, you can enable Trunk Auto Answer on a CO, Mobile or SIP trunk, only if you have selected Operator or
Extensions as the destination for incoming calls on that trunk.
ETERNITY NE handles incoming calls on the trunk according to the type of Trunk Auto Answer selected for the
trunk: For all Calls or When Busy
When Trunk Auto Answer–For all Calls is enabled on a CO, Mobile or SIP Trunk, for each incoming call on the
trunk,
• The System answers the with a Greeting message, known as the Trunk Auto Answer Greeting, and rings
the landing destination—Operator or Extensions—selected for the time of the day.
The system starts the DID Inactivity Timer (default: 60 seconds). The Trunk Auto Greeting message is
played once. You may assign a Trunk Auto Answer Greeting of your preference.
• If the landing destination does not answer before the Trunk Auto Answer Greeting message ends, the
system plays Trunk Auto Answer Ring Back Tone message to the caller.
The Ring Back Tone message is played repeatedly for the duration of the DID Inactivity Timer.
However, if no Trunk Auto Answer Ring Back Tone message is assigned, the system will plays Ring Back
Tone to the caller for the duration of this timer.
• If any of the landing destinations answers the call before the expiry of the DID Inactivity Timer, the system
stops the DID Inactivity Timer and the Ring Back Tone message, and connects the caller to the extension
that answered the call.
If no Trunk Auto Answer Busy Bye message is assigned, the system plays the Busy Tone for the duration
of the Busy Tone Timer and releases the trunk port.
When Trunk Auto Answer–When Busy is enabled on a CO, Mobile or SIP Trunk, for each incoming call on the
trunk,
• The System answers the with the Trunk Auto Answer Greeting message and loads the DID Inactivity
Timer.
• The System waits for any of the landing destinations (Operator or Extensions) selected for the time of the
day to be free.
• If no landing destination is free at the end of the Trunk Auto Answer Greeting message, the system plays
Ring Back Tone or Trunk Auto Answer Ring Back Tone message, if assigned, to the caller for the duration
of the DID Inactivity Timer.
• If any of the landing destinations is free before the expiry of the DID Inactivity Timer, the system places the
call on that destination.
• If none of the landing destinations is free at the end of the DID Inactivity Timer, the system plays the Trunk
Auto Answer Busy Bye message, if assigned, and releases the trunk port.
If the Busy Bye message is not assigned, the system will play the Busy Tone to the caller for the duration
of the Busy Tone Timer.
How to configure
For this feature to work, you must do the following:
1. Enable Trunk Auto Answer on the CO Trunk, Mobile, SIP trunk, as required.
• Under Basic Settings, open the links to the desired trunk port type.
• Open the Route Incoming Calls link on the page of the selected trunk number.
• If Operator or Extensions is selected as the destination to Route calls during the Day and to Route
calls During the Night, configure Trunk Auto Answer.
It is not mandatory to configure Trunk Auto Answer for both time zones. You may configure for any time
zone, according to your requirement.
• Select the Trunk Auto Answer Greeting message, the Trunk Auto Answer Ring Back Tone
Message, and the Trunk Auto Answer Busy Bye Message for the Day and Night.
You may select different Greeting, Ring Back Tone and Busy Bye Message for each time zone.
2. Configure the Trunk Auto Answer related Timers, if required. The following Timers are of relevance to the
Trunk Auto Answer Feature:
• The DID Inactivity Timer (default: 60 seconds)
• The Ring Back Tone Timer (default: 45 seconds)
• The Busy Tone Timer (default: 7 seconds)
• You may change the duration of these timers from the“System Timers and Counts” page under
Advanced Settings.
The Ring Back Timer and the Busy Tone Timer are also applicable for the Ring Back Tone and the Busy
Tone played for internal calls.
3. Record and assign Voice Modules for the following Voice Messages related to this feature:
• Trunk Auto Answer Greeting Message.
• Trunk Auto Answer Ring Back Tone Message
• Trunk Auto Answer Busy Bye Message
For each of these messages, you can record four different messages.
If you do not want to use messages, you may select the options: Music-on-Hold or Do not Play any
message.
What’s this?
This feature enables any extension user to reserve a trunk for exclusive use, for a specific time period.
Trunk Reservation can be requested from an SLT extension, a DKP extension and from an Extended IP Phone
extension.
How it works
Let us understand this feature with the help of an example:
In an organization there are four CO trunk lines, CO 1, 2, 3 and 4, but all of these have full traffic throughout the
day.
Extension user A is a sales executive. To complete the sales target, A needs to make long-distance calls to
customers. Since there this full traffic on all the four trunks throughout the day, and these trunks are constantly
busy, A would need a dedicated trunk line to save time and complete the target.
So, A can reserve one of the four trunk lines for the desired duration. To do this,
• Extension A dials the feature access code for Trunk Reservation Access Code for the busy trunk.
• A answers the call and gets connected to the trunk, and gets dial tone.
• The trunk remains reserved for the duration of the Trunk Reservation Timer. This timer is configurable, and
by default is set to 10 minutes. A can have this Timer configured to the desired duration.
• All other extension users who try to access this trunk get error tone, even if this trunk is free.
• If A is finished with the calls before the expiry of the Trunk Reservation Timer, A has two options:
a) release the trunk manually, by cancelling Trunk Reservation.
or
b) wait for the expiry of Trunk Reservation Timer.
• Only when the trunk is released (by A or at the end of the Timer) will other users be able to access it.
See “Basic Settings” for instructions on configuring Class of Service of different extension port types.
If you want to increase or decrease the duration of the Trunk Reservation Timer,
• Change the value of the Trunk Reservation Timer to the desired duration. Default: 010 Minutes. Valid
Range: 001 to 255 minutes.
How to use
For EON & Extended IP Phone Users
OR
To release a reserved Trunk, wait for the Timer to expire, or cancel Trunk Reservation, manually.
OR
• Dial 102
• Lift handset.
• Dial 6 on Busy Tone.
• Replace handset.
What's this?
Extension users may sometimes want to leave their desks, and expecting to return soon, they may not have
forwarded their calls or set Do Not Disturb on their extensions. In such cases, incoming calls will continue to land on
the extension and go unanswered. The callers have no way of knowing that the extension user is not present at the
extension and may try the extension number repeatedly.
With the User Absent/Present feature of ETERNITY, extension users, including the Operator, can set 'User Absent'
when they leave their desks. By doing so, they can block all incoming external as well as internal calls from landing
on their extension. When they return to their desks, they can set 'User Present' and receive incoming calls again.
While Do Not Disturb blocks only internal calls, User Absent can be set when you want to block all
incoming calls (external as well as internal). Thus User Absent can be used as an extension of DND to also
block external calls.
There are more options for indicating availability to other extensions. Refer the topic “Presence” to know
more.
How it works
When an extension user of EON sets 'User Absent', the letter 'A' appears on the phone's display:
EON48P
The letter 'A' disappears when the extension user sets 'User Present'.
When an extension user of EON48/SETU VP248 calls the extension which has set 'User Absent', the text message
'User Absent' will appear on the caller's phone display.
When an SLT extension user calls the extension which has set 'User Absent', callers who dial this extension will get
an error tone.
External callers who call the extension, on which 'User Absent' is set, will get an error tone only.
• Outgoing calls can be made from the extension which has set 'User Absent'. Only incoming calls are
restricted.
• User Password is required for this feature. The default User Password, 1111, will not work. Change the
User Password first.
How to use
The System Administrator can also set an extension as Absent/Present using Jeeves. For instructions, read
“Additional Security to Extension Users” under the topic System Security.
What’s this?
The User Password is a 4-digit code for extension users to protect their extension phones from unauthorized use.
The default User Password is 1111. It can be changed by the extension users from their phones to any desired
value, not exceeding 4 digits.
In case the extension user forgets the password, it can be cleared and restored to the default value 1111 by the
System Engineer (SE) or the System Administrator (SA). Refer the topic “System Security” for instructions.
The User Password is also required to access and use certain features of ETERNITY NE, which are listed below.
• Call Follow Me
• Dynamic Lock
• Direct Inward System Access (DISA)
• Walk-In Class of Service
• User Absent/Present
• Hot Desk
• Phone Settings of the DKP
• Personal Mailbox
The extension user must change the default password for all the above listed features except: Phone Settings,
Mailbox of Voicemail. Both these features allow the extension user to use the default User Password, whereas in
the case of others, the system will not allow feature access without changing the User Password.
In the case of Hot Desking, the default password will work only for one extension involved.
The User Password for an extension can be changed only from that extension phone only.
Since the Mailbox can be accessed using the default User Password, extension users who are assigned a
mailbox are recommended to change their User Password to prevent unauthorized access to their mailbox.
How to use
What’s this?
The Virtual Extension feature of ETERNITY NE enables multiple users to share one telephone instrument as their
extension, yet be considered as individual extensions by the system, with distinct extension properties and class of
service.
Such shared extensions are called Virtual Extensions, as their users do not have individual phones for their use.
Virtual extensions are useful in laboratories, common rooms, dormitories, shop floors, and wherever it is not
feasible to provide dedicated telephone instruments to individual extension users. Virtual extensions allow you
make optimum use of the existing phones without investing in new ones.
How it works
The shared telephone instrument is called the Master Extension. A Master Extension can be an SLT, a DKP or the
Matrix Extended IP Phone.
• Virtual Extensions are assigned to the Master Extension. A Master Extension can have multiple Virtual
Extensions, but a Virtual Extension can have only one Master Extension.
• The Virtual Extension functions as any SLT, DKP or SIP extension of ETERNITY NE. It can be assigned all
features and facilities, like Class of Service, Toll Control, Call Forward, Voice Mail, just like any other
extension of ETERNITY NE.
• All incoming, outgoing, internal and external calls of the Virtual Extensions are recorded in the Station
Message Detail Records.
• To make outgoing calls, the Virtual Extension user must use the feature “Walk-In Class of Service”.
• The Virtual Extension user is logged out of the Master Extension according to the Walk Out mode
assigned to it: Walk out automatically on completion of call or Walk out on user request.
How to configure
To configure Virtual Extensions,
• Open Jeeves.
• Scroll to Virtual Extensions under Advanced Settings and click the link.
• Click Allow Features as per Extension to expand. Select the extension whose features and facilities you
want to assign to this Virtual Extension from the list.
• Click Landing Destination to expand. Select the Master Extension to which you want to assign this Virtual
Extension. Incoming calls for the Virtual Extension will land on this Master Extension, referred to as
Landing Destination.
• In the field DDI Number for SIP Trunk 1, enter the DDI number you want to assign to this Virtual
Extension from the DDI numbers provided by the ITSP with whom you have registered SIP Trunk 1.
• In the field DDI Number for SIP Trunk 2, enter the DDI number you want to assign to this Virtual
Extension from the DDI numbers provided by the ITSP of SIP Trunk 2.
• Similarly, in the fields DDI Number for SIP Trunk 3 and DDI Number for SIP Trunk 4 enter the DDI
numbers you want to assign to this Virtual Extension from the DDI numbers provided by the ITSP of
SIP Trunk 3 and 4 respectively.
• To configure another Virtual Extension, click the Virtual Ext. <number> tab. Follow the same instructions
as above to configure the extension.
How to use
For making outgoing calls users of Virtual Extensions must use “Walk-In Class of Service”.
What’s this?
Voice Help is a recorded voice message that can be used for providing quick help or guidance to extension users.
You can use Voice Help to announce important phone numbers or access codes of frequently used features, and
similar functions. Extension users can access Voice Help by dialing its feature access code and listen to the
recorded help message.
How to configure
To configure Voice Help, first, you need to record a Voice Module with the desired help message, and then assign
the recorded Voice Module to the Voice Help application. For instructions, read the topic “Voice Message
Applications”.
The Voice Module on which you will record the message has a duration of 16 seconds (maximum). The length of
your help message must be within this limit, or the message will be truncated when you record.
How to use
For EON & Extended IP Phone Users
• Dial 1090
• Listen to message
• Press any key to stop.
What’s this?
ETERNITY NE allows you to record different voice messages which can be played to callers/extension users for
specific situations. For example, if Trunk Auto Answer or Direct Inward Dialing are enabled on a trunk line, you can
configure the system to play Voice Messages to greet and guide callers. When a feature like Do Not Disturb is set
you can configure the system to play a DND notification to the callers. You can also configure the system to play
Voice Messages instead of Tones like the Error Tone, Busy Tone or Ring Back tone.
How it works
The voice messages are recorded in Voice Modules and the voice modules are assigned to the features/
applications for which they are to be used.
The ETERNITY NE supports 16 Voice Modules of a maximum duration of 16 seconds each. You can record a short
message of a maximum 16 seconds each or less in a Voice Module.
When the recorded voice modules are assigned to the features/applications, they are played to the callers/
extension users whenever the feature/application is activated.
Depending on the feature/application, the voice message recorded in the voice modules are played once or
continuously.
At a time, the system can play four voice modules simultaneously to external callers/extension users.
Voice messages can be used for different applications as described in the following.
For example:
• DID Greeting Message for non-working hours: "Welcome to Cotton Software. Sorry we are closed.
• DID Wrong Dial Message: "Sorry you have dialed an invalid number."
• DID Ring Back Tone (RBT) Message: "The number you dialed is ringing.”
• DID No-Dial Message: "Sorry you have not dialed any number".
• DID Call Transfer Message: "Transferring the call to the Operator." This message is played when the
caller does not dial any number and the call is transferred to the Operator.
You are recommended to record the message "Please press 0 to acknowledge." in the voice module for
the Alarm/Reminder message, so that extension users can acknowledge Snooze calls. Refer the topics
“Alarms” and “Reminder” to know more.
• Security/Emergency Message: A voice message is played to the external number and to the operator
extension which answers the Emergency call.
• Voice Help: A voice message can be recorded for providing quick help or guidance to extension users.
You can use Voice Help to announce important phone numbers or access codes of frequently used
features, and similar functions.
• Music-on-Hold: Callers who are put on hold are usually played music as they wait. You can play a voice
message instead of music to the callers.
• Message Wait - Notification: A voice message can be played to extension users to inform them of the
new messages in their mailbox, when the extension users go off-hook.
• DND Notification: A voice message can be played to extension users who try to call an extension that has
set DND.
When you record a message for the Dial Tone, make sure that the length of the message is shorter than
the duration of the Dial Tone Timer.
When you record the Ring Back Tone message, make sure that the length of the message is shorter than
the duration of the Ring Back Tone Timer.
When you record the Busy Tone message, make sure that the length of the message is shorter than the
duration of the Busy Tone Timer.
Voice Message Notification for DND is particularly useful when the extension phone on which DND is set is an SLT
and the callers too are using SLT.
Similarly, a voice message can be played to the extension user who attempts to invoke a feature that is not
included in the Class of Service (COS) of the extension.
These time based greetings are played to callers before the DID Greetings Message on a DID enabled trunk. On
an Auto-Answer enabled trunk these messages are played before the Trunk Auto Answer greeting message.
The Time Based Greetings will be played during following time periods:
Voice
Module Voice Message Application Voice Message
Number
01 Music-On-Hold
06 DID Welcome Greeting for Night time (Non-working Welcome! I am sorry, we are closed.
hours)
08 DID - No Dial message Sorry! You have not dialed any number.
11 DID - Destination Ringing message (Ring Back The number you have dialed is ringing.
Tone)
12 DID - Destination No Reply message The person you dialed is not responding.
13 DID Call Transfer to Operator message Please hold, transferring your call to the Operator.
If these default Voice Modules and Voice Messages serve your purpose, you may use them. No further
configuration is required, as these modules are loaded in the system by default.
However, if you want to use custom messages, you must first record the messages on voice modules.
For example, if you want to a different voice message for the DID Welcome Greeting for Night Time, record a
suitable message of you choice on Voice Module number 06.
If you want to use a voice Message for DND Notification, you may record a suitable message on a blank module
and assign this module to the DND Notification application.
You can reassign the default Voice Modules to other Voice Message Applications as required.
The same Voice Module can also be assigned to multiple Voice Message Applications.
Or
• Any other source, and upload the audio files of the Voice messages on to the system configuration files,
using the embedded FTP server of ETERNITY NE.
• At the end of the dial tone, start recording the message, by speaking into the mouthpiece of telephone
instrument.
• Limit your message to 16 seconds or less or the message will be truncated when played.
To check or verify the Voice Message you recorded, dial (from SE Mode):
• Exit SE mode.
• Replace handset.
If you are not satisfied with the quality of the recording, you may repeat this procedure again.
Voice Module 01 is reserved for Music-on-Hold by default. You are advised not to assign this module to
any other Voice Message Application.
• Make sure the audio files are recorded in .wav file format, with the attributes listed below:
• Upload the voice messages from the embedded FTP server of ETERNITY NE.
• Click the Click here to Upload Voice Messages in Voice modules link.
• The FTP window opens, containing all the voice module folders. Each Voice Module Number has a
separate folder. The folder name indicates the Voice Module number.
• Click the folder to view the existing audio files in the folder.
• To upload a Voice Message on a Voice module, right click the desired Voice Module folder and select the
option Open Link in FireFTP from the right click menu.
• Browse for the folder where you have saved the Voice Modules, and click OK.
• To copy the audio file from the location it is saved on the computer, select the file, and click the Upload
Arrow.
The voice message you assigned to the Voice Module will be uploaded.
• Now, assign the Voice Module Number to the desired Voice Message application.
You can assign the same Voice Module to more than one Voice Message Application.
• Click Submit.
• The Voice Module Number you assigned, and the duration of the Voice Message recorded in the module,
appear.
ETERNITY NE automatically detects and displays the duration of the voice message you recorded. So you
need not define the duration.
However, you may define the duration, if you want the recorded voice message to be played for a specific
duration. For example, the message you recorded in the voice module 14 for DND Notification is 16
seconds long, but you want to play only the message contents of the first 12 seconds, you can define the
duration of the message as 12 seconds.
If you do not have FireFTP Add-on in your browser, you may upload files using Windows FTP. Follow these steps:
• Click My Computer.
• Type the current IP Address of the Ethernet Port of ETERNITY NE in the Address bar as ftp://
192.168.1.101
• In the Password field, enter the SE password (default 1234) and click the Log On button.
• Open the voicemodule folder. All the Voice Module folders appear in the window.
• Open the desired voice module folder and delete the existing audio file.
• Copy the new audio file you recorded from the location it is saved on the computer. Paste the file in the
Voice Message folder.
• Return to the Voice Message Applications page of Jeeves and assign the Voice Module Number to the
desired Voice Message Application.
What’s this?
Every extension of ETERNITY NE is allowed a distinct Class of Service and Toll Control defining its access to
features and its calling permission.
Extension users may be required to make calls from another extension, which does not have the same Class of
Service and Toll Control as their own extension.
With Walk-In Class of Service, all extension users of ETERNITY NE can make calls from any other extension of the
system as per the Class of Service and Toll Control of their own extension.
This feature is particularly useful to extension users who frequently move away from their desk. It allows them the
same level of feature access and Toll Control on the other extensions from which they make calls.
Extension users can 'Walk-In' in to any extension port—DKP, SLT and SIP.
• The extension user is automatically logged out from the extension into which the user has walked in after
the call made by the user is completed.
• The extension user remains 'walked-in' until the user dials the feature command for Walk-Out, or until a
second person walks in to the same extension port.
You can allow extension users either of the above types of Walk-In by configuring the Walk-Out Mode in their
extension port parameters, as:
To be able to use this feature, extension users must first change the default User Password 1111.
How it works
• Extension user A has a DKP with the number 301. A has long distance calling facility on the extension.
• Extension user B has an SLT with the number 201. B can make only local calls from the extension.
• Extension user A is at B's desk and needs to make a long distance call.
• Extension user A can walk into B's extension number 201 by dialing
• the feature code for 'Walk-In Class of Service'.
• the User Password (default password 1111 will not be accepted).
• Extension user A can now dial the trunk access code and the external, long distance number.
• If Extension user A has 'Walk out automatically on completion of call' set as the Walk-Out Mode on
extension 301, A will be walked out when the current call ends or if A goes ON-Hook at any time after
walking in to extension 201.
• If Extension user A has 'Walk out on user request set as the Walk-Out Mode on extension 301, A can walk
out of 201 only by dialing the feature code for Walk-Out.
However, if Extension user A does not walk out by dialing the feature code, the system will walk out A only
when another extension user walks in to extension 201.
• Calls made by Extension user A from extension 201 using Walk-In will be calculated and charged to
extension 301.
• Call record details of calls made by Extension user A from extension 201 using Walk-In will be recorded in
the Station Message Detail Record of extension 301.
How to configure
This feature is available to all extensions of ETERNITY NE. It does not require any other configuration except
selection of the Walk-Out Mode for the extension in its port parameters.
If you want to allow different walk-out modes to different extensions, then decide which of the extensions are to be
allowed 'Walk out automatically on completion of call' and which are to be allowed 'Walk out on user request'.
• To select an extension number on the page, select the Extension Number tab.
To walk out by dialing a command from the extension you walked in:
When your own extension has 'Walk out automatically on completion of call’ as the Walk-Out mode, you
will be walked out if you go ON-Hook before you make the call. You need to walk in again.
To walk out by dialing a command from the extension you walked in:
• Dial 111-0
• You get a confirmation tone.
• Replace the handset.
What’s this?
The Voice Mail System of ETERNITY NE offers voice-guided Alarms and Reminders, which can be set by the
Operator as well as extension users.
How it works
Voice-guided Alarm and Reminder requests are served as per the date and time set by extension users. The
different ways in which Alarm or Reminder requests will be served are described in the following:
• When user press '0', VMS prompts: "Your Alarm is Acknowledged." (Acknowledge.wav)
• When user press '0', VMS prompts: "Your Alarm is Acknowledged." (Acknowledge.wav)
How to configure
The VMS allows you to enable/disable the Alarm Verification for alarms and reminders, allowing extension users
who want to use alarms and reminders to confirm
• Time set for an alarm
• Date and time set as a reminder.
When Alarm Verification is disabled, the VMS will not confirm the alarm and reminder set by the extension user.
• Open Jeeves.
• Select the Voice Guided Alarm Verification check box to enable Alarm Verification. Default: Disabled.
How to use
• If no time is entered, VMS prompts: "You have not entered any input" (NoInput.wav)
• If invalid time is entered, VMS prompts: "You have entered invalid input." (InvalidInput.wav)
• To set alarm, dial valid time VMS prompts: "To set once, Press '1', To set Daily Press '2'."
(SetOnceDaily.wav)
Once Only
• Dial 1 VMS responds with: "You have set Wake up Alarm at …." (WakeupVeri.wav) followed by the
prompt: To Confirm67, Press 1, To Re-enter, Press 2." (AlarmConf.wav).
• Dial 1 to confirm the time set for alarm VMS responds with: "Your Wake up Alarm is set."
(WakeupSet.wav) followed by the prompt: "Thanks for using this Service." (Thankservice.wav)
• If alarm is not set, the VMS responds with: "Sorry! Your Wake Up Alarm cannot be set. Please call
Operator for further assistance." (AlarmNoset.wav). VMS further responds with: "Thanks for using this
Service." (Thankservice.wav)
Daily Alarm
• Dial 2 VMS responds with: "You have set Daily Wake up Alarm at …." (DailyWakeupVeri.wav) followed
by the prompt: "To Confirm, Press 1, To Re-enter, Press 2." (AlarmConf.wav)
• Dial 1 to confirm the time set for alarm VMS responds with: "Your Daily Wake up Alarm is set."
(DailyWakeupSet.wav) followed by the prompt: "Thanks for using this Service." (Thankservice.wav)
• If no alarm is set, the VMS responds with: "Sorry! Your Wake Up Alarm cannot be set. Please call
Operator for further assistance." (AlarmNoset.wav). The VMS further responds with: "Thanks for using
this Service." (Thankservice.wav)
• Dial # (pound/hash) to cancel all alarms VMS responds with: "Your all Wake up Alarm are cancelled."
(WakeupCancel.wav) followed by the prompt: "Thanks for using this Service." (Thankservice.wav)
• If no alarms are set, the VMS responds with: "Sorry! There is no Alarm to cancel." (Alarmnocancel.wav)
followed by the prompt: "Thanks for using this Service." (Thankservice.wav).
66. The Date and time format depends on the Region/Country selected for the system.
67. This option will not be played if Alarm Verification is disabled in the System Parameters.
• Dial 034 the VMS prompts: "Enter the Extension number for which you have to set or cancel Wake Up
Alarm." (RemoteExt.wav)
• Dial 1 to select the extension user for which the Alarm is to be set. VMS responds with: "Enter the time, HH
MM in twenty-four hour format. To cancel all alarms, press '#’ (pound/hash)'." (EntertimeI.wav)
• If no time is entered, the VMS prompts: "You have not entered any input" (NoInput.wav)
• If invalid time is entered, the VMS prompts: "You have entered invalid input." (InvalidInput.wav)
• To set alarm, dial valid time VMS prompts: "To set once, Press '1', To set Daily Press '2'."
(SetOnceDaily.wav)
Once Only
• Dial 1 the VMS responds with: "To set it as Personal, Press 1. To set it as Automated, Press 2."
(Alarmmode.wav)
• Dial 1 VMS responds with: "You have set Personal Wake up alarm at…." (PerWakeupVeri.wav)
followed by the prompt: "To Confirm, Press 1, To Re-enter, Press 2." (AlarmConf.wav)
• Dial 1 VMS responds with: "Your Personal Wake up Alarm is set." (PerWakeupSet.wav) followed by
the prompt: "Thanks for using this Service." (Thankservice.wav)
• If alarm is not set, the VMS responds with: "Sorry! Your Wakeup Alarm cannot be set. Please call
Operator for further assistance." (AlarmNoset.wav) VMS further responds with: "Thanks for using this
Service." (Thankservice.wav)
OR
• Dial 2 the VMS responds with: "You have set Automated Wake up alarm at…."
(AutoWakeupVeri.wav) followed by the prompt: "To Confirm, Press 1, To Re-enter, Press 2."
(AlarmConf.wav)
• Dial 1 the VMS responds with: "Your Automated Wake up Alarm is set." (PerWakeupSet.wav)
followed by the prompt: "Thanks for using this Service." (Thankservice.wav)
• If alarm is not set, VMS responds with: "Sorry! Your Wakeup Alarm cannot be set. Please call
Operator for further assistance." (AlarmNoset.wav) VMS further responds with: "Thanks for using
this Service." (Thankservice.wav)
Daily Alarm
• Dial 2 VMS responds with: "To set it as Personal, Press 1. To set it as Automated, Press 2."
(Alarmmode.wav)
• Dial 1 VMS responds with: "You have set Daily Personal Wake up alarm at…."
(DailyPerWakeupVeri.wav) followed by the prompt: "To Confirm, Press 1, To Re-enter, Press 2."
(AlarmConf.wav)
• If the alarm is not set, the VMS responds with: "Sorry! Your Wakeup Alarm cannot be set. Please
call Operator for further assistance." (AlarmNoset.wav) VMS further responds with: "Thanks for
using this Service." (Thankservice.wav)
OR
• Dial 2 the VMS responds with: "You have set Daily Automated Wake up alarm at…."
(DailyAutoWakeupVeri.wav) followed by the prompt: "To Confirm, Press 1, To Re-enter, Press 2."
(AlarmConf.wav)
• Dial 1 VMS responds with: "Your Daily Automated Wake up Alarm is set." (DailyPerWakeupSet.wav)
followed by the prompt: "Thanks for using this Service." (Thankservice.wav)
• If alarm is not set, VMS responds with: "Sorry! Your Wakeup Alarm cannot be set. Please call
Operator for further assistance." (AlarmNoset.wav) VMS further responds with: "Thanks for using this
Service." (Thankservice.wav)
• Pick up handset of your telephone and dial 164 VMS prompts: "Enter the Date in DD MM YYYY
format68. To Cancel all Reminders, Press '#’ (pound/hash)'. For example, To enter Date 17th March 2008,
Dial One Seven Zero Three Two Zero Zero Eight." (AlarmDateI.wav)
• If no date is entered then VMS prompts: "You have not entered any input" (NoInput.wav)
• If invalid date is entered then VMS prompts: "You have entered invalid input." (InvalidInput.wav)
• Dial valid Date the VMS prompts: "Enter the time, HH MM in twenty four hour format." (Entertime2I.wav)
• If no time is entered, the VMS prompts: "You have not entered any input" (NoInput.wav)
• If invalid time is entered, the VMS prompts: "You have entered invalid input." (InvalidInput.wav)
• Dial valid time VMS prompts: "You have set Reminder for..." (ReminderVeri.wav) followed by the
prompt: "To Confirm, Press 1, To Re-enter, Press 2." (AlarmConf.wav)
• Dial 1 to confirm the date and time set for Reminder the VMS responds with: "Your Reminder is set."
(ReminderSet.wav) followed by the prompt: "Thanks for using this Service." (Thankservice.wav)
• If Reminder is not set, the VMS responds with: "Sorry! Your Reminder cannot be set. Please call
Operator for further assistance." (ReminderNoset.wav) VMS further responds with: "Thanks for using
this Service." (Thankservice.wav)
• Dial # (pound/hash) to cancel Reminder the VMS responds with: "Your Reminder is cancelled."
(ReminderCancel.wav) followed by the prompt: "Thanks for using this Service." (Thankservice.wav)
• If no reminder is set, the VMS responds with: "Sorry! There is no Reminder to cancel."
(Remindernocancel.wav) followed by the prompt: "Thanks for using this Service." (Thankservice.wav)
68. The date format in the prompt will be MM DD YYYY, if you selected USA as the Region/Country for your system.
• Dial 1072 to enter SA mode followed by 035 VMS prompts: "Enter the Extension number for which you
have to set or cancel Reminder." (RemoteExtRem.wav)
• Dial 1 to select the station user for which the Reminder is to be set. VMS responds with: "Enter the Date in
DD MM YYYY format. To Cancel all Reminders, Press '# (pound/hash)'. For example, To enter Date 17th
March 2008, Dial One Seven Zero Three Two Zero Zero Eight." (AlarmDateI.wav)
• If no date is entered, the VMS prompts: "You have not entered any input" (NoInput.wav)
• If invalid date is entered then VMS prompts: "You have entered invalid input." (InvalidInput.wav)
• Dial valid date VMS prompts: "Enter the time, HH MM in twenty-four hour format." (Entertime2I.wav)
• If no time is entered then VMS prompts: "You have not entered any input" (NoInput.wav)
• If invalid time is entered then VMS prompts: "You have entered invalid input." (InvalidInput.wav)
• Dial valid time VMS prompts: "To set it as Personal, Press 1. To set it as Automated, Press 2."
(Alarmmode.wav)
• Dial 1 VMS responds with: "You have set Personal Reminder for…." (PerReminderVeri.wav)
followed by the prompt: "To Confirm69, Press 1, To Re-enter, Press 2." (AlarmConf.wav)
• Dial 1 VMS responds with: "Your Personal Reminder is set." (PerReminderSet.wav) followed by the
prompt: Thanks for using this Service." (Thankservice.wav)
• If Reminder is not set, the VMS responds with: "Sorry! Your Reminder cannot be set. Please call
Operator for further assistance." (AlarmNoset.wav) VMS further responds with: "Thanks for using
this Service." (Thankservice.wav)
OR
• Dial 2 VMS responds with: "You have set Automated Reminder for…." (AutoReminderVeri.wav)
followed by the prompt: "To Confirm, Press 1, To Re-enter, Press 2." (AlarmConf.wav)
• Dial 1 VMS responds with: "Your Automated Reminder is set." (PerReminderSet.wav) followed by
the prompt: "Thanks for using this Service." (Thankservice.wav)
• If Reminder is not set, VMS responds with: "Sorry! Your Reminder cannot be set. Please call
Operator for further assistance." (AlarmNoset.wav) VMS further responds with: "Thanks for using
this Service." (Thankservice.wav)
• Dial # to cancel Reminder VMS responds with: "Your Reminder is cancelled." (ReminderCancel.wav)
followed by the prompt: "Thanks for using this Service." (Thankservice.wav)
69. This option will not be played, if Alarm Verification is disabled in the System parameters.
What’s this?
Broadcasting Message allows you to send the same message to all extension users having voicemail, at the same
time. You can use Broadcast Message to make general announcements like hosting of an event, an unplanned day
off, and other such activities or events.
How to use
To Broadcast Message,
• Enter SA Mode.
• Dial 1072-301
• VMS prompts: "Record your message after the beep and press any digit to end". (Recmsg.wav)
• Speak to record your message after the beep, and press # (hash/pound) to end the message.
• VMS prompts: "To re-record the message press 1, to confirm press 2". (RecAgain.wav)
• Dial 2 to confirm VMS responds with "Your message has been sent." (Msgsent.wav)
• If you press '1', the VMS prompts: "Record your message after the beep and press any digit to end"
(Recmsg.wav). Follow the prompts.
• If “Message Verification” is disabled, the VMS will not offer to verify and re-record your message. Your
message will be sent to all mailboxes and the VMS will respond with ‘Your message has been sent’
(Msgsent.wav).
The length of the message you want to broadcast must be equal to or less than the minimum of message
length programmed for the mailboxes, or your message will be truncated. For instance, if the Maximum
message length for a mailbox is configured as 15 seconds, maximum length of the message to be
broadcast must be less than or equal to 15 seconds. If the broadcast message exceeds this limit, the
system will play the first 15 seconds and truncate the remaining part of the message.
What’s this?
The VMS Auto Attendant answers calls of external callers and extension users (referred to here as ‘callers’) and
transfers the call to the extensions according to the Call Transfer type set for the extension.
The VMS Auto Attendant offers the following types of Call Transfer, which can be set on the extensions for the Day
and Night:
• Transfer to Mailbox: When the caller dials the extension number, the VMS Auto Attendant checks if the
extension number has a mailbox assigned and transfer the call to the mailbox of the extension.
• Transfer immediately: When the caller dials the extension number, the VMS Auto Attendant transfers the
call on the extension without checking whether it is busy or free.
• Transfer when extension rings: When the caller dials the extension number, the VMS Auto Attendant
waits for the extension to start ringing and then transfer the call.
If the extension is busy the VMS Auto Attendant transfers the call to the mailbox of the extension, if
assigned. If no mailbox is assigned, the VMS Auto Attendant takes the caller back to the Home Node.
• Transfer when extension answers: When the caller dials the extension number, the VMS Auto Attendant
transfers the call when the extension answers (goes OFF-Hook).
If the extension does not answer70, the VMS Auto Attendant transfers the call to the mailbox of the
extension. If no mailbox is assigned to the extension, the VMS Auto Attendant takes the caller back to the
Home Node.
• Transfer when extension permits: The VMS Auto Attendant prompts the caller to record his/her name. It
puts the caller on hold and places the call on the desired extension. If the extension is free and answers
the call, the VMS announces the caller’s name to the extension user and prompts the extension user to
choose whether or not to speak to the caller. If the extension user chooses to talk, the VMS transfers the
call.
If the extension user chooses not to talk, the VMS transfers the call to the mailbox of the extension user
and asks the caller to leave a message.
If no mailbox is assigned to this extension user, the VMS Auto Attendant takes the caller back to the Home
Node.
70. The VMS will wait for the duration of the Wait for Answer Timer (default: 15 seconds; the timer is configurable). If the call is not
answered before this timer expires, it is treated as No Reply.
To select Call Transfer Type in the Voice Mail Auto Attendant parameter on the extensions and Department
Groups, for instructions:
For Department groups, see the feature description for “Department Call” for instructions.
What’s this?
The VMS Auto Attendant allows external callers and extension users to reach directly the desired person in an
organization by dialing the extension number of that person at the home position.
How to use
External Callers
• The VMS Auto Attendant answers the call.
• The VMS greets the caller with the Greeting message followed by the Welcome Message: "Welcome!
Please dial the extension number Or to dial by name press 7. To leave message, press 6. To go to
operator, press 9. For more options, press 0. To disconnect, press #."
• The caller dials valid extension number VMS transfers the call as per the Call Transfer Type selected for
the extension.
• If the extension number dialed by the caller is invalid, the VMS prompts: "The number is not valid."
(Invalno.wav) followed by the prompt: "Please dial the extension number or to dial by name press 7. To
leave message, press 6. To go to operator, press 9. For more options, press 0. To disconnect, press #".
Extension Users
• VMS responds with a greeting followed by Welcome Message: "Welcome! Please dial the extension
number Or to dial by name press 7. To leave message, press 6. To go to operator, press 9. For more
options, press 0. To disconnect, press #.
• Dial valid extension number VMS transfers the call as per the Call Transfer type selected for the dialed
extension number. Talk.
• If the extension number you dialed is invalid, the VMS prompts: "The number is not valid." (Invalno.wav)
followed by the prompt: "Please dial the extension number or to dial by name press 7. To leave
message, press 6. To go to operator, press 9. For more options, press 0. To disconnect, press #" .
What’s this?
The VMS Auto Attendant allows external callers and extension users to reach the desired person in an organization
by dialing the name of that person. This feature useful when caller/extension user cannot recall the extension
number of the person they want to speak to.
How to configure
For this feature to work, each extension user’s name must be abbreviated and configured on the extension. It is
recommended that the extension users’ names be abbreviated to the first three letters of the name. As far as
possible, abbreviate names such that no two names are the same.
You must configure the Abbreviated Name in the Voice Mail Auto Attendant settings of the SLT, DKP, and SIP
extensions.
• To configure Abbreviated Name for SLT extensions, see “Voice Mail Auto Attendant” under “SLT
Extensions” for instructions.
• To configure Abbreviated Name for DKP extensions, see “Voice Mail Auto Attendant” under “DKP
Extensions” for instructions.
• To configure Abbreviated Name for SIP extensions, see “Voice Mail Auto Attendant” under “SIP
Extensions” for instructions.
How to use
External Callers
• The VMS Auto Attendant answers the call. The VMS greets the caller with the Greeting message followed
by the Welcome Message: "Welcome! Please dial the extension number Or to dial by name press 7. To
leave message, press 6. To go to operator, press 9. For more options, press 0. To disconnect, press #."
• The caller dial 7 VMS prompts: "Please enter first three letters of the name." (DialName.wav)
• The caller dials valid digits VMS prompts: "To confirm press '1', to cancel press '2'." (Xfromot.wav)
• The caller dials 1 VMS transfers the call as per the transfer type of the selected station. Talk.
• If the caller dials invalid digits, the VMS prompts: "Sorry no match found." (Empty.wav) followed by the
prompt: "Welcome! Please dial the extension number Or to dial by name press 7. To leave message,
press 6. To go to operator, press 9. For more options, press 0. To disconnect, press #".
Extension Users
• When you call the VMS by dialing 3931,
• VMS responds with a greeting followed by Welcome Message: "Welcome! Please dial the extension
number Or to dial by name press 7. To leave message, press 6. To go to operator, press 9. For more
options, press 0. To disconnect, press #."
• Dial 7 VMS prompts: "Please enter first three letters of the name." (DialName.wav)
• Dial valid digits VMS prompts: "To confirm press '1', to cancel press '2'." (Xfromot.wav)
• Dial 1 VMS transfers the call as per the transfer type of the selected extension. Talk.
• If the digits you dialed are invalid, VMS prompts: "Sorry no match found." (Empty.wav) followed by the
prompt: "Welcome! Please dial the extension number Or to dial by name press 7. To leave message,
press 6. To go to operator, press 9. For more options, press 0. To disconnect, press #".
Extension users can use Dial by Name also to reach another extension, if the called extesnion is busy or
does not reply.
What’s this?
The VMS supports E-mail Based Notification to inform extension users about the arrival of new messages in their
mailbox. Extension users can also receive new messages as attachments to the email.
How to configure
To be able to use this feature, extension users must have Email-based notification enabled in Voice Mail Settings.
• Open Jeeves.
• On this page, select the extension number to which you want to provide this feature, by clicking the
extension number tab.
• Select the desired notification option: Send without attachment or Send with attachment. Default:
Do not send.
• In the E-mail Address field, enter the email address of the extension user to which the notification for
new messages should be sent.
• To configure the same feature on another extension, click the desired extension number tab, and follow
the same steps as described above.
What’s this?
The VMS enables extension users to forward messages of their mailbox to other mailboxes.
How it works
The Forwarding Messages feature of the VMS offers to extension users the following options:
• forward messages after adding a comment.
• forward messages without adding comment.
• forwarding messages with Message Read Receipt request.
Before forwarding a message, the VMS asks the Sender, if the Sender needs a confirmation that the message has
been read by the Recipient.
If the Sender requests for 'Message Read Receipt', the VMS remembers stores this request. When the Recipient
reads the message, the VMS generates a file containing the first 5 seconds of the message that was sent by the
Sender and delivers it to the Sender's mailbox in the form of a new message with a prompt: "This message was
read by <Extension Name><5 seconds of the message sent>" with the Date and Time (if this is enabled in Play
Message Details) at which the message was read.
In case the message was not delivered to the Recipient, the VMS generates a file containing the first 5 seconds of
the message that was sent by the Sender and delivers it to the Sender's mailbox in the form of new message with
the prompt: "This message was not delivered to <Extension Name><5 seconds of the message>" with the Date
and Time (if enabled in Play Message Details).
How to use
• Call the VMS by dialing 3931.
• VMS responds with: "You have <n> new messages" followed by "Enter your mailbox password"
(Enterpwd.wav)
• Enter your mailbox password VMS responds with: "You have 0/n new messages". (Nonewmsg.wav/
Newmsg.wav)
• The VMS prompts: "To listen to new messages press '1', to listen to old message press '2', to send a
message press '3', to change your mailbox settings press '4', to go to home position press '0'".
(Mmmm.wav)
• Dial 1 or Dial 2 and listen to the messages VMS prompts: "To replay the message press '1', for Date
and Time stamp press '2', to delete the message press '3', to play the next message press '4', to forward
the message press '5', to save the message as new press '6', to go to previous menu press '0'."
(Mmsmo.wav)
• Dial 5 VMS prompts: "To forward the message without comment press '1', to forward the message with
comment press '2', to go to previous menu press '0'." (Fwdhow.wav)
OR
• To forward message with comment dial '2'. The VMS prompts: "Enter the Destinations". (Msgdest.wav)
• The VMS prompts: "Record your message after the beep and press any digit to end." (Recmsg.wav)
• Speak to record your comment and press # to end recording. VMS prompts: "To re-record the
message press '1', to confirm press '2'." (RecAgain.wav) Dial 2 to confirm.
• The VMS prompts: "To request read receipt press '1', to ignore read receipt press '2'." (Askcfrm.wav)
• Dial 1 or 2. VMS responds with "Your message has been sent". (Msgsent.wav)
Extension users must be careful in dialing destination numbers. If invalid destination is entered then the
VMS will clear all the entries and will ask the mailbox owner to re-enter all the destinations again.
What’s this?
A General mailbox is a shared mailbox between extension users. The General Mailbox is used for recording
messages when the mailbox of an extension is full.
How it works
The VMS offers the following options to extensions when their mailbox is full:
• Not offer the caller to record a message.
• Overwrite the existing messages in the mail box with the new message.
• Deliver the new message to the General Mailbox.
If you configure Delivery of new messages to General Mailbox on an extension, whenever the mailbox of the
extension is full, the VMS will offer the caller to record a message. This message will be recorded in the General
Mailbox.
The extension user, whose mail box is full, can listen to the new message by accessing the General Mailbox.
The extension user can access the General Mailbox, only if this feature is enabled in the Class of Service of the
extension.
How to configure
To offer extension users the facility of the General Mailbox when their mailbox is full, you must do the following:
• Configure the option When Mailbox is full in the Voicemail Settings of the extension.
For the option When Mailbox is full, select Deliver New Message in General Mailbox.
• Enable the feature General Mailbox in the “Class of Service (COS)” of the extension.
For instructions on configuring these parameters on the different extension types, see “SLT Extensions”, “DKP
Extensions”, “SIP Extensions” under “Basic Settings”.
How to use
What’s this?
The VMS allows,
To leave a message, the called extension must have a mailbox. The length of message recorded by the callers/
extension users must not exceed the message length set for the called extension’s mailbox. If the message
recorded by the callers/extension users exceeds the message length set for the called extension’s mailbox, the
VMS will stop recording the message after the time set and save the partially recorded message.
How to use
External Callers
• The VMS Auto Attendant answers the call. The VMS greets the caller with the Greeting message followed
by the Welcome Message: "Welcome! Please dial the extension number Or to dial by name press 7. To
leave message, press 6. To go to operator, press 9. For more options, press 0. To disconnect, press #."
• The caller dials 6 to leave message VMS prompts: "Enter the Extension number for which you wish to
leave message." (LeavemsgE.wav)
• The caller dials the desired extension number VMS prompts: "Record your message after the beep and
press # (hash/pound) to end. (Recmsg.wav)
• If no mailbox is assigned to the dialed extension number, the VMS prompts: "Mailbox not assigned. To
disconnect, press 1. To go to Home Position, Press 0." (NoMailbox.wav)
• The caller speaks to record the message and presses # to end recording. The VMS plays back the recorded
message and prompts: To re-record the message press 1, to confirm press 2." (RecAgain.wav)
• If Message Verification flag is disabled, the VMS does not playback the recorded message and prompts:
"Your message has been recorded" after caller press # to end message recording.
Extension Users
• When you call the VMS by dialing 3931,
• The VMS plays the Welcome Message: "Welcome! Please dial the extension number Or to dial by name
press 7. To leave message, press 6. To go to operator, press 9. For more options, press 0. To disconnect,
press #."
• Dial the desired extension number VMS prompts: "Record your message after the beep and press #
(hash/pound) to end. (Recmsg.wav)
• If no mailbox is assigned to the dialed extension number, the VMS prompts: "Mailbox not assigned. To
disconnect, press 1. To go to Home Position, Press 0." (NoMailbox.wav)
• Record your message and press # to end recording. The VMS plays back the recorded message and
prompts: To re-record the message press 1, to confirm press 2." (RecAgain.wav)
• If Message Verification is disabled, the VMS does not playback the recorded message and prompts:
"Your message has been recorded", “press # (hash/pound) to end’.
• If the extension you called is busy, you may leave a message for another extension:
It is mandatory for the caller/extension user to terminate the recording by dialing # (or the digit configured).
If recording of the message is terminated simply by going on-hook, the VMS will not terminate the recording
and the call will be disconnected only after time-out, i.e. the Maximum Message Length configured for the
extension.
What’s this?
When extension users access their mailbox, the VMS offers them the When callers/extension user leave messages
in the mailbox of extension users, n they are inaccessible or the user has forwarded his calls to the mailbox. User
should access their mailboxes to listen to the messages.
Once the message is heard by the mailbox owner, VMS treats it as an old message and places it in the old
message list. VMS also gives a facility to save the heard message as a new one and every-time the mailbox is
accessed the VMS plays such 'save-as-new' message as a new message.
How to use
• Call the VMS by dialing 3931,
• VMS responds with: "You have <n> new messages" followed by the prompt: "Enter your mailbox
password". (Enterpwd.wav)
• Enter your mailbox password VMS prompts: "You have n new/no new messages."
• VMS prompts: "To listen to new messages press '1', to listen to old messages press '2', to send a message
press '3', to change your mailbox settings press '4', to go to home position press '0'." (Mmmm.wav)
What’s this?
The VMS allows extension users to change the settings of the following facilities of their mailbox:
• Record the Extension Name for their mailbox.
• Redirect Messages from their mailbox.
• Delete all old messages
How to use
• Call the VMS by dialing 3931,
• VMS responds with: "You have <n> new messages" followed by the prompt: "Enter your mailbox
password". (Enterpwd.wav)
• VMS prompts: "To listen to new messages press '1', to listen old message press '2', to send a message
press '3', to change your mailbox settings press '4', to go to home position press '0'." (Mmmm.wav)
• The VMS responds with: “For Mailbox Name, Press '1', For message redirection, Press '2', To delete all
old messages of your mailbox, press '3', to go to previous menu press '0'. (Chgmbset.wav)
• Dial 1 VMS prompts: "To record Name, press '1'. To play Name, press '2'. To go to Previous menu,
press '0'." (MailboxName.wav).
• Dial 1 to record name for mailbox VMS prompts: "Record your name after the beep and press # (hash/
pound) to end." (Recname.wav)
Names for extensions can also be recorded from the System Administrator mode. See “Recording
Extension Names”.
• Dial 3 VMS prompts: “You are about to delete all old messages of your mailbox. To proceed press 1, to
cancel press 2.” (DelAllCnf.wav)
• Dial 1 to delete all old messages in your mailbox VMS responds with: “Your old messages have been
deleted”. (DelAllDone.wav)
What’s this?
The Voice Mail System (VMS) sets Message Wait on the extension, whenever a new message arrives in its
personal Mailbox of the extension. The VMS indicates the new message to the extension as per the Type of
Message Wait Notification set for the extension. This may be in the form of a Stuttered Dial Tone, a Voice
Message, Ring or LED Lamp. See, “Message Wait set by the Voice Mail System” under “Message Wait” to know
more.
The Message Notification feature of the VMS is an extension of the Message Wait feature. When Message Wait
Notification for an extension is set as Ring, the VMS makes a Message Notification call to the extension.
How it works
• Extension A has Message Wait Notification type set as Ring.
• Whenever there is a new message in A’s mailbox, the system will play the Message Wait Ring (Short,
Fast) on extension A. See “Distinctive Rings”.
• Extension A will ring for the duration of the Message Wait Ring Timer (configurable; default: 30 seconds).
• When Extension A answers the call within this timer, Extension A gets connected to the VMS.
• The VMS answers the call and allows access to Extension A’s mailbox.
• If Extension A does not answer the Message Notification call within the Message Wait Timer, the system
will ring on the extension again for as many times as the Message Wait Ring Count (configurable; default:
10 times), and at the interval set as the Message Wait Ring Timer Interval (configurable; default: 30
minutes).
How to configure
To provide Message Notification Call to extensions, you must configure Ring as Voicemail/Message Wait
Notification Type for the extension. See “Basic Settings” for instructions on configuring the different Extension
types: SLT, DKP, SIP.
You may also configure the related Message Wait Timer, the Message Wait Ring Count and the Message Wait
Ring Interval, if required. See “System Timers and Counts” for instructions.
How to use
• Go Off-hook, when your extension rings to indicate Message Wait (short, fast ring),
• The VMS greets with the message:“This is your message notification call”.
• The VMS plays the message: "You have <n> new messages" followed by the prompt: "Enter your mailbox
password". (Enterpwd.wav)
What’s this?
Message Verification enables extension users to check the message they have recorded before sending it to
someone.
Message Verification allows callers to check the message they have recorded in the mailbox of an extension user.
Thus Message Verification is used in the VMS features “Leaving a Message”, “Sending Messages”, and “Broadcast
Message”.
How it works
• For Message Verification to work, it must be enabled in the VMS.
• With Message Verification enabled, each time a caller or an extension user records a message, the VMS
offers to the caller/extension user the option verify the recorded message and re-record the message, if
they want.
• When the caller/extension user uses the option to verify and re-record the message, the VMS sends the
message to the mailbox of the receiver.
How to configure
By default, Message Verification is enabled in the VMS. So, callers and all extension users with voice mail facility
can verify the message they have recorded. If you want to disable this feature,
• Open Jeeves.
• Scroll to and click the VMS Configuration link under Advanced Settings.
• By default, Message Verification is enabled. To disable Message Verification, clear the check box.
What’s this?
The VMS enables extension users to send messages to other extensions that have a mailbox. An extension user
can send a message to as many as 10 destinations at a time. The extension user can send the message either to a
specific mailbox or to a Distribution List.
VMS also gives facility to the Sender of the message to request a read receipt of the message sent. When the
Recipient has read the message, the VMS generates a file containing the first 5 seconds of the message that was
sent and delivers it to the Sender's mailbox in the form of a new message with the Date and Time stamp (if enabled
in Play Message Details) and the prompt: "This message was read by <Extension Name> <5 seconds of message
sent>". If the Sender does not request ‘read receipt’, no such message is delivered to the Sender.
If the sent message is not delivered to the Recipient, the VMS generates a file containing the first 5 seconds of the
message that was sent by the Sender and delivers it to the Sender's mailbox in the form of a new message with the
prompt: "This message was not delivered to <Extension Name><5 seconds of the message>".
How to use
• Call the VMS by dialing 3931,
• VMS responds with: "You have <n> new messages" followed by the prompt: "Enter your mailbox
password". (Enterpwd.wav)
• Enter your mailbox password VMS prompts: "You have n new/no new messages."
• VMS prompts: "To listen to new messages press '1', to listen to old messages press '2', to send a message
press '3', to change your mailbox settings press '4', to go to home position press '0'." (Mmmm.wav)
• Dial valid extension numbers/distribution list number VMS prompts: "Record your message after the
beep and press # (hash/pound) to end".
• Speak to record the message and press # (hash/pound) to end. VMS plays back the recorded message
and prompts: "To request read receipt press '1', to ignore read receipt press '2'."
• If Message Verification flag is disabled then VMS will not playback the recorded message.
• If the message could not be delivered to the destination you dialed, the VMS responds: "Sorry the
message cannot be sent." (Pending.wav)
Once a valid destination number is entered and no more extensions are selected, the VMS
understands it to be the end of list and sends the message.
What’s this?
The VMS offers extension users to re-direct the messages in their mailbox to another mailbox. The feature can be
used by employees who are out of office or unable to access their mailbox. Using Redirect Messages, they can
ensure that important messages are attended to by their colleagues in their absence.
How to use
• Call the VMS by dialing 3931,
• VMS responds with: "You have <n> new messages" followed by the prompt: "Enter your mailbox
password". (Enterpwd.wav)
• Enter your mailbox password VMS prompts: "You have n new/no new messages."
• VMS prompts: "To listen to new messages press '1', to listen to old messages press '2', to send a message
press '3', to change your mailbox settings press '4', to go to home position press '0'." (Mmmm.wav)
• Dial 4 VMS prompts: “For Mailbox Name, Press '1', For message redirection, Press '2', To delete all old
messages of your mailbox, press '3', to go to previous menu press '0'. (Chgmbset.wav)
• Dial 2 VMS prompts: "To set message redirection press '1', to cancel message redirection press '2', to
go to previous menu press '0'." (MbMsgRd.wav)
• Dial 1 to set message redirection VMS prompts: "Enter the destination extension." (MsgRdHdl.wav)
• Dial valid destination extension VMS responds: "Command has been executed." (Okcmd.wav)
• Dial 2 to cancel message redirection VMS responds: "Command has been executed." (Okcmd.wav)
Configuration Upload
ETERNITY NE provides you the facility to upgrade the system software at the click of a button. ETERNITY NE
supports an embedded FTP71 server which can be used for Uploading and Downloading Configuration Files.
• Type the current IP Address of the Ethernet Port of ETERNITY NE in the Address bar as ftp://
192.168.1.101
71. FTP or File Transfer Protocol is a standard Internet Protocol that is used to exchange files between computers on the IP network.
• In the Password field, enter the SE password (default 1234) and click the Log On button.
• You may either delete the existing files or copy these files to another location as backup.
• Restart the system after uploading the files. The new configuration will be applicable only after the system
restarts.
• Open Jeeves on the browser, by typing the current IP Address of the Ethernet Port of ETERNITY NE.
• Right click the first file, select the option Open link in FireFTP
• The folders and files on the computer appear on the left. The FireFTP pane on the right displays the
existing file(s) in the config folder. Delete the existing file in the config folder.
• Select the config files from the computer pane. Click the upload arrow to copy the file in config folder in the
FireFTP pane.
Before you upgrade the system firmware, you can save the existing application/configuration/driver/ web
pages folder and files as backup for retrieval, later. When required, you can copy and paste the same
folder back on the FTP server of ETERNITY NE.
ETERNITY NE provides you the facility to upgrade the system software at the click of a button. ETERNITY NE
supports an embedded FTP72 server which can be used for Uploading and Downloading System files.
• Type the current IP Address of the Ethernet Port of ETERNITY NE in the Address bar as ftp://
192.168.1.101
72. FTP or File Transfer Protocol is a standard Internet Protocol that is used to exchange files between computers on the IP network.
• Open the system folder. All the system files and web folder will be displayed.
• You may either delete the existing files/folder or copy these files/folder to another location (as backup).
• Restart the system after uploading the files.The new firmware will be applicable only after the system
restarts.
• Open Jeeves on the browser, by typing the current IP Address of the Ethernet Port of ETERNITY NE.
• Select the system files from the computer pane. Click the upload arrow to copy the file in system folder in
the FireFTP pane.
• Restart the system after uploading the files. The new firmware will be applicable only after the system
restarts.
Before you upgrade the system firmware, you can save the existing application/configuration/driver/ web
pages folder and files as backup for retrieval, later. When required, you can copy and paste the same
folder back on the FTP server of ETERNITY NE.
You can monitor the state of ports and IO operations, the Voice Mail System and the VoIP Ethernet port using
System Debug.
Matrix ETERNITY NE supports Syslog73 Client for debugging. The Syslog Client enables the system to send debug
messages in syslog format to the remote ‘Syslog Server’ on the IP network. You can view the system debug
messages on the remote server.
Each debug message includes the MAC Address of Ethernet port of ETERNITY NE which sends the debug
messages to the Syslog server.
• Open Jeeves.
• Click the System Debug sub-link. The System Debug page opens.
73. Syslog is one of the protocols used extensively for sending debug messages, and is defined in RFC 3164.
• In the Syslog Server IP Address: Port field, enter the server address and the port number. Default: port
number 514.
VMS Debug
• In the Syslog Server IP Address: Port field, enter the server address and server port number where the
debug log is to be sent. Default: port number is 514.
• Select the application/processes you want to debug: VMS Application, SMTP, Configuration Transfer
• In the Syslog Server Address field, enter the server address and server port number where the debug log
is to be sent. Default: port number is 514.
• Select the desired Debug Level from the following supported by ETERNITY NE:
• Serial
• SIP
• CALL
• Registered User
• Registered Trunk
• BLF/MWI
• Media
• VoPP
• Call Advance
• Registered User Advance
• Registered Trunk Advance
• BLF/MWI Advance
• Media Advance
• Presence
• IM
The debug log will be generated for the options you selected. For example: if debug log of Call is required,
enable 'CALL' level and disable all other debug levels.
• In the Start Port field, enter the number of the selected Port Type from which the system should start
debugging.
• In the End Port field, enter the number of the selected Port Type after which the system should end
debugging.
If you want to debug a range of ports, configure the starting and ending port numbers in the Start Port and
End Port fields.
I/O Debug
• Select the Port Type for which you want to debug I/O communication:
• DKP
• CO-SLT
• VoIP
• Mobile
• Door Phone
• VMS
• In the Start Port field, enter the number of the selected Port Type from which the system should start
debugging.
• In the End Port field, enter the number of the selected Port Type after which the system should end
debugging.
If you want to debug a range of ports, configure the starting and ending port numbers in the Start Port and
End Port fields.
To debug a single port, define the Start and End Port numbers as the same.
ETERNITY NE supports PCAP Trace, which you can use to detect and diagnose network related problems for the
Ethernet Port of ETERNITY.
Packets traveling over a network are captured and saved in the system. You can save these trace files (packets
captured by the system) on a computer and open these trace files using a graphical packet capture and protocol
analysis tool such as Wireshark or Ethereal.
PCAP Trace for the Ethernet Port allows a maximum of 1 MB of packets can be captured and stored in the
ETERNITY.
ETERNITY also supports Filters and 'Promiscuous' mode for capturing packets, which you can use to specify the
types of data packets to be captured.
How to use
To use PCAP Trace for the Ethernet Port of ETERNITY,
• In the Filter Settings, decide the type of packets to be captured and set the Filter accordingly. The Filter
should be maximum 60 characters in length; all ASCII characters are allowed. By default, this field is
blank. So all packets will be captured.
• To capture packets which are transmitted from the system i.e. from IP address 192.168.1.191:
• Filter Settings = src 192.168.1.191
• To capture packets which are received for the system i.e. to IP address 92.168.1.191:
• Filter Settings = dst 192.168.1.191
• To capture only packets which are received by the system having IP address 192.168.1.191:
• Filter Settings = src 192.168.1.191 or dst 192.168.1.191
• To capture packets which are transmitted from the system for particular port number only i.e. from IP
address 192.168.1.191 and port number 161:
• Filter Settings = src 192.168.1.191 and port 161
src port number src port 5060 Capture packets if the packet has
a source port value of 5060.
dst port number dst port 80 Capture packets if the packet has
a destination port value of 80.
port number port 5060 Capture packets if the packet has either source
or destination port value of 5060.
src host ip address src host 192.168.1.176 Capture packets if the source IP address
is 192.168.1.176
dst host ip address dst host 192.168.1.176 Capture packets if the destination IP address
is 192.168.1.176.
host ip address host 192.168.1.176 Capture packets if either source or destination IP address
is 192.168.1.176
Jeeves will throw up an alert message 'Invalid filter! Please enter valid filter' if you do not enter a valid filter.
It is not mandatory to set Filters. When the Filter Settings field is left blank, the system will capture all
packets.
• Select the Enable Promiscuous Mode check box, if you want specific data packets to be captured.
When you enable Promiscuous mode, the ETERNITY will capture all network traffic. However, this will
work only in a non-switched environment.
When Promiscuous Mode flag is disabled, the system will capture only traffic that is directly related to it.
Only traffic to, from or routed through the ETERNITY will be picked up by the PCAP Trace.
'Filter Settings' and 'Promiscuous Mode' (enabled) will not be cleared during power down.
OR
• Wait for the system to stop packet capturing. The system stops packet capturing once the maximum
allotted memory of 1 MB (RAM) is utilized.
• Under Status, the Number of Packets and bytes captured as per the filter setting will be displayed as
Packets Captured and Total Bytes respectively.
Capturing of packets will not stop if you open any other page of Jeeves. So, you may continue using
Jeeves for any other purpose while PCAP Trace is being used.
• When the packet capturing is stopped (by you or the system), click the Save Trace File button to
download the captured files from the system’s FTP server.
The current packets captured will not be deleted after you have saved the trace file. The current packets
will be deleted when you start the PCAP capture again.
• Now, you can open the trace files using Wireshark or Ethereal or any other similar software which supports
opening of trace files.
System Details
• Open Jeeves.
You can view the memory status of the VMS as well as individual mailboxes. Checking the memory status of the
voice mail and individual mailboxes helps you to monitor usage of VMS and make optimal allocation of the VMS
resources.
To view Voice Mail Memory Status,
• Open Jeeves.
Mailbox Status
You can also view the status of individual mailboxes on this page under Mailbox Status.
2. Click OK
• Extension Number: This is the extension number you selected for viewing mailbox status.
• Allotted Voice Mail?: If mailbox is assigned to the extension, it displays Yes. If no mailbox is assigned
No will appear.
• Allotted Voice Mail Box Size (Minutes): The size of the mailbox in minutes assigned to the extension.
• Memory Consumed by Voice Mail User : The current status of the mailbox memory of the extension.
• New Voice Mail/Total Voice Mail: The number of new (unread) messages out of the total number of
messages in the mailbox.
• Open Jeeves.
• Port Status: This is the status of the connection - showing Initialization with the Network, Registering
with the Network, Idle or Busy state of the network. It also shows errors and alerts when SIM is absent,
the wrong SIM PIN has been entered, SIM PUK is required.
• IMEI: This is the unique identification number of (the GSM engine) each Mobile port.
• SIM ID: This is the Integrated Circuit Card ID (ICC-ID) of the SIM Card inserted in the Mobile port. Each
SIM is internationally identified by its ICC-ID. ICC-IDs are stored in the SIM Card and are also printed
on the SIM card body.
• IMSI: International Mobile Subscriber Identity (IMSI) is a unique number stored in the SIM card.
• Registered with Network: This shows the type of network with which the Mobile port is registered,
whether GSM, GSM Compact, 3G or UMTS.
• Network Operator Code: This is the MCC-MNC code of the network with which the mobile port is
registered.
• Network Operator Name: This is the name of the service provider/network operator with which the
Mobile Port is registered.
• Signal Strength (dBm): This is the signal strength in '-dBm' as received from the network with which
the Mobile port is registered.
• Bit Error Rate (BER): BER is Bit Error Rate which defines the quality of the channel.
• Cell ID: This is the 16-bit identifier that identifies the cell. The cell is the radio coverage area given by
one BTS (Base Transceiver Station).
• Location Area Code (LAC): The LA (Location Area) is a group of cells defined by the Operator. The
LAC (Location Area Code) uniquely identifies a LA within a PLMN (Public Land Mobile Network).
• Call Duration: This is the total call duration of matured outgoing calls74 on the Mobile port. This data is
used for calculating Answer Seizure Ratio (ASR) for the port. It is displayed in MMMMMM:SS format.
• Call Budget Type: This shows the Call Budget Type, i.e. whether Amount, Minutes or Number of
Calls, set on the Mobile port.
• Allotted Amount (Rs.) / Minutes/Calls: This shows the sum/number of minutes/number of calls
allotted as Call Budget on the Mobile port.
• Consumed Amount (Rs.) / Minutes/Calls: This shows the sum/number of minutes/number of calls of
the allotted Call Budget that has been used up on the Mobile port.
• Call Budget Reset Mode: This shows the whether manual or scheduled reset of the consumed call
budget is set on the Mobile port.
• Call Budget Reset Schedule (Date): This shows whether the consumed Call Budget on the Mobile
port is to be reset Daily or on a particular date of a month.
• Reset Consumed Amount/Minutes/Calls: This editable field allows the System Engineer to reset the
consumed Call Budget Amount/Minutes/Calls at any time, manually.
74. Matured calls are outgoing calls for which 'CONNECT' message was received from the network.
• Successful Calls: This is the total number of matured outgoing calls made from the Mobile port. This
data is used for calculating Answer Seizure Ratio (ASR) and Average Call Duration for the port.
• ASR: This is the Answer Seizure Ratio (ASR) calculated by the system for the Mobile port, in terms of
percentage. ASR is the sum of all outgoing matured calls from the Mobile port, divided by the total
number of outgoing calls made from the Mobile port, multiplied by 100. The system calculates ASR
after the completion of the outgoing call.
• ACD: This is the Average Call Duration (ACD) of outgoing calls made from the Mobile port. It is an
indicator for monitoring the network condition. Decreasing ACD is indicative of trouble in the network
condition.
The system calculates ACD after the completion of the outgoing calls, by dividing the total call duration
by the number of outgoing matured calls.
• Reset ASR and ACD: This field allows the System Engineer to reset manually the ASR and the ACD
of the Mobile port.
The parameters Total Call duration, Number of matured calls, Total Number of OG Calls, ASR and
ACD are saved in the configuration, and are not reset on Power OFF condition. The system maintains
the statistics for the last 999 calls. When the total number of outgoing calls exceeds 999, the system
will stop calculating ACD and ASR and will display ASR and ACD calculated on the basis of the last
999 calls only.
Therefore, the System Engineer must manually reset ASR and ACD when the total number of calls
reaches 999. When you reset ASR and ACD the number of call matured and the number of calls dialled
is reset to 0.
ASR and ACD can be reset anytime, even when the total number of calls is less than 999.
When ACD is reset, only the 'Total Call Duration' maintained for the ACD calculation will be reset. The
'Total Call Duration' of the Call Budget, i.e. the consumed minutes maintained for the Call Budget on the
mobile port will remain unaffected.
75. The total number of outgoing calls made includes the number of times the ATD has been sent from the Mobile port to the network.
• Open Jeeves.
• Open Jeeves.
• To view the status of SIP trunks 2, 3 and 4 click the tab SIP-2, SIP-3, SIP-4.
• Open Jeeves.
• Open Jeeves.
Technical Specifications
TECHNOLOGY
PCM/TDM Digital Switching (100% Non-
Type of Switching
blocking)
Processor 32-bit RISC
CO (2-WIRE TRUNK)
Signaling Loop Start
Loop Limit 1200?
Off Hook AC Impedance 600/900/Complex
Pulse Dialing 10/20PPS
DTMF Dialing and Reception ITU-T Q.23 & Q.24
Return Loss >18dB
Longitudinal Balance >50dB
GSM
Quad-Band: GSM850,EGSM900, DCS1800,
Frequency Band (MHz)
PCS1900
Compliant ETSI GSM Phase 2/2+
SIM Card One SIM Per GSM Port
SIM Interface 1.8V, 3V
Class 4 (2W) at GSM850 MHz and EGSM900
Transmission Power MHz Band
Class 1 (1W) at DCS 1800 MHz and PCS1900
MHZ Band
RF Sensitivity Better than -106dBm
Protocol AT Command Interface
One Antenna per GSM Port, 1.8/3.0*dBi, 50?
External Antenna SMA (Male) Connector, Omni Directional with
Cable of 3 Meters Length
UMTS (3G)
Quad-Band: GSM850,EGSM900, DCS1800,
PCS1900
UMTS A Module: Tri-Band: WCDMA 850/1900/
Frequency Band (MHz)
2100
UMTS E Module: Tri-Band: WCDMA 900/1900/
2100
Compliant ETSI GSM Phase 2/2+
SIM Card One SIM Per UMTS (3G) Port
SIM Interface 1.8V, 3V
Class 4 (2W) at GSM850 MHz and EGSM900
MHz Band,
Class 1 (1W) at DCS 1800 MHZ and PCS1900
Transmission Power
MHz Band,
Class 3 (0.25W) at WCDMA 850/1900/2100 MHz
Band
Better than -106dBm at GSM850, EGSM900,
DCS1800 and PCS1900,
RF Sensitivity
Better than -108 dBm at WCDMA 850,
Better than -108 dBm at WCDMA 1900/2100
Protocol AT Command Interface
VoIP
SIP v2, SIP over TCP, Symmetric RTP, RTCP,
VoIP Protocols
100rel/PRACK
Network Protocol IPv4,TCP,UDP, STUN,ARP,ICMP,PPP,DNS
Maximum 4 SIP Accounts Per System, Out
Bound Proxy Support, Display Name, User
SIP
Name, Password, URL, Proxy URL, Register
Interval
Line Echo Cancellation G.168 With 32/64/128ms Tail Length
Dynamic Jitter Buffer (Adaptive), Comfort Noise
Voice
Generation and Voice Activity Detection
NAT STUN and NAT Keep Alive
G.711 (A-law, µ-Law), G.723, G.729AB, GSM-FR
Voice CODECs
and iLBC
Fax T.38 Relay and Pass Through
Quality of Service SIP QoS and RTP QoS
MD5 Authentication for SIP, Password Protected
Security
Configuration by Admin and User
Ethernet (RJ45) Port, Auto MDIX (10/100 base
Data Network
T)
Auxiliary Ports
Power Relay, Resistive Load - 1.0A 24VDC,
Digital Output Port
Operation Time - 8ms (Max.), RJ45
Speaker Output - 1.41Vrms (Max.), Microphone
Door Phone Port
Input - 1.34Vrms (Max.), RJ45
LED Indications
Power 1 LED Single Color (GREEN)
System Status 1 LED Dual Color (RED/GREEN)
Power Supply
Input External Adaptor - 24VDC, 2A
Power Consumption (Typical) 15W
Environment
Operating Temperature -10°C to +50°C (14°F to 122°F)
Operating Humidity 5-95% RH, Non-Condensing
Storage Temperature -40°C to +85°C (-40°F to +185°F)
Mechanical
Dimension (WxHxD)
Table Top, Wall Mount
19" Rack Mount
30x4.25x20cm (11.81"x1.67"x7.87")
48.3x4.25x20.2cm (19"x1.67x7.95")
Installation Wall Mount, Table-Top, 19" Rack Mount
* Depends on GSM/UMTS(3G) Frequency Band
System Resources
ETERNITY Paltforms
ETERNITY NE GSM Module to insert 1 SIM card for Voice Telephony over GSM Networks
ETERNITY NE UMTS A Module to insert 1 SIM card for Voice Telephony over UMTS (3G) Networks
ETERNITY NE UMTS E Module to insert 1 SIM card for Voice Telephony over UMTS (3G) Networks
ETERNITY NE IP
Module to provide VoIP telephony with 4 SIP Trunk Connectivity
SERVER
Module to connect 4-Wire Door phone and Relay activated device (i.e. Door
ETERNITY NE 4WDP
Opener)
4 Channel Voice Mail to attend 4 Simultaneous Calls and Voice Mailboxes for
ETERNITY NE VMS
Individual Extensions
056 PreName.wav More than one match found. Matching names will be played one by one.
Press '1' after the name you wish to select., To skip to the next name press 2, to
go to home position press 0
061 Recmsg.wav Record your Message after the beep and press any digit to end
066 Chkinwel.wav It is our pleasure receiving you. We will do our best to make your stay comfortable
092 PreLeave.wav More than one match found. Matching names will be played one by one.
Press '1' after the name to leave a message. Press 2 to skip to next station name.
Press 0 to go to home position
099 Noreply.wav The person you are trying to reach is not available
101 Recname.wav Record your name after the beep and press any digit to end
103 Takecall.wav Dial '1' to take the call, '2' and disconnect if you do not want to take the call
105 LeavemsgG.wav Dial '1' to leave a message, '2' to disconnect, '0' to go to Home Position
109 LeavemsgE.wav Enter the extension number for which you wish to leave message.
110 LeavemsgH.wav Enter the Room number or the Extension number for which you wish to leave
message
111 NoMailbox.wav Mailbox not assigned. To Disconnect, press '1'. To go to Home Position, Press '0'
125 Mmmm.wav To listen to new message, press '1', to listen to old message press '2', to send a
message press '3', to change your mailbox settings press '4', to go to home
position press '0'
127 Mmsmo.wav To replay the message press '1', for date and time stamp press '2', to delete the
message press '3', to play the next message press '4', to forward the message
press '5', to save the message as new press '6', to go to previous menu press '0'
128 Fwdhow.wav To forward the message without comment press '1', to forward the message with
comment press '2', to go to previous menu press '0'
130 Askcfrm.wav To request read receipt press '1', to ignore read receipt press '2'.
134 PreEntry.wav More than one match found for the extension. Names will be played one by one.
Press '1' after the name to make selection, to skip to next name press '2', to go to
home position press '0'
135 Chgmbset.wav To change password, press 1. To change to auto mode, press '2'. To change to
manual mode, press '3'. For message notification, press '4'. For date-time stamp,
press '5'. For announce name, press '6'. For message redirection, press '7'. For
delivery option in mailbox full condition, press '8'. For Mailbox Name, press '9'. To
go to home position press '0'
137 Cam.wav For morning zone settings press '1', for afternoon zone settings press '2', for
evening zone settings press '3', for non-working zone settings press '4', to go to
previous menu press '0'
138 CmbTt.wav For start time press '1', to change transfer type press '2', to record greeting press
'3', to play greeting press '4', to go to previous menu press '0'
139 Ttopt.wav To change your transfer type for none press '1', for blind press '2', for wait for ring
press '3', for wait for answer press '4', for screen press '5', to go to previous menu
press '0'
140 Cmm.wav To change transfer type press '1', to record greeting press '2', to play greeting
press '3', to go to previous menu press '0'
142 MbNotTyp.wav To cancel message notification press '1', to set immediate message notification
press '2', to set scheduled message notification press '3', to go to previous menu
press '0'
143 MbMsgNot.wav For message notification in morning zone press '1', for message notification in
afternoon zone press '2', for message notification in evening zone press '3', for
message notification in non-working zone press '4', to go to previous menu press
'0'
144 PrgPhone.wav Enter the phone number to which you wish to send message notification
145 SchMsgNot.wav For scheduled message notification 1 press '1', for scheduled message
notification 2 press '2', for scheduled message notification 3 press '3', to go to
previous menu press '0'
146 EntrTmPh.wav To enter time press '1', to enter phone number press '2', to go to previous menu
press '0'
148 MbDtTmCh.wav To enable date-time stamp press '1', to disable date-time stamp press '2',to play
date-time stamp on-demand press '3', to go to previous menu press '0'
149 AnnName.wav To enable announce name press '1', to disable announce name press '2', to go to
previous menu press '0'
150 MbMsgRd.wav To set message redirection press '1', to cancel message redirection press '2', to
go to previous menu press '0'
152 Mbfulstg.wav To refuse new messages press '1', to deliver new message to general mailbox
press '2', to overwrite old messages press '3', to go to previous menu press '0'
153 At.wav At
158 MailboxName.w To record Name, press '1'. To play Name, press '2'. To go to previous menu, press
av '0'
163 RecMsgStopCo Record your message after the beep and press
de.wav
166 RecNameStopC Record your name after the beep and press
ode.wav
171 EntertimeI.wav Enter the time, HH:MM in Twenty four hour format. To cancel all alarms, press £.
172 EntertimeU.wav Enter the time, HH:MM in Twelve hour format. To cancel all Alarm, press £. For
am, press 1. For pm, press 2.
174 SetOnceDaily.w To set Once, Press '1'. To set Daily, Press '2'.
av
185 AlarmDateI.wav Enter the Date in DD MM YYYY format. To Cancel all reminders, press £. For
example, to enter date 17th March 2008, Dial One Seven Zero Three Two Zero
Zero Eight
186 AlarmDateU.wa Enter the Date in DD MM YYYY format. To Cancel all reminders, press £. For
v example, to enter date March 17th 2008, Dial Zero Three One Seven Two Zero
Zero Eight
190 Alarmnoset.wav Sorry! Your Wake Up Alarm cannot be set. Please call Operator for further
assistance
191 Remindernoset. Sorry! Your Reminder cannot be set. Please call Operator for further assistance
wav
192 RemoteExt.wav Enter the Extension number for which you have to set or cancel Wake Up Alarm
193 RemoteExtH.wa Enter the Room number for which you have to set or cancel Wake Up Alarm
v
194 PreEntry2.wav More than one match found. Names will be played one by one. Press '1' after the
name you want to select. Press '2' to Skip to next name
200 DailyPerWakeup You have set Daily Personal Wake up Alarm at…
Set.wav
201 DailyAutoWakeu You have set Daily Automated Wake up Alarm at…
pVeri.wav
214 SWakeUpgreeti This is your Wake up Call. For Acknowledge, Please Press '0'. Music of 5 seconds
ng.wav
215 SDailyWakeUpg This is your Daily Wake up Call. For Acknowledge, Please Press '0'. Music of 5
reeting.wav seconds.
216 SRemindergreet This is your Reminder call. For Acknowledge, Please Press '0'. Music of 5
ing.wav seconds
219 Entertime2U.wa Enter the Time, HH MM in Twelve hour format. For am, Press 1. For pm, Press 2.
v
220 RemoteExtRem. Enter the Extension number for which you have to set or cancel Reminder.
wav
221 RemoteExtHRe Enter the Room number for which you have to set or cancel Reminder
m.wav
Index Area Code Area Name Pulse Rate Type Ignore Digit Count
1 1201 NJ 2 0
2 1202 DC 2 0
3 1203 CT 2 0
4 1204 Manitoba 2 0
5 1205 AL 2 0
6 1206 WA 2 0
7 1207 NE 2 0
8 1208 ID 2 0
9 1209 CA 2 0
10 1210 TX 2 0
11 1212 NY 2 0
12 1213 CA 2 0
13 1214 TX 2 0
14 1215 PA 2 0
15 1216 OH 2 0
16 1217 IL 2 0
17 1218 MN 2 0
18 1219 IN 2 0
19 1224 IL 2 0
20 1225 LA 2 0
21 1226 Ontario 2 0
22 1228 MS 2 0
23 1229 GA 2 0
24 1231 MI 2 0
25 1234 OH 2 0
26 1239 FL 2 0
27 1240 MD 2 0
28 1242 Bahamas 2 0
29 1246 Barbados 2 0
30 1248 MI 2 0
31 1250 BC 2 0
32 1251 AL 2 0
33 1252 NC 2 0
34 1253 WA 2 0
35 1254 TX 2 0
36 1256 AL 2 0
37 1260 IN 2 0
38 1262 WI 2 0
39 1264 Anguilla 2 0
40 1267 PA 2 0
41 1268 Antigua 2 0
42 1269 MI 2 0
43 1270 KY 2 0
44 1276 VA 2 0
45 1281 TX 2 0
46 1284 BVI 2 0
47 1289 Ontario 2 0
48 1301 MD 2 0
49 1302 DE 2 0
50 1303 CO 2 0
51 1304 WV 2 0
52 1305 FL 2 0
53 1306 Saskatchewan 2 0
54 1307 WY 2 0
55 1308 NE 2 0
56 1309 IL 2 0
57 1310 CA 2 0
58 1312 IL 2 0
59 1313 MI 2 0
60 1314 MO 2 0
61 1315 NY 2 0
62 1316 KS 2 0
63 1317 IN 2 0
64 1318 LA 2 0
65 1319 IA 2 0
66 1320 MN 2 0
67 1321 FL 2 0
68 1323 CA 2 0
69 1325 TX 2 0
70 1330 OH 2 0
71 1331 IL 2 0
72 1334 AL 2 0
73 1336 NC 2 0
74 1337 LA 2 0
75 1339 MA 2 0
76 1340 USVI 2 0
77 1345 Cayman 2 0
78 1347 NY 2 0
79 1351 MA 2 0
80 1352 FL 2 0
81 1360 WA 2 0
82 1361 TX 2 0
83 1386 FL 2 0
84 1401 RI 2 0
85 1402 NE 2 0
86 1403 Alberta 2 0
87 1404 GA 2 0
88 1405 OK 2 0
89 1406 MT 2 0
90 1407 FL 2 0
91 1408 CA 2 0
92 1409 TX 2 0
93 1410 MD 2 0
94 1412 PA 2 0
95 1413 MA 2 0
96 1414 WI 2 0
97 1415 CA 2 0
98 1416 Ontario 2 0
99 1417 MO 2 0
101 1419 OH 2 0
102 1423 TN 2 0
103 1424 CA 2 0
104 1425 WA 2 0
105 1430 TX 2 0
106 1432 TX 2 0
107 1434 VA 2 0
108 1435 UT 2 0
110 1440 OH 2 0
112 1443 MD 2 0
115 1469 TX 2 0
117 1478 GA 2 0
118 1479 AR 2 0
119 1480 AZ 2 0
120 1484 PA 2 0
122 1501 AR 2 0
123 1502 KY 2 0
124 1503 OR 2 0
125 1504 LA 2 0
126 1505 NM 2 0
128 1507 MN 2 0
129 1508 MA 2 0
130 1509 WA 2 0
131 1510 CA 2 0
132 1512 TX 2 0
133 1513 OH 2 0
135 1515 IA 2 0
136 1516 NY 2 0
137 1517 MI 2 0
138 1518 NY 2 0
140 1520 AZ 2 0
141 1530 CA 2 0
142 1540 VA 2 0
143 1541 OR 2 0
144 1551 NJ 2 0
145 1559 CA 2 0
146 1561 FL 2 0
147 1562 CA 2 0
148 1563 IA 2 0
149 1567 OH 2 0
150 1570 PA 2 0
151 1571 VA 2 0
152 1573 MO 2 0
153 1574 IN 2 0
154 1575 NM 2 0
155 1580 OK 2 0
156 1585 NY 2 0
157 1586 MI 2 0
159 1601 MS 2 0
160 1602 AZ 2 0
161 1603 NH 2 0
162 1604 BC 2 0
163 1605 SD 2 0
164 1606 KY 2 0
165 1607 NY 2 0
166 1608 WI 2 0
167 1609 NJ 2 0
168 1610 PA 2 0
169 1612 MN 2 0
171 1614 OH 2 0
172 1615 TN 2 0
173 1616 MI 2 0
174 1617 MA 2 0
175 1618 IL 2 0
176 1619 CA 2 0
177 1620 KS 2 0
178 1623 AZ 2 0
179 1626 CA 2 0
180 1630 IL 2 0
181 1631 NY 2 0
182 1636 MO 2 0
183 1641 IA 2 0
184 1646 NY 2 0
187 1650 CA 2 0
188 1651 MN 2 0
189 1660 MO 2 0
190 1661 CA 2 0
191 1662 MS 2 0
194 1671 GU 2 0
195 1678 GA 2 0
196 1682 TX 2 0
197 1684 AS 2 0
199 1701 ND 2 0
200 1702 NV 2 0
201 1703 VA 2 0
202 1704 NC 2 0
204 1706 GA 2 0
205 1707 CA 2 0
206 1708 IL 2 0
208 1710 US 2 0
209 1712 IA 2 0
210 1713 TX 2 0
211 1714 CA 2 0
212 1715 WI 2 0
213 1716 NY 2 0
214 1717 PA 2 0
215 1718 NY 2 0
216 1719 CO 2 0
217 1720 CO 2 0
218 1724 PA 2 0
219 1727 FL 2 0
220 1731 TN 2 0
221 1732 NJ 2 0
222 1734 MI 2 0
223 1740 OH 2 0
224 1754 FL 2 0
225 1757 VA 2 0
227 1760 CA 2 0
228 1762 GA 2 0
229 1763 MN 2 0
230 1765 IN 2 0
232 1769 MS 2 0
233 1770 GA 2 0
234 1772 FL 2 0
235 1773 IL 2 0
236 1774 MA 2 0
237 1775 NV 2 0
238 1778 BC 2 0
239 1779 IL 2 0
241 1781 MA 2 0
243 1785 KS 2 0
244 1786 FL 2 0
247 1801 UT 2 0
248 1802 VT 2 0
249 1803 SC 2 0
250 1804 VA 2 0
251 1805 CA 2 0
252 1806 TX 2 0
254 1808 HI 2 0
256 1810 MI 2 0
257 1812 IN 2 0
258 1813 FL 2 0
259 1814 PA 2 0
260 1815 IL 2 0
261 1816 MO 2 0
262 1817 TX 2 0
263 1818 CA 2 0
265 1828 NC 2 0
267 1830 TX 2 0
268 1831 CA 2 0
269 1832 TX 2 0
270 1843 SC 2 0
271 1845 NY 2 0
272 1847 IL 2 0
273 1848 NJ 2 0
274 1850 FL 2 0
275 1856 NJ 2 0
276 1857 MA 2 0
277 1858 CA 2 0
278 1859 KY 2 0
279 1860 CT 2 0
280 1862 NJ 2 0
281 1863 FL 2 0
282 1864 SC 2 0
283 1865 TN 2 0
288 1870 AR 2 0
291 1878 PA 2 0
294 1901 TN 2 0
296 1903 TX 2 0
297 1904 FL 2 0
299 1906 MI 2 0
300 1907 AK 2 0
301 1908 NJ 2 0
302 1909 CA 2 0
303 1910 NC 2 0
304 1912 GA 2 0
305 1913 KS 2 0
306 1914 NY 2 0
307 1915 TX 2 0
308 1916 CA 2 0
309 1917 NY 2 0
310 1918 OK 2 0
311 1919 NC 2 0
312 1920 WI 2 0
313 1925 CA 2 0
314 1928 AZ 2 0
315 1931 TN 2 0
316 1936 TX 2 0
317 1937 OH 2 0
319 1940 TX 2 0
320 1941 FL 2 0
321 1947 MI 2 0
322 1949 CA 2 0
323 1951 CA 2 0
324 1952 MN 2 0
325 1954 FL 2 0
326 1956 TX 2 0
327 1970 CO 2 0
328 1971 OR 2 0
329 1972 TX 2 0
330 1973 NJ 2 0
331 1978 MA 2 0
332 1979 TX 2 0
333 1980 NC 2 0
334 1985 LA 2 0
335 1989 MI 2 0
349 01144 UK 2
367 01164 NZ 2
543 2
544 2
545 2
546 2
547 2
: 2
998 2
999 2
LICENSED FEATURES
IP USERS LICENSE IP8: License to enable 8 IP Users
VMS LICENSE License to enable Voice Mail System
Abbreviated Dialing
Account Code
Alarms
Auto Redial
Auto Redial 17
Barge-In
Barge-In 4
Call Chaining
Call Forward-If Busy-All Calls to External Number 132-Trunk Access Code-Dest. Number-#*
Call Forward-If No Reply-All Calls to External Number 133-Trunk Access Code-Dest. Number-#*
Call Hold
Call Park
Call Pick Up
Call Toggle
Call Transfer
Conference 3-Party
Conference-Unsupervised Flash-#
Conference Dial-In
Conference Multiparty
Conversation Recording
Department Call
Do Not Disturb
DND Override 4
Door Phone
Set Door Phone to route calls to an extension 1172-Access Code of Door Phone-1
Set Door Phone to route calls to an external number 1172-Access Code of Door Phone-2
Dynamic Lock
Flashing on Trunks
Floor Service
Follow Me
Forced Answer
Forced Answer 5
Hot Desking
Hotline
Interrupt Request
Interrupt Request 3
Meet Me Paging
Mute
Mute 1052
Operator
Call to Operator 9
Paging
Presence
Raid
Raid 5
RCOC
Reminder
Room Monitor
SA Command 1072
Exit SE Mode 00
Trunk Reservation
Reserve a Trunk 6
User Absent/Present
User Password
Voice Help
Voice Mail
To Walk-Out 111-0
BH Break Hours
BI Barge-In
CD Carrier Detect
CO Central Office
EID Exchange ID
ENQ Enquiry
FM Frequency Modulation
GND Ground
IC Incoming call
IP Internet Protocol
IR Interrupt
LA Left Align
LE Local Exchange
MS Mobile Station
OG Outgoing
PC Personal Computer
PS Power Supply
QSIG Q-Signaling
RA Right Align
RF Radio Frequency
RI Ring Indicator
SA System Administrator
SE System Engineer
SP Service Provider
WH Working Hours
Matrix Comsec Pvt. Ltd. (Matrix) warrants to its consumer purchaser any of its products to be free of defects in
material, workmanship and performance for a period of 15 months from date of manufacturing or 12 months from
the date of installation which ever is earlier.
During this warranty period, Matrix will at its option, repair or replace the product at no additional charge if the
product is found to have manufacturing defect. Any replacement product or part(s) may be furnished on an
exchange basis, which shall be new or like-new, provided that it has functionality at least equal to that of the
product, being replaced. All replacement parts and products will be the property of Matrix. Parts repaired or
replaced will be under warranty throughout the remainder of the original warranty period only.
1. Products that have been subjected to abuse, accident, natural disaster, misuse, modification, tampering,
faulty installation, lack of reasonable care, repair or service in any way that is not contemplated in the
documentation for the product or if the model or serial number has been altered, tampered with, defaced or
removed.
2. Products which have been damaged by lightning storms, water or power surges or which have been
neglected, altered, used for a purpose other than the one for which they were manufactured, repaired by
customer or any party without Matrix’s written authorization or used in any manner inconsistent with
Matrix’s instructions.
4. Products damaged due to operation of product outside the products’ specifications or use without
designated protections.
Warranty Card:
• When the product is installed, please return the warranty card with:
• Matrix assumes that the customer agrees with the warranty terms even when the warranty card is not
signed and returned as suggested.
The Purchaser shall have to bear shipping charges for sending product to Matrix for testing/rectification. The
product shall be shipped to the Purchaser at no-charge if the material is found to be under warranty. The Purchaser
shall have to either insure the product or assume liability for loss or damage during transit.
Matrix reserves the right to waive off or make any changes in its warranty policy without giving any notice.
In no event will Matrix be liable for any damages including lost profits, lost business, lost savings, downtime or
delay, labor, repair or material cost, injury to person, property or other incidental or consequential damages arising
out of use of or inability to use such product, even if Matrix has been advised of the possibility of such damages or
losses or for any claim by any other party.
Except for the obligations specifically set forth in this Warranty Policy Statement, in no event shall Matrix be liable
for any direct, indirect, special, incidental or consequential damages whether based on contract or any other legal
theory and where advised of the possibility of such damages.
Neither Matrix nor any of its distributors, dealers or sub-dealers makes any other warranty of any kind, whether
expressed or implied, with respect to Matrix products. Matrix and its distributors, dealers or sub-dealers specifically
disclaim the implied warranties of merchantability and fitness for a particular purpose.
This warranty is not transferable and applies only to the original consumer purchaser of the Product. Warranty shall
be void if the warranty card is not completed and registered with Matrix within 30 days of installation.
All legal course of action subjected to Vadodara (Gujarat, India) Jurisdiction only.
• The source of the open source software used in this product is available on CD, upon written request from:
R&D Team
Matrix Comsec Pvt Ltd
394, Makarpura GIDC,
Vadodara - 390 010
Gujarat
India.
Customer shall bear the shipping and handling charges.
Preamble
The licenses for most software are designed to take away your
freedom to share and change it. By contrast, the GNU General Public
License is intended to guarantee your freedom to share and change free
software--to make sure the software is free for all its users. This
General Public License applies to most of the Free Software
Foundation's software and to any other program whose authors commit to
using it. (Some other Free Software Foundation software is covered by
the GNU Lesser General Public License instead.) You can apply it to
your programs, too.
Also, for each author's protection and ours, we want to make certain
that everyone understands that there is no warranty for this free
software. If the software is modified by someone else and passed on, we
want its recipients to know that what they have is not the original, so
that any problems introduced by others will not reflect on the original
authors' reputations.
You may charge a fee for the physical act of transferring a copy, and
you may at your option offer warranty protection in exchange for a fee.
2. You may modify your copy or copies of the Program or any portion
b) You must cause any work that you distribute or publish, that in
whole or in part contains or is derived from the Program or any
part thereof, to be licensed as a whole at no charge to all third
parties under the terms of this License.
3. You may copy and distribute the Program (or a work based on it,
under Section 2) in object code or executable form under the terms of
Sections 1 and 2 above provided that you also do one of the following:
The source code for a work means the preferred form of the work for
making modifications to it. For an executable work, complete source
code means all the source code for all modules it contains, plus any
associated interface definition files, plus the scripts used to
control compilation and installation of the executable. However, as a
special exception, the source code distributed need not include
anything that is normally distributed (in either source or binary
form) with the major components (compiler, kernel, and so on) of the
operating system on which the executable runs, unless that component
itself accompanies the executable.
5. You are not required to accept this License, since you have not
signed it. However, nothing else grants you permission to modify or
distribute the Program or its derivative works. These actions are
prohibited by law if you do not accept this License. Therefore, by
modifying or distributing the Program (or any work based on the
Program), you indicate your acceptance of this License to do so, and
all its terms and conditions for copying, distributing or modifying
the Program or works based on it.
6. Each time you redistribute the Program (or any work based on the
Program), the recipient automatically receives a license from the
original licensor to copy, distribute or modify the Program subject to
these terms and conditions. You may not impose any further
restrictions on the recipients' exercise of the rights granted herein.
You are not responsible for enforcing compliance by third parties to
this License.
9. The Free Software Foundation may publish revised and/or new versions
of the General Public License from time to time. Such new versions will
be similar in spirit to the present version, but may differ in detail to
address new problems or concerns.
NO WARRANTY
<one line to give the program's name and a brief idea of what it does.>
Copyright (C) <year> <name of author>
You should have received a copy of the GNU General Public License along
with this program; if not, write to the Free Software Foundation, Inc.,
51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
Also add information on how to contact you by electronic and paper mail.
The hypothetical commands `show w' and `show c' should show the appropriate parts
of the General Public License. Of course, the commands you use may be called
something other than `show w' and `show c'; they could even be mouse-clicks or
menu items--whatever suits your program.
You should also get your employer (if you work as a programmer) or your
school, if any, to sign a "copyright disclaimer" for the program, if
necessary. Here is a sample; alter the names:
This General Public License does not permit incorporating your program into
proprietary programs. If your program is a subroutine library, you may
consider it more useful to permit linking proprietary applications with the
library. If this is what you want to do, use the GNU Lesser General
Public License instead of this License.