2N NetStar

www.2n.cz

2N NetStar®Communication System

HS NetStar manual Version: 4.4

The 2N TELEKOMUNIKACE a.s. is a Czech manufacturer and supplier of telecommunications equipment.

The product family developed by 2N TELEKOMUNIKACE a.s. includes GSM gateways, private branch exchanges (PBX), and door and lift communicators. 2N TELEKOMUNIKACE a.s. has been ranked among the Czech top companies for years and represented a symbol of stability and prosperity on the telecommunications market for almost two decades. At present, we export our products into over 120 countries worldwide and have exclusive distributors on all continents.

2N is a registered trademark of 2N TELEKOMUNIKACE a.s. Any product and/or other ®names mentioned herein are registered trademarks and/or trademarks or brands protected by law.

2N TELEKOMUNIKACE a.s. administers the FAQ database to help you quickly find information and to answer your questions about 2N products and services. On www.faq.2n.cz you can find information regarding products adjustment and instructions for optimum use and procedures „What to do if...".

2N TELEKOMUNIKACE a.s. hereby declares that the 2N product complies with all ®basic requirements and other relevant provisions of the 1999/5/EC directive. For the full wording of the Declaration of Conformity see the CD-ROM (if enclosed) or our website at www.2n.cz.

The 2N TELEKOMUNIKACE a.s. is the holder of the ISO 9001:2009 certificate. All development, production and distribution processes of the company are managed by this standard and guarantee a high quality, technical level and professional aspect of all our products.

2N TELEKOMUNIKACE a.s., www.2n.cz 3/625

Introduction

1. About Help

2. Terms and Symbols

2N® NetStar Hardware Manual

1. System Installation

2. Technical specifications

3. Power Source

4. Technical parameters

NetStar Admin Tool Manual

1. About Application

2. Hardware

3. Virtual Ports

4. SIM

5. Network

6. Global data

7. Routing

8. Users

9. Setting Properties

10. Billing and Tariffs

11. Configuration Examples

2N® NetStar Billing Data

2N® NetStar CTI documentation

2N® NetStar Assistant Manual

1. About 2N® NetStar Assistant Application

2. Launching Application

3. Quick Navigation

4. Telephone Parameter Settings

5. Tray Launcher

6. Advanced Settings

2N® Contact Centre Solution Manual

1. 2N® Contact Centre Solution Installation Wizard

2. 2N® Contact Centre Solution Configuration

3. 2N® Contact Centre Solution User Manual

2N® NetStar Records Manager Manual

1. 2N® NetStar Records Manager Installation Wizard

2. 2N® NetStar Records Manager Configuration

3. 2N® NetStar Records Manager User Manual


2N® NetStar User Manuals

1. Incoming and Outgoing Calls

2. Connection Services

3. Secured Services

4. Time Related Services

5. VoiceMail

6. Profiles

7. Other Services

8. SMS Operated Services

9. Table of Service

10. 2N® StarPoint manual

11. Manuál 2N® OpenStage 30T


13. 2N® Mobility Extension

Supplementary Information

1. Troubleshooting

2. Directives, Laws and Regulations

3. General Instructions and Cautions


Introduction

NetStar2N® is a new generation of PBX which supports VoIP technology. It works like

a normal PBX and VoIP softswitch and fully connects services. It enables a number of devices to be used such as analogue telephone lines ISDN BRI, ISDN PRI and VoIP. The individual networks' properties are not separated but integrated into one compact system.

The system's call capacity corresponds to the needs of large companies and operators. Transfer between various networks is governed by dynamic least cost routing (LCR), which selects the cheapest and best route for making a call to the number required. Thanks to the well thought out LCR, in which every subscriber or individual port can have its own routing table, it is no problem to create a virtual PBX or simply create restrictions for certain subscriber groups or individuals.

The options for the subscriber terminals and communications are not dependent on the communications technology used. All extensions, from analogue phones to web terminals have the Web Assistant available for simple user configuration of the services. The Web Assistant combines web interfaces with voiced instructions and guides to the individual services. It is equally easy to operate the system phones by using the context menu. It is possible to define all of the phones' settings by means of the profiles, which can be easily created by the user or system administrator by means of the guide.

Here is what you can find in this chapter:

1. About Help



1. About Help

Document FormatDocument version: 4.3.0

This document serves as a Help and Manual for the configuration of the 2N® NetStar

communication system by the program. reserves the right to NS Admin 2Nmodifications.

ConventionsThe following fonts are used in the text:

#Hypertext – Hyperlink to another place in the document or outside the document

Important words – The text used in NSAdmin and important words in the text we want to emphasise are highlighted.

– This form is used instead of a verbal description of the location of HW – Rackan item in the TreeView (tree in the left-hand part of the NsAdmin window). It means that in the TreeView you need to click the HW item and then the Rack item in it.

PartHeadings of the individual page topics listed in the table of contents. You can click this heading and jump to the top (table of contents) of the page where other topics can be selected.

Important subpartsUsed for better clarity. It separates logical units of the menu – for example, separate section of configuration parameters.

Section of subpartsUsed for better clarity. It separates logical units of subparts.



Manual SymbolsThe following symbols and pictograms are used in the manual:

Safety

Always abide by this information to prevent persons from injury.

Warning

Always abide by this information to prevent damage to the device.

Caution

Important information for system functionality.

Tip

Useful information for quick and efficient functionality.

Note

Routines or advice for efficient use of the device.

Future Functions, InnovationsThe in this document designates the functions that are under grey-marked text preparation or development at present.


2N® NetStar Hardware Manual

Content:

1. System Installation

1.1 Select of the Installation part

1.2 Unwrapping and Control

1.3 Installation

1.3.1 System Connection

1.3.2 System Installation

1.4 Terms and Symbols

2. Technical specifications

2.1 Basic Module

2.1.1 Basic Module

2.1.2 Basic Module Pro

2.2 Extension Module

2.3 Position and Port Numbering

2.4 CPU Card

2.5 CPU Card - Extension module

2.6 SWITCH card

2.7 PRI Card

2.8 BRI Card

2.9 DVL card

2.10 DIGITAL COMBO Card

2.11 4ASL/4AVL Card

2.12 AVL Card

2.13 GSM Card

2.14 UMTS Card

2.15 Audio Card_IO_Relay

2.16 IP Card


3. Power Source

3.1 Power Supply Connection

3.2 System Grouping

3.3 Power supply backup

4. Technical parameters


1. System InstallationHere is what you can find in this section:

1.1 Select of the Installation part


1.3 Installation


Selection of the Installation SiteThe following aspects shall be taken into consideration while selecting a site for the system installation:

Good accessibility:

All interfaces for the connection of the telecommunication interfaces or data network connections, as well as the switch-on, switch-off and system reset controls are placed on the front side.

Protection against humidity and extreme temperatures:

The appliance may never be placed close to heat sources (radiators) or places exposed to direct sunshine. Also places with high humidity (such as bathrooms and cellars), places with significant temperature fluctuation (next to doors, windows, air-conditioning), dusty places (workshops, etc.) or places exposed to aggressive gases (accumulator room, boiler room) as well as places with intensive vibrations and places exposed to shocks (compressor rooms, heavy industrial operations) should be avoided. The system should be installed horizontally.

Installation of infrastructure in the building:

The system was designed for installation in a 19" rack and connection to the structured wiring infrastructure with termination at a CAT 3 or CAT 5 patch panel. If the infrastructure is already ready the switchboard installation site is usually given and can not be further changes. The interface must however be adjusted for the connection via RJ 45 8/8 plugs.

GSM signal quality:

If a GSM card is a part of your system (or will become later on), it is necessary to consider that the cable leading from the GSM card to the GSM antenna is usually 3m long (max. 10m). You should ensure that within this diameter from the selected system installation site a place with good GSM signal suitable for placement of the GSM antenna is available. *)


Theft protection:

While selecting the installation site the insurance company conditions should be regarded as well.

Caution

In the areas with very poor GSM signal level or high number of GSM gates the use of internal (takes the position of a line card) or external GSM splitter is advantageous.


1.1 Select of the Installation partThe following aspects shall be taken into consideration while selecting a site for the system installation:

Good accessibility – All interfaces for the connection of the telecommunication interfaces or data network connections, as well as the switch-on, switch-off and system reset controls are placed on the front side.

Protection against humidity and extreme temperatures – The appliance may never be placed close to heat sources (radiators) or places exposed to direct sunshine. Also places with high humidity (such as bathrooms and cellars), places with significant temperature fluctuation (next to doors, windows, air-conditioning), dusty places (workshops, etc.) or places exposed to aggressive gases (accumulator room, boiler room) as well as places with intensive vibrations and places exposed to shocks (compressor rooms, heavy industrial operations) should be avoided. The system should be installed horizontally.

Installation of infrastructure in the building – The system was designed for installation in a 19" rack and connection to the structured wiring infrastructure with termination at a CAT 3 or CAT 5 patch panel. If the infrastructure is already ready the switchboard installation site is usually given and can not be further changes. The interface must however be adjusted for the connection via RJ 45 8/8 plugs.

GSM signal quality – If a GSM card is a part of your system (or will become later on), it is necessary to consider that the cable leading from the GSM card to the GSM antenna is usually 3m long (max. 10m). You should ensure that within this diameter from the selected system installation site a place with good GSM signal suitable for placement of the GSM antenna is available. *)

Theft protection – While selecting the installation site the insurance company conditions should be regarded as well.

Note

In the areas with very poor GSM signal level or high number of GSM gates the use of internal (takes the position of a line card) or external GSM splitter is advantageous.



Caution

If you have not already done so, please check the completeness of the delivery.

The wrapping of the basic or extension module is usually used also for the transportation of other, separately acquired system components (antennas and other small additional equipment). These can be checked by matching the items on the supply list and the ordering numbers stated on all components. Usually, the system is delivered as tailor made according to the submitted order. The installation set, electronic documentation on a CD and warranty certificate are a part of the delivery. The remaining components such as telephone units and other terminals or additional appliances are delivered separately.


1.3 InstallationHere is what you can find in this section:

1.3.1 System Connection

1.3.2 System Installation

1.3.1 System ConnectionThe system is installed into a deep. The height of each module is RACK min. 400 mm 3U–132 mm.

The system was not designed to be equipped with an active cooling unit, however the producer recommends in order to achieve a better air ventilation in case of bigger installations (more modules above each other) to leave a gap of e.g. 1U between the separate modules. For this purpose an organizer, which at the same time will ensure a better manipulation with the wiring, can be used.

For a shelf installation the system is equipped with 4 stands that create a gap between the shelf and the module body to ensure sufficient system ventilation.

1.3.2 System InstallationThe same applies for the connection to the 230 V or 115 V power supply Network, which is subject to the local and national regulations and directives.

The connection of the system to the telecommunication network is subject to local national regulations and directives. The interface of the line cards terminates in a RJ45 plug.

Only appliances complying with the basic requirements and further specifications of the directive no. 1999/5/ES (Government Directive No. 426/200 Coll., which stipulates the technical requirements of radio and telecommunication terminal appliances, in the wording of the later regulations) may be connected to the internal analogue interface.

The supplied wiring including an additional connection to the single ground point is used for connection. If you do not use the additional system grounding, the over voltage protection integrated in the line cards will not work.

The communication system was designed as digital appliance (TDM circle 2N NetStar®switching) with sufficiently powerful processor and LAN connection with integrated VoIP telephony.




Safety


Warning


Caution


Tip


Note




2. Technical specificationsHere is what you can find in this section:

2.1 Basic Module

2.1.1 Basic Module


2.2 Extension Module

2.3 Position and Port Numbering

2.4 CPU Card

2.5 CPU Card - Extension module

2.6 SWITCH card

2.7 PRI Card

2.8 BRI Card

2.9 DVL card

2.10 DIGITAL COMBO Card

2.11 4ASL/4AVL Card

2.12 AVL Card

2.13 GSM Card

2.14 UMTS Card

2.15 Audio Card_IO_Relay

2.16 IP Card


2.1 Basic ModuleHere is what you can find in this section:

2.1.1 Basic Module



2.1.1 Basic ModuleThe Basic Module Lite represents a business alternative of the basic module equipped with an CPU card, Switch X card that does not enable the connection of extension modules and one AVL card.

Ord. no. 101034 2N NETSTAR BASIC rack basic module 8x AVL

Ord. no. 101014 2N NETSTAR BASIC module 8x ASL

2.1.2 Basic Module ProBasic Module Profi is a business alternative of the basic module equipped with a CPU card, Switch card that enables the connection of up to 4 extension modules.

Ord. no. 101031 2N NETSTAR PRO rack basic module (empty)Ord. no. 101011 2N NETSTAR PRO basic module (empty)


2.2 Extension ModuleThe extension module is equipped with an 8-bit CPU, which controls up to 11 analogue line cards. The following cards are supported: AVL, ASL, GSM and AUDIORELE.

Ord. no. 101036 2N NETSTAR PRO rack extender analogue onlyOrd. no. 101016 2N NETSTAR PRO extender analogue only


2.3 Position and Port NumberingEach card and port is assigned a specific unique HW address.

The address consists of 4 parts and is stated in the form: R : C : B : P, whereas the separate letters stand for the following:

R – rack

Rack represents the entire system appliance consisting of the basic module and four extension modules (case), in the current version only one Rack is supported. The rack address is thus 0.

C – case

Covers the basic or one of the four extension modules.

The address of the basic module is 1, the extension modules are assigned addresses from 2 to 5 according to their SWITCH card Connections.

B – board

The separate cards (board) are assigned addresses from left to the right, both in the basic as well as the extension modules. The address of the first position in the module is 1 and is reserved for maincase CPU or for extender CPU.

Compared to the extension module, in which the cards are assigned addresses from 1 to 12, the assigning of addresses in the basic module is governed by predefined rules.

The basic module uses following rules for the card positions and assigning of addresses.

Address 1

Reserved for controlling CPU only (position 1)

the card address is 0:1:1

Addresses 2,3,4

Reserved for the PRI and VoIP cards (positions 2,3 and 4)

the cards addresses are 0:1:2, 0:1:3 and 0:1:4

Address 5

Reserved for the SWITCH card (position 5)

the card address is 0:1:5

Addresses 6,7,8,9,10,11,12,13 and 14

The addresses correspond to the positions 6,7,8,9,10,11,12,13 and 14 and can be used for any line card. An exception are the positions with the addresses 6, 9 and 12, in which if you use a BRI or DIGITAL COMBO card the first 2 ports on the card are always NT and may not be changed.


P – port

The card ports are assigned the addresses from left to the right and from top to the bottom.

The addresses start with 1.

For the purpose of better orientation the PRI, GSM and Audio/IO/Relay cards have the serial numbers of the ports printed on their front side.


2.4 CPU CardThe card is an inseparable part of the basic module.

It contains all necessary interfaces needed for the communication with the system.

It is only installed in the basic module, always in the position with the 0 HW address.

LAN

Interface type – Eth T-Base 10/100

Connection – RJ45

Pin1= transit +,

Pin2= transit -,

Pin3= receive +,

Pin6= receive -

USB


USB

Interface type – USB 1.1

Connection – 2x USB-A

Physical layer – Low Speed – 1.5Mbits/s

Power supply – 5V

Consumption – max. 100mA

COM

RS 232C interface connection

Interface type – RS232C

Configuration – 115200Bd, 8bit, parity: none, stop bit: 1

Connection – RJ11 (6x6)

Pin1= TxD -,

Pin2= TxD +,

Pin3,4= GND,

Pin5= RxD +,

Pin6= RxD -


MMC

MMC cards slot – MMC 7pins

MMC 4.0 and MMC Plus 13 pins

Buses – 1-bit (MMC) and 4-bit (MMC4.0 and MMC+)

Power supply – 3,3V

Reading speed – up to 22, Mb/s

Writing speed – up to 18 Mb/s

Maximum recommended capacity – 4 GB

RESThe Reset button will reboot the entire system.

Maintenance switch

ON position – appliance switched on

OFF position – appliance switched off, the position is equipped with a 3s by-pass for the switching into the Maintenance Mode

Maint. position – appliance in the maintenance mode, the line cards supply switched off and the card drivers stopped in order to enable their maintenance or exchange, or changes in the port set-up.

After the switching into the ON position, the card drivers are launched and their power supply is switched on – no need for the re-launching of the entire system and starting of all processes.


2.5 CPU Card - Extension moduleThe card is an inseparable part of the extension module.

Contains the proprietary interface for the communication with the .SWITCH card

It is only installed in the extension module, always into the positions with the 0 HW address.

The SWITCH board connection interface

Interface type – proprietary

Connection – RJ45 (8/8)

Maximum connection length between SWITCH card and extension modul CPU card is 160 cm (wiring supplied as a part of the extender package)

Total interface throughput

between SWITCH card and extension module CPU card is available 31 channels (and 1 for signaling)

total extension module throughput is 88 calls


2.6 SWITCH cardThe SWITCH card is an inseparable part of the basic module.

It can be supplied in two HW alternatives.

Switch X without the possibility of extension modules connection, standardly supplied as a part of the Basic Module Lite version.

Switch with the possibility of the connection of up to 4 extension modules.

Both card versions are always equipped with 1 PRI interface and basic connection field that is in charge of the connection of call in the entire system.

It is only used in the basic module and is always installed in the position with 5 as the HW address.


The set-up of the Jumpers on the board determines the NT or TE port type

NT port set-up

For NT port the pins 1–3, 2–4, 5–7, 6–8 should be connected

TE port set-up

For TE port the pins 1–2, 3–4, 5–6, 7–8 should be connected.

Extender connection interface

Interface type – proprietary

Connection – RJ45 (8x8)

Maximum connection length – 160cm (wiring supplied as a part of the extender package)

Total interface capacity – 124 calls

Ratio capacity of the interface – 4x 31 calls

Maximum capacity per 1 extender 88 calls


1x PRI interface

Interface type – S0 without power supply / NT–TE

Protocol – DSS1 – EURO ISDN

Connection – RJ45 with LED control

Pin1, Pin2 = broadcasting couple

Pin4, Pin5 = reception couple

LED control lamp

– port not allowed by the licence RED

NO LIGHT – port allowed by the licence, the line is not connected

– communication interface connected to the 2nd layer GREEN


Connection field

Type – ZARLING

Total capacity 512 channels

Maximum number of simultaneous calls up to 172 – according to configuration

CPU: 0,1,2,3

Pozice 2 on PRI = 0 1 on PRI = 1 2 on PRI = 2 0 on PRI = 3 3



Switch: 0,1,2,3

The card ports are marked blue, the Switch buses are marked red.

The CPU card and the SWITCH card are filled 1:1.

If more than 3 ports are occupied on the card or cards in the positions 2, 3 and 4, the PRI cards must be equipped with another ZARLING connection field. This ensures that the call channels of all ports are connected to the basic connection field of the SWITCH card.


2.7 PRI CardThe PRI card can be supplied in several HW configurations.

The card can be configured to contain 1, 2 or 4 PRI ports, with or without included ZARLING connection field.

If more than 3 PRI ports (the SWITCH card PRI port not including) are to be served by the system, the system needs to be equipped with the ZARLINK connection field cards.

The factory pre-set of the ports is TE.

Attention: the NS Admin programming tool uses these port settings during the HW activation.

The card is installed into the position 1, 2 and 3 of the basic module, in between the CPU and SWITCH cards.



NT port set-up

For NT port the pins 1–3, 2–4, 5–7, 6–8 should be connected

TE port set-up

For TE port the pins 1–2, 3–4, 5–6, 7–8 should be connected


1.

2.

1 to 4 x PRI port

Interface type – S0 without power supply / NT–TE


Connection – RJ45 with LED control lamp

TE modus



NT modus



LED control lamp

RED – port not allowed by the licence

NO LIGHT – port allowed by the licence, the line is not connected

GREEN – communication interface connected to the 2nd layer


2.8 BRI Card

The BRI card services 8 fully functional BRI ports. The port configuration is TE or NT, with PTP or MPT line interface support.

On the NT interface the power supply can be switched off – 42V.

The factory settings of the ports are as follows:

ports 0,2,4 and 6 preset as NT with power supply

ports 1,3,5 and 7 preset as TE

if you desire to use the BRI ports for the synchronisation, no line card position may be skipped between this card and the SWITCH card.

The card is installed into the basic module on the line card position.

Caution

The NS Admin programming tool uses these port settings during the HW activation..



NT

Pins 1–2, 3–4, 5–6, 7–8 and power supply pins to be connected

NT without power supply

Connect pins 1–2, 3–4, 5–6, 7–8 and do not connect the power supply pins

TE


Attention:

on the positions with the HW addresses 5, 9 and 13 the ports 0 and 1 always have the NT configuration.


8x BRI port

Interface type – S0 (NT–S/T, LT–T) with phantom power supply – 42V

S0 (TE–S/T, LT–S) without power supply

*Protocol * – DSS1 – EURO ISDN

Connection

RJ45

TE modus



NT modus



Order no. 1011118 2N NETSTAR module 8x BRI


2.9 DVL cardThe DVL card contains 8 ports used for the connection of the Star Point digital telephone sets.

the board always contains two sources of power per each four ports 0,1,2,3 a 4,5,6,7. If there occurs a short-circuit in the wiring (e.g. because of connection of an analogue phone), the power supply will be disconnected. After the technical deficiency was repaired, the power supply source will start to feed power into the ports again.

The card is only installed into the position of the line card of the basic module.


8x DVL port

Interface type

Upn (NT–U), LAP–S with power supply 48V

Protocol

Cornet II

Connection

RJ45

Pin4, Pin5

broadcasting/ reception

Order no. 1011128 2N NETSTAR module 8x DVL


2.10 DIGITAL COMBO CardThe Digital Combo card consists of 4 BRI ports and 4 DVL ports used for the connection of the digital telephone sets.

The specifications and characteristics of the ports match the specifications and characteristics of the card itself.

If you desire to use the BRI port for synchronisation purposes, no line card position may be skipped between this card and the SWITCH card.

The card is installed into the line card position of the basic module.



NT

Pins 1–2, 3–4, 5–6, 7–8 and power supply pins to be connected

NT without power supply


TE


Caution

On the positions with the HW addresses 5, 9 and 13 the ports 0 and 1 always have the NT configuration.

4x DVL port


4x DVL port

Interface type – Upn (NT–U), LAP–S with 48V power supply

Protocol – Cornet II

RJ45 Connection – two conductors, Pin4 and Pin5 = broadcasting/ reception

4x BRI port

Interface type

S0 (NT–S/T, LT–T ) with phantom power supply –42V

S0 (TE–S/T, LT–S ) without power supply


Connection – RJ45 – four conductors



Order no. 1011124 2N NETSTAR module 4x BRI/4x DVL


2.11 4ASL/4AVL CardThe card contains 4 ASL ports and 4 AVL ports.

In case of power supply shortage the ASL ports are 1:1 switched to the AVL ports.

We recommend to only connecting the telephone sets, which do not need additional power supply and are in compliance with the basic requirements and further respective provisions and stipulations of the 1999/5/ES Directive to the AVL ports.

The card can be installed into any line position of the basic or the extension modules.


4 x CO port

Interface type – analogue with power supply –60V

Signalisation – reverse polarity,

CLIP broadcasting – FSK, DTMF

Connection – RJ45 – two-wire

Pin4, Pin5 = a/b wires

Order no. 1011224 2N NETSTAR module 4xCO / 4x ASL


2.12 AVL CardThe card contains 8 AVL ports.

Support of broadcasting of CLIP into the connected telephone sets via DTMF or FSK.

Support of status signalisation, for pulse signalisation a constant polarity reverse is generated.

The card is installed into any line position of the basic or the extension modules

Order no. 1011218 2N NETSTAR module 8x ASL

8x ASL port

Interface type – U analogue with power supply –60V

Signalisation – reverse polarity,

CLIP broadcasting – FSK, DTMF

Connection – RJ45 – two-wire

Pin4 = GND, 0V

Pin5 = –60V


2.13 GSM CardThe GSM card can be supplies in several HW configurations.

Configuration with 1, 2 or 4 GSM ports (channels)

The exchange of the SIM card in the GSM modules is only allowed if the card power supply is switched off, i.e. if the system is switched to the maintenance mode or completely switched off.

An antenna must be connected to the SMA antenna connection; otherwise GSM module damage may occur.

1x SIM for 1 GSM module is supported

The card is installed into any line position of the basic or the extension modules.

GSM module

Module type – SIEMENS TC35i, SIEMENS MC39

SIM

SIM card type – SIM small 3V


2.14 UMTS CardThe UMTS card can be supplies in several HW configurations.

Configuration with 2 or 4 UMTS ports (channels)

The UMTS card is equipped with antenna splitter.

An antenna must be connected to the SMA antenna connection; otherwise UMTS module damage may occur.

The exchange of the SIM card in the UMTS modules is only allowed if the card power supply is switched off, i.e. if the system is switched to the maintenance mode or completely switched off.

1x SIM for 1 UMTS module is supported


UMTS module

Module type – SIERRA WIRELESS MC8795V

SIM

SIM card type – SIM small 3V


2.15 Audio Card_IO_RelayThe card can be supplied in two HW configurations.

The card configuration is 2x AUX and 4 x IO/Relay Or 4 x AUX and 8 x IO/Relay


4 or 8x IO-relay port

Interface type – IO/relay

Port function – according to chart

Connection – removable WAGO connector, 8 pins

1port – consisting of a pin couple

2 or 4x AUX port

Interface type – analogue

Connection – Jack plug 3,5 mm stereo

Pin0 (back ring) = GND

Pin1 (middle ring) = Right channel

Pin2 (front ring) = Left channel

Input resistance: 160–380 kOhm (type 270kOhm)

Input voltage range: max + - 2,223V AC

Output resistance: type 0.25Ohm

Output voltage range: max + - 2,223V AC




VoIP Card

The VoIP card can be supplies in several HW configurations.

4, 8, 16, 24 or 32 voice channels.

(IEEE 802.3) Ethernet 10/100 FullDuplex

The card is installed into the position 1, 2 and 3 of the basic module, in between the CPU and SWITCH cards.


3. Power Source

Safety

These parts are connected with the power supply network and operate with a life dangerous voltage. Any repairs and exchanges of the parts may only be done by the producer!! Only the following maintenance works are permitted:

Safety fuse exchange

Connecting and disconnecting of the power supply cable

Such maintenance works may only be executed while the system is not connected to the power supply network!!!

Here is what you can find in this section:

3.1 Power Supply Connection

3.2 System Grouping

3.3 Power supply backup

3.1 Power Supply ConnectionThe detachable power supply cable is used for the connection to a regular power outlet. If necessary, the cable may easily be disconnected from the switch-board. In case of the switch-board installation outside the 19" rack the power supply cable may be conducted to a power outlet either freely or through a cable ledge.


3.2 System GroupingOf course, the system is also grounded via the power supply network (thanks to the protective earth conductor – if this is connected to the outlet). However, such grounding is not considered sufficient – with regard to the danger resulting from the fact that the system may easily be disconnected from the power supply network (over-voltage stemming from any line may thus easily be spread to the system chassis and from there to any other line!).

ATTENTION!

The system should thus optimally be connected from the main switchboard through a yellow-green conductor with a diameter of not less than 4mm2 to the grounding ledge. In case of emergency the "earth" should be connected by such conductor (i.e. non-disconnectedly) to at least the closest power supply outlet (make sure that the outlet is connected correctly).

The conductor should be attached to the system using the earthed connector (on the backside of the system panel, marked with the grounding symbol) that is to be fixed properly. The grounding should be connected prior to the connection of any other wiring!!!

The grounding conductors are supplied with any package containing either basic or extension modules. The length of the conductor is for the basic and the extension unit 1500 mm. The ends of the conductor are equipped with a 4 mm diameter eyelet, one end of the basic module conductor has an 8 mm diameter eyelet.


Grounding conductor of the extension unit

Grounding conductor of the basic unit

3.3 Power supply backupFor the backup of the power supply in case of power shortage an on-line UPS appliance is used. The UPS is to be connected pursuant to the documentation supplied for the separate types and producers.

The following chart will help you to design an optimal UPS performance.

The values stated in the unconducting chart serve for information only, the

consumption and the performance of the system change according to 2N® NetStar

the current operational system load and the HW configuration of the separate modules.

UPS – 800W / 1000VA

Number of cases 1 2 3 4 5

Backup time 150 min 70 min 50 min - -

UPS – 1600W / 2200VA

Number of cases 1 2 3 4 5

Backup time 240 min 150 min 100 min 70 min 50 min

NSVL8 , NSCOVL FXS – MINIMUM SERIAL RESISTOR <= 20 OHMTable 1: NSVL8 , NSCOVL – FXS

2N TELEKOMUNIKACE a.s., www.2n.cz 53/625COVL – 7320F3 SL PACKAGE


COVL – 7320F3 SL PACKAGETable 2: CO – FXO

Table 3: COVL – FXO Itsp=max I



4. Technical parametersBasic technical specifications

Nominal power supply network voltage: 230 V or 115V

Power supply voltage tolerance: +/- 10 %

Nominal frequency: 50 Hz

Input: 160

Fuse: 10A

Recommended power backup source: –

External backup power source voltage: –

Max. backup source consumption: –

Typical operation time with fully charged batteries: –

Basic unit size: 482x133x310 mm (84HPx3U)

Basic unit weight: 12.5 kg

Extended unit size: 482x133x310 mm (84HPx3U)

Extended unit weight: 12.5 kg

Communication interface

LAN

Signals: Pin1=transit +, Pin2=transit -, Pin3= receive +, Pin6= receive -

Transmission speed: 10/100 Mb/s

Interface type: Eth T-Base 10/100

USB

Signály: USB -A

Transmission speed: 1,5 Mb/s

Interface type: 2xUSB v1.1

R232

Signals: Pin1= TxD-, Pin2= TxD+, Pin3aPin4= GND, Pin5= RxD+, Pin6= RxD-

Transmission speed: 115200 Bd

Interface type: RS 232C / 115200Bd,8bit ,parity=none, stop bit=1


MMC

Signals: MMC 7pins and MMC Plus 13 pins

Transmission speed: up/down 18/22 Mb/s

Interface type: 1-bit (MMC) and 4-bit bus

Climate conditions

Temperature range: +5...+45 °C.

Relative air humidity: max. 85 % (40 °C) without condensation

Premises description – basic characteristics: standard premises

Operational position: horizontal

Internal lines

Analogue input impedance: 600 Ω

Ringing voltage

Line feeding voltage: 60V

Line current: cca 25 mA

External lines

Analogue input impedance: 600 Ω

SS voltage of closed loop at the I power =?? mA

Maximum loop current: 65 mA

SS resistance during pulse: min. 1 MΩ

SS resistance during interruption: max. 320 Ω

Earth-leakage power when hanged-up: max. 100 μA

Ringing impedance: 2–10 kΩ

Ringing voltage detection: 10–25 V

Digital system lines

Input power: 48V

Call conduction impedance: 600 Ω

Number of connectible conductors: 2


ISDN BRI

ISDN BRI S0 TE: Interface type – S0 (TE–S/T, LT–S ) – for connection to VTS

Protocol: DSS1 EURO ISDN with power supply

Interface configuration: P–MP (point – several points) – supports MSN, P–P (point – point) – supports DDI

Connection: 4-wires

Connection type: RJ 45

ISDN BRI S0 NT: Interface type – S0 (NT–S/T, LT–T) – inner S0 BUS

Protocol: DSS1 EURO ISDN with power supply

Interface configuration: P–MP (point – several points), P–P (point – point)

Connection: 4-wires

:Connection type RJ 45

ISDN PRI

Typ rozhraní: S0

Protocol: DSS1 EURO ISDN

Synchronisation: Master/Slave

Connection: 4-wires

Connection type: RJ 45

GSM Gateway

Standard GSM900/1800, phase II

Voice transmission: EFR, FR (enhacced full rate, full rate)

Receiver sensitivity: -104 dBm

Transmitter output: Up to 2W during pulse

SMS support: TXT, PDU

SIM card: small , 3V

Antenna connection: SMA

VoIP

Specifications of the DTMF broadcasting selection:

Minimum/ maximum level for the assessment of the lower or higher frequency: According to Q23

Specifications of the pulse selection broadcasting:

Pulse duration: 60 ms

Interruption duration: 40 ms


Specifications of the pulse selection reception:

Pulse duration: 30 ms to 70ms

Interruption duration: 30 ms to 70ms

FLASH detection

FLASH broadcasting specifications: 100 ms

FLASH reception specifications: 80 ms...150 ms (Preset values, can not be changed)

FSK detection

FSK reception specifications: 40 dB

FSK broadcasting specifications: 40 dB

Other signal detection

Modem detection specifications: 1300 Hz

Facsimile detection specifications: 1100 Hz

Dial tone frequency: 425 Hz

Dial tone level: -7 dB

Busy tone frequency: 425 Hz

Busy tone level: -7 dB

Nominal transmission inhibition:

Analogue – Analogue connection: -7 dB

Analogue – Digital connection: -7 dB

Digital – Digital connection: 0 dB

Audio/IO/Rele

The rights of the producer to adjust the product specifications in order to improve the characteristics of the product as compared to the product documentation are reserved.

The product may only be used in compliance with the User's Manual and for purposes, for which the product was designed and manufactured.

After the product lifetime was terminated please dispose the product or its parts in compliance with the valid environmental protection regulations.


NetStar Admin Tool Manual

Content:


1.1 Connecting to PBX

1.2 Configuration Menu

1.3 PBX Activation

2. Hardware

2.1 Basic

2.2 Boards

2.3 Synchronisation

2.4 Board and Port List

3. Virtual Ports

3.1 BRI a PRI

3.2 Cornet

3.3 ASL

3.4 CO

3.5 GSM

3.6 SIP

3.7 SMTP

3.8 SMPP

3.9 Software and Dialler

3.10 Virtual Port Options

4. SIM

4.1 SIM cards

5. Network

5.1 Network Interface

5.2 Routing table

5.3 Service Settings


5.3.1 Time Synchronisation (NTP)

5.3.2 TFTP Root Storage

5.3.3 TCP/IP Communication Port

5.3.4 System Services

5.3.5 DHCP server

5.3.6 Directory Service (LDAP)

5.3.7 API

5.4 Dohledové služby

5.4.1 Event Reporter

5.4.2 Remote Control (SNMP)

5.5 DB Conncetors

6. Global data

6.1 Global Parameters

6.2 Emergency Calls

6.3 Localisation

6.4 Licences

6.5 Language Packages

6.6 Services

6.7 Conference Rooms

6.8 Active Conferences

6.9 Progress Tones

6.10 Ring Tones

6.11 AutoClip Parameters

6.12 Storage Manager

6.13 Scheduled Tasks

6.14 Status Control Parameters

6.15 DTMF

6.16 Causes

6.16.1 Cause Objects

6.16.2 User Causes

6.16.3 Cause Mapping Tables

6.17 Time Parameters

6.17.1 Date and Time

6.17.2 Time Conditions

6.17.3 Holidays

6.18 Assistant


6.18.1 Nastavení administrace

6.18.2 Uživatelská relace

7. Routing

7.1 Routers

7.2 External Routers

7.3 Complex Routers

7.4 Switch Routers

7.5 Routing Objects

7.5.1 Bundles

7.5.2 ACD groups

7.5.3 DISA/IVR objects

7.5.4 Contact Centres

7.5.5 Ring Group

7.5.6 Ring Tables

7.5.7 Modems

7.5.8 Sets

7.5.9 Audio Inputs and Outputs

7.5.10 Binary Inputs and Outputs

7.5.11 CallBack

7.5.12 Status Control Objects

7.5.13 HTTP SMS CallBack

7.6 Identification Tables

7.7 AutoClip router

8. Users

8.1 Users & Groups

8.2 User Rights

8.3 Extension Types

8.4 Extensions

8.5 Phone Directories

8.5.1 User Phone Directories

8.5.2 Group Phone Directories

8.5.3 Group Phone Directories (Generated)

8.5.4 Common Phone Directories

8.5.5 SIP Phone Directories

9. Setting Properties

9.1 Setting Properties



10. Billing and Tariffs

10.1 Billing and Tariffs

11. Configuration Examples

11.1 Other Useful Information

11.2 Mobility Extension Configuration

11.3 2N® NetStar Installation Guide



About Application

NSAdmin is a configuration tool that helps configure the communication 2N® NetStar

system, version 2. The application is designed for an x86 platform using the

WINDOWS operating system connected with through a network. It is 2N® NetStar

controlled by a mouse and, secondarily, a keyboard. NS Admin tool uses the TCP

connection or modem and communicates with basically via port 6992.2N® NetStar

As a necessary condition for using this configuration tool under the Windows XP OS, a service pack 2 and Framework v.3. have been installed. The program does not work without these components.

Configuration Tool Main MenuOnce the configuration tool has been started, a window is displayed helping to configure connections to PBXs, analyse traces and start up the Help.

The main menu offers the following options:

Admin

Settings – open a dialogue with the global configuration tool settings.

Language – choose one of the supported languages.

Exit – push this key to exit the configuration tool.

Trace – operations with previously saved traces are only available here at the current stage.

Load trace from file – upload the saved trace.

Add trace from file – add the saved trace to the current trace.

Start trace analyzer – push to open a trace analysis window.

Help – start the Help in the chosen language.

Configuration Tool Global SettingsThis dialogue includes three tabs with the following parameters:

XML script


XML script

Type of debugging – use this section to enable and define the range of displaying the xml trace in the configuration tool. The PBX trace cannot be set here.

Indent size – use this parameter to define the indent size for the xml trace.

View – this is an auxiliary function for trace analysis using database.

Show current window name – check this option to display the current window name in the right-hand bottom corner.

Show object IDs – check this option to display the current object Id in the right-hand bottom corner. This is an easy way how to know e.g. the port Id and find the port in the trace.

Separete windows

Open last created separate windows – define whether the open windows of the tool must be displayed after repeated login to the PBX.

Default window width and height – define the width and height of the open window.

General

Set colours of virtual ports – define the colour for each virtual port or disable this function.

Set colours of tables – define the background colours for tables or disable this function.

Set colours of logins – define the colour for each user login or disable this function.

Set colours of extensions – define the colour for each extension (SIP, external, etc.) or disable this function.

Click the button, which is available in each subtab of the Default tab, to set the output object colour.General

Advanced – enable the Ask before delete if object is used in configurationparameter to allow/restrict indication of the warning information when the router or routing objects are being removed from configuration.

Login

Copy devices structure from old version storage – restrict/allow copying of the PBX structure from the earliest configuration tool versions. If the PBX structure has already been copied, erase (not tick of) the checkbox to remove the displayed PBX.

Click to confirm the configuration setting and to exit the dialogue without OK Cancel saving.


1.1 Connecting to PBX


1.3 PBX Activation


1.1 Connecting to PBXThe figure below presents all icons of this section.

Figure: View of PBX Login Icons

Connect to PBX – use this icon to connect the configuration tool to the PBX via a selected connection.Icon meanings from the left:

Create group – create a group of PBXs on the same level as the selected object or nested into the existing group.

Create PBX – create a PBX on the same level as the selected object or nested into the existing group.

Create connection – create a connection to the selected PBX.

Properties – set or change the properties of a selected object. A name is only assigned to groups. For details on PBX and connection settings see below.

Delete – sdelete a group, PBX or connection.

Auto login – enable an automatic PBX connection via the current connection after starting the configuration tool. One automatic connection may only be active at a time. By selecting another setting you cancel the previous one.

Cancel auto login – cancel the automatic connection without specifying any object.

In addition, the following options are available in the context menu:

Import PBX structure – import a predefined PBX structure as described below.

Export PBX structure – export the current PBX structure for later PC connection use.

Import database – import the database of a selected PBX in the off-line mode only. In the on-line mode, the database is replaced with the PBX data.

Export database – export the database of a selected PBX in the off-line mode only.

Show versions – display the PBX database version and the version downloaded in the off-line mode. The information is shown after the first login to the PBX.

Connection StructureIn this menu you can create groups and subgroups (nested groups) and, subsequently, add PBXs to them. You can create PBXs without groups too, but this might be confusing if you use a higher number of PBXs. Then you can create connections for particular PBXs using the TCP/IP and modem protocols. The records are arranged alphabetically.


For easier administration of existing records, a record moving option using the mouse has been implemented on this screen, also designated as drag & drop.

Figure: View of Possible PBX Connection StructureTo change the PBX connection settings use the icon.Properties

PBX PropertiesThe dialogue shown in figure below helps you create a PBX or change the properties of an existing PBX. The parameter meanings are as follows:

Name – define the name of the PBX to be connected.

Folder – define the path to the folder for the configuration to be saved.

Autosave – enable automatic database saving in the off-line mode.

Autosave database after – set the interval for automatic database backup. This function may only be used for the off-line mode.

Delete autosave item older than – set the maximum time for keeping old database backups in the storage folder. This function can only be used in the off-line mode.

Autosynchronisation – If this selection is active, the tool tries to synchronise the data automatically with the PBX without inquiring. This function can only be used in the off-line mode. If this selection is active, 'Loading from PBX' is performed if the off-line database is empty. 'Display changes' is performed if the database has been synchronised, the identifier matches and 'Database from Autosave' has not been selected. Otherwise, the synchronisation type dialogue is invoked. If this selection is inactive, the dialogue is invoked in all cases.


Figure: View of PBX Property Settings

Connection PropertiesThe dialogue shown in figure below helps you create a PBX connection or change the properties of an existing PBX connection. The parameter meanings are as follows:

Connection name – enter the name of the selected connection.

Modes – define whether the connection will support the on-line, off-line or both modes.

Download trace – disable or restrict trace downloading from the PBX. Particularly useful for modem connections. *Any of the following modes can be selected:

Only new – only the new trace is sent to the tool existent since the moment of PBX connection.

Never – no trace is sent to the tool (useful for trouble making modem connections).

Always – approximately a 300kB – 1MB trace is read out of the PBX buffer upon connection and sent to the tool. Useful for sending the trace after the PBX start up.

Connection type – define the PBX connection type. Typically, the TCP/IP or modem connection is selected. Use the TCP/IP, enter the PBX CPU IP address and the port to be used (6992 by default). With a modem choose the one that supports the X.75 protocol.


If unsuccessful try again – define the time interval between the PBX connection attempts in case of failure (PBX switch-off or restart).

Connect as – define the login and password data for a secured PBX access.

Figure: View of Connection Property Settings


1.

2.

3.

4.

5.

6.

Connecting to PBXAfter an automatic or manual initiation of the PBX connection, the dialogue shown in the figure below is displayed. It provides information on the PBX to be connected, the PBX firmware version (if detected), the last known connection error and, in the case of automatic connection attempts, also the remaining time to the next attempt. To connect immediately (before the timeout), push the button. The Try again Cancelbutton is used for leaving the dialogue.

Figure: View of Connection Course DialogueIf you are unable to connect to your PBX, please check whether:

the PBX has been switched on;

the PBX has been connected to the network;

both sides have the same IP address and port;

the used communication port has been opened;

the appropriate firmware and configuration tool versions are used;

the used communication port is not blocked by your antivirus software.



Main MenuAfter a successful connection to the PBX, the configuration part of the application is displayed. The main menu of this view is shown in figure below and contains the following options:

Administrator

Logout PBX – log out the configuration tool from the PBX and return to the previous menu for another connection as described above.

Connect/Disconnect – use these options off-line to connect/disconnect the configuration tool to/from the PBX.

Save changes – save all changes made since the last save.

Undo changes – cancel all changes made since the last save in a menu.

Settings – invoke a global setting dialogue as described in Chapter 1. About .Application

Language – choose one of the supported languages.

Exit – exit the configuration tool.

Trace

Load trace from file – load a trace from the file, thus clearing the previous one.

Add trace from file – load a trace from the file and add it to the existing one. You can interconnect traces for easy analysis.

Save trace to file – save the current on-line trace to a file. The configuration tool always saves an entire trace independently of whether the filter is being applied or not.

Start trace analyser – open the trace analysis window.

PBX

Upgrade – display a firmware upgrade dialogue. Having been chosen, the firmware file is uploaded into the PBX and unpacked. After a restart, the new firmware is used.

Import logs from PBX – get an easy access to the PBX logs without using other applications. You can import logs or selected logs only (All

).Selectively

All – after selecting a directory, the config.db and aoc.db files and the contents of the internal/log and /var/log directories are imported.

Selectively – select this item to display a dialogue for downloading selected logs from the PBX. The user can enter the files through a storage as defined in the menu. Global Data – Storage Manager

Restart PBX – use this option to initiate the PBX restart.

Restore factory settings – delete all the PBX settings. After selection, a


Restore factory settings – delete all the PBX settings. After selection, a dialogue window will appear with 2 options. Having selected one of them, confirm the selection by clicking OK twice.

Wizards

Activation wizard – refer to the next subsection, , for 1.3 PBX Activationdetails.

Import/export company structure – invoke the company structure import/export dialogue. The csv and xml files are supported.

Database import – open the database importing window. There are 2 options in the dialogue window: import database or . If From file From PBX

is selected, all the created PBXs will be displayed with available From PBX databases. The parameter specifies the parameters that must not be Ruleoverwritten by the import. This option is available for the off-line mode only.

Database export – export the PBX database to a file in the xml format. This option is available for the off-line mode only.

VoiceMail wizard – add automatically routers, DISA dial-in and progress tones for user VoiceMail.

Help – start the help menu in the chosen language.

View of Configuration Tool Main Menu

The figure above also includes all configuration menu icons. Icon meanings from the left:

Logout PBX – log out the configuration tool from the PBX and return to the previous menu for another connection as described above.

Connect/Disconnect – use these icons off-line to connect/disconnect the configuration tool to/from the PBX.

Save changes – save all changes made since the last save.

Undo changes – cancel all changes made since the last save in a menu.

Language – flags are used to mark the configuration tool language versions.

Eject to window – open the current tool indication in a separate window.

WindowsOn the left-hand side of the configuration tool you can find the TreeView where you can choose a menu item to be configured. The selected menu then opens on the right-hand side and is mostly divided into two subwindows; one for selecting and the other for configuring an object. The configuration is divided into tabs for easier orientation. All the above mentioned windows are shown in the figure below.


Figure: View of Configuration Tool Tabs and Windows

The and sections situated in the lower part of the screen above the Trace Databasestatus bar are important configuration tool components. helps you monitor calls Traceand analyse configuration errors if any. provides a direct view of the data Databasestored (depending on the connection mode). We strongly recommend that you should

This menu tab is governed not change the data in this view if you do not know how!by the read & write rights assigned to each login.

Status BarThe status bar is a lower task bar in the configuration tool. It provides two important pieces of information. The first one is the connection mode, which is either or on-line

. The other is the connected PBX name, which includes the and off-line PBX connectiondata separated with a dash.

Figure: View of Configuration Tool Status Bar

1.3 PBX Activation


1.3 PBX Activation

What You Need

To activate and configure you need the , a computer 2N® NetStar 2N® NetStar system

running the supported Windows version, a keyboard and a mouse. The PC and 2N®

PBX have to be interconnected in a LAN. Furthermore, it is necessary to NetStardisplay the redirected standard PBX input and output on your PC console. To do this, you need a six-wire, crossed cable with a six-pin RJ–12 connector on one end and a serial connector on the other. This cable is supplied as standard equipment. You can use a 'Tera Term Pro' console or any other functional console. You can also work in the HyperTerminal mode, but longer lines are interlaced.

Step 1: IP Address SettingPrior to starting configuration you need to set up the PBX IP address to establish network communication between the system and the computer. To do this, you can

use a serial console or the default IP address 192.168.100.100. The – PC 2N® NetStar

communication can be on-line or off-line. To configure the console use the table below.

Speed 115200

Bits 8

Parity None

Stop bits 1

Flow control None

If necessary, a serial port or modem may be used for PBX connection through the configuration tool, yet at a considerably lower bit rate.


Step 2: Hardware ActivationAfter the first connection to the selected PBX according to the Connection to PBX section, a configuration wizard is displayed as shown in figure below. This wizard is displayed only if the PBX has a new empty database (has not been preconfigured according to your demands).


Figure: View of Hardware Configuration Wizard DialogueWhen the wizard gets displayed, you can define the basic configuration settings of the virtual BRI ports. If you are not sure, you can push the button to proceed to the Nextnext configuration step because these settings may be changed any time later. Once

you do that, the configuration tool (together with the PBX) starts to detect the hardware as shown in figure below.


Figure: View of Wizard Hardware Detection OperationsThe boards are detected both in the basic unit and extenders. Once some hardware is detected, activation takes place, which means that virtual ports are assigned to all the boards detected (except for VoIP boards). The terminals connected are detected in the last stage of the wizard hardware configuration. This should make the PBX ready for further configuration, which is signalled by green board LEDs. The GSM board is

the only board without LED indicators and so its ready status is signalled by the port LEDs. The current firmware version is loaded into the PBX after the first start-up or every firmware upgrade and may cause a short delay in the GSM board activation.

Step 3: Localisation Setting

Localisation setting is another important step of the configuration. In this 2N® NetStar

step you can define the parameters shown in figure below and described in detail in the subsection. In addition, you can add a language package of your 6.3 Localisationown including texts and progress tones. Two language packages – Czech and English – are available in the PBX by default.


Figure: View of Wizard Localisation Setting Operations


Step 4: Time SettingIn this step, the wizard helps you set time, date and time zone. And also define the NTP server for automatic time synchronisation.

Figure: View of Configuration Wizard Time Setting

Step 5: PBX Function SelectionIn this step, choose one of the PBX modes. The setting is not final, it just defines the wizard's next configuration steps. The following options are available:

Private branch exchange

Virtual branch exchange

GSM gateway

Hotel


The offer of settings depends on the PBX mode selected. The lowest number of settings are available in the , where some steps are omitted and GSM gatewayconfiguration starts as late as the SMTP. The settings are identical for the other modes except for step 1, which is always adapted to the particular version. The following steps are available in the .Private branch exchange

Step 6: Creation of Groups, Users and ExtensionsIn this step, the configuration wizard enables an automatic creation of a group and its users and extensions. There are three types of extensions to be generated – analog, SIP and Cornet extensions. Analog extensions are used for the ASL virtual ports. SIP extensions are used for connecting SIP–supporting VoIP terminals and are assigned to the SIP proxy terminals. Cornet extensions are used for the StarPoint key (system) phones connected to the Cornet virtual ports. You can define the first extension number and count of extensions for each group (every other extension has a number increased by one). The extensions are then assigned to ports according to their types (if possible).

If you do not want to create extensions automatically, you can import the company structure from a pre-prepared file in the or format. In this way you can create xml csva relatively complex company structure including user logins and multiple user extensions.

Three functions are added to this section, the first two of which are used for re-launching of the wizard.

Add new and remove deleted – compare the existing structure of extensions with the selected file. New extensions are added and those which are present in the PBX yet undefined in the file are removed.

Add new only – compare the existing structure of extensions with the selected file and new extensions are added only. Undefined extensions are retained in the configuration.

Assign ports randomly – enable random assignment of extensions to ports.

If you neither want to automatically create extensions nor intend to import the company structure, you can click and select to proceed to Next Don't create anythingthe next step.


Figure: View of Wizard Extension Creation or Intra-Plant Structure Import

Step 7: Settings for AssistantThis configuration step includes just two functions with the following meanings:

Launch web server – launch the internal PBX web server, to which you can log in by entering the CPU IP address from your web browser.

Generate default logins – generate logins for the users created in the preceding step. With them, you can log in to the web server as a user.

Step 8: SIP Domain SettingThis step helps you define a specific SIP domain. If this option is not selected, the CPU IP address is used as the domain.

Step 9: SMTP SettingWithin this step you can define the SMTP server to be used by the PBX. Port 25 and the CPU Ethernet interface are set automatically for the SMTP. No security is used by default.


Step 10: Creation of RoutersThe wizard's last step is creating PBX routers. Routers are used for call/SMS routing from one PBX port to another. The wizard offers several default sets of routers, which are sufficient for your basic call routing. For special routing demands, reconfigure the routers and add new ones. All new routers are automatically filled with services, extensions and users and linked with each other.

Click Add routers for VoiceMail to create the routers and DISA objects for user VoiceMail.

Step 11: Data SavingChanges are not automatically stored into the PBX during the wizard process. To save the changes, click the button after completing the wizard. To cancel all new Savesettings, push the button.Undo


2. HardwareHere is what you can find in this chapter:

2.1 Basic

2.2 Boards

2.3 Synchronisation

2.4 Board and Port List

2.1 Basic

Service ModeThe section helps you put the PBX in the service mode and back if necessary. The service mode is used for quick changes such as card replacement. The PBX start is much faster after the service mode than after the PBX power off.

OFF – a normal PBX running status. To reuse the PBX while in the service mode, select and . Having returned from the service mode OFF save the changessuccessfully, you can see in the section in the column the Detected rack Statusstate .RUN

ON – the PBX service mode. To switch the PBX into the service mode while it is in a normal running status, select and . Having transferred ON save the changesinto the service mode successfully, you can see in the section in Detected rackthe column the state .Status STOP

Detected RackThe detected rack table gives you a clear overview of all parts of your PBX.

MAC address – MAC address of the detected rack.

Serial number – serial number of the detected CPU card.

State – current rack state. This may differ from the CPU card state (e.g. the CPU is ON while the rack if OFF in the service mode).

RUN – normal rack operation. The power supply is connected to the cards.

STOP – the rack is stopped. The power supply is disconnected from the cards. Typical for the service mode.

ERROR_LICENCE_EXPIRED – the rack is running, but the trial licence or time-limited main licence has expired. A new licence has to be requested.

1:Basic – indicates the basic unit state.

PRESENT – the basic unit is detected.

MISSING – the basic unit is not detected.

2–5:Extender – displays the extender state.

PRESENT – the extender is detected.


PRESENT – the extender is detected.

MISSING – the extender is not detected. Check the power supply connection and the switch card – extender CPU interconnection.

Hardware ProfileTo set the hardware profiles and improve your system efficiency use the HW – HW

menu. The menu contains ten different hardware profiles. For their benefits profilesand d i sadvantages see the tab le be low .

Each VoIP card allocates 32 channels in hardware profile 0 and 64 channels in all the other profiles. The PRI card always allocates 32 bus channels.

HW profile number – since different hardware profiles use different bus frequencies, the new configuration saving system is switched into the service mode and back automatically to bring the bus frequency changes into effect. The clock frequency of the bus can be 2, 4 or 8 MHz thus it corresponds to the amount of the channeles in extender: 32, 64 or 128. Amount of extender channels, used for calling, are always less by 4 because one channel is occupied for signaling.

Profiles 0 1 2 3 4 5 6 7 8 9

Extenders 4 0 4 4 4 0 4 4 4 0

Extender channels 128 0 32 64 128 0 32 64 128 0

Trunk positions 128 256

(164)

224 192 128 224

(132)

192 160 96 288

(192)

Main case case – digital 128 64 64 64 64 64 64 64 64 32

Main case – analog 32 32 32 32 32 64 64 64 64 32

Detectors 32 64 64 64 64 64 64 64 64 64

Players 32 64 64 64 64 64 64 64 64 64

Table: Benefits and Disadvantages of Hardware Profiles


Caution

The amount of channels for trunk positions in some hardware profiles depends on the type of the Switch card used. The default values are valid for the Switch card with one PRI interface or for the card with one PRI and 4 ports used for extender connection. If the Switch card with 4 PRI ports is used, the values in the brackets are valid.

2.2 Boards

HW ArrangementUnfolding the menu you can see the rack fitting as shown in the figure HW – Rackbelow.

Figure: View of PBX Basic Unit Panel

Do not remove or insert the board from the PBX without prior PBX switching off or into the service mode to avoid the PBX damage.

Use the list on the left-hand side of the PBX to switch between the basic unit and extender views. Click on the right-hand mouse button in the basic unit or extender view to display the following options:

Add board – click on an empty (no-board) PBX position to use this option. Add a board that has been detected by the PBX (using the Detected option) or a board from the list of supported boards for this position.


Remove board – remove a selected board. If virtual ports or resources have been assigned to the board, you will be asked whether they should be removed or retained.

Migrate virtual port/resource – use only if the context menu has been displayed on a port to initiate a virtual port substitution dialogue.

Synchronise with detected – synchronise the current unit or extender with the detected PBX boards. Before a board is removed from the current configuration, you are asked to confirm the removal.

Expert menu – access the advanced unit, board or port configuration functions. For details see later.

Note

Let us explain the terms "virtual port" and "physical port" and their difference for convenience. Basically, a virtual port is used for software setting of basic properties of a physical port. The advantage of this approach is the fact that the defined set of properties is attributed to a physical port only if the virtual port is assigned to it. Thus, you can esily move virtual ports between physical ports and change their functions as necessary

Expert menu – Virtual port

Assign virtual port – assign an existing virtual port to a physical port. Select a virtual port from the list of existing ports.

Create virtual port – use this option only for physical ports without any assigned virtual port. The new virtual port is automatically assigned to this physical port.

Remove virtual port – remove a virtual port from a selected physical port without deleting it. This virtual port can be used later including all settings (routing, assigned extensions, etc.). Rename the virtual port from XXX to UnassignedXXX.

Delete virtual porte – remove and delete a virtual port once forever. You will not be able to use this port any more.

Regenerate name – rename a selected virtual port according to its physical port.

Expert menu – Board and Case

Create virtual ports/resources – create virtual ports (resources) for all of those physical board or unit ports at once that have not been assigned a virtual port.

Remove virtual ports/resources – remove all virtual board or unit ports at once without deleting them. These ports can be used later including their settings. Rename these virtual ports from XXX to UnassignedXXX.

Delete virtual ports/resources – delete all virtual board or unit ports once forever. You will not be able to use this port any longer.

Regenerate unchanged names – change all unchanged names of the virtual


Regenerate unchanged names – change all unchanged names of the virtual board or unit ports according to their physical ports.

Regenerate all names – change the names of all virtual board or unit ports according to their physical ports.

BoardThe figure below shows all possible signalling statuses on the board ports.

Figure: View of Available Analog Board Signalling Statuses

Earphone

Cross

Green – signals a physical port with an assigned virtual port.

Yellow – signals a physical port with an assigned virtual port and active call (or call establishment).

Green – signals a physical port with an assigned virtual port and assigned extension.

Yellow – signals a physical port with an assigned virtual port, assigned extension and active call (or call establishment).

Exclamation mark

Yellow – signals a physical port without any assigned virtual port or physical port without detected status.

Red – signals a hardware error, e.g. a low signal level for a GSM, GSM port without SIM card, ISDN virtual port with deactivated L1 or L2 (adjustable), etc.


Tisk hardwarové konfigurace ústřednyK vytisknutí aktuální hardwarové konfigurace ústředny slouží tlačítka umístěná vpravo od obrázku základní jednotky nebo extenderu. Po kliknutí na jedno z tlačítek tisku se zobrazí okno s možností nastavení tisku. Volbou Tisk zobrazíme náhled, který lze pomocí tlačítka v levém horním rohu okna vytisknout. Tlačítka v pravém horním rohu slouží k zobrazení náhledu na připojené extendery.

– Vytiskne pohled zvolený pomocí tlačítek vlevo od Tisk aktuálního pohledudesky CPU (základní jednotku nebo extender).

– Vytiskne základní jednotku a extendery.Tisk všeho

BoardsA two-part is available under the PBX view. The upper part shows basic Boards taginformation on the selected board. The parameters mean the following:

Position – gives the board position number in the case as described below.

Type – gives the type of the board to be configured.

Enabled – disables the selected board. This option is useful, for example, while changing SIM cards without switching off the PBX.

State – provides the current board status including information on a mismatch of the board to be configured with the detected one.

Detected – shows the parameters of the board detected.

Type – type of the board detected on a selected position.

Serial number – serial number of the board detected on a selected position.

MAC address – MAC address of the board detected on a selected position.

A window showing the list of physical ports of the selected board is displayed under the above-mentioned part. The meaning of each list column is explained in the 2.4

chapter.Board and Port List

Tab Virtual portThe tag helps you configure your virtual ports easily. You can set all Virtual portparameters of the selected virtual ports and simultaneously see the panel layout. Use the tag for an easy configuration of virtual ports. The tag includes all Virtual port configuration settings for the selected virtual port while keeping the PBX view. Click on a card or its port to display the port assignment to a virtual port type in the left


section of the screen. Use the drag&drop function to move a virtual port to another type. If you select the CPU card, all the virtual ports that use the card's LAN interface are displayed. Thus, you will see the SIP Proxy and SIP Gateway virtual ports as well as the SMTP and SMTPD virtual ports.

For details on the setting options associated with the parameters in the right-hand section of the tag refer to the User Manual chapters dedicated to particular virtual ports (especially ).3. Virtual Ports


AddressingThe position of each board is specified in the format and the position of a R : C : Bport in the format. The characters have the following meanings:R : C : B : P

R – rack number;

C – rack unit number;

B – unit board number;

P – board port number.

Currently, takes up the value of 0 and ranges from 1 to 5, the basic unit being 1, R Cthe first extender 2 and so on up to the fourth extender with number 5. The board positions ( ) in the basic unit are numbered 1 to 14 from the left. The extender Bpositions are numbered similarly, from 1 to 12. The first basic unit and extender position is always reserved for the CPU board. The 1x/2x/4x ISDN PRI (with or without Zarlink) or Surf Ethernet boards can be mounted into positions 0:1:2 to 0:1:4 only. The position 0:1:5 is reserved for the Switch board, which contains the digital switching array.


2.3 SynchronisationUpon connection to a public or private ISDN network, remember to configure one port for synchronisation at least. The PBX works in two modes at the same time: as a source of synchronisation (Master) and a device that receives synchronisation (Slave). There are two fields in the menu. The left-hand one contains all HW – Synchronisationdigital virtual ports that can be selected for synchronisation, i.e. all PRI and BRI virtual ports in the TE mode. The other field contains a list of virtual ports that have been selected for synchronisation. All ports in the NT mode can be used as synchronisation sources.

Figure: View of Synchronisation Port Assigning and PrioritiesUse the and buttons to move the selected synchronisation up or down for a Up Down

higher or lower synchronisation priority respectively – the port listed first has the highest priority (255). Every other carrier has a priority lower by 1 (254, 253, ..). A

newly assigned carrier is always placed last (the lowest priority). In the case of synchronisation loss, the following port in the list (with a lower priority) is selected

automatically. Upon synchronisation restoration, the PBX returns automatically to the port with the highest priority.

Push the and buttons to transfer the virtual ports from one field to another Right Leftand thus ensure the PBX synchronisation.


2.4 Board and Port ListThe menu contains a list of boards that are physically present in Hardware – Board listthe PBX. The board list has four columns with the following meanings:

Address – shows the physical board address within the PBX according to the 2.2 chapter.Boards

Type – shows the board type.

Serial number – shows the factory-programmed board serial number.

MAC address – shows the board MAC address.

Module IMEI – shows IMEI of GSM module.

Virtual port – shows the complete name of the carrier or resource assigned to a physical port.

Stack – shows the general carrier / protocol stack (DSS1, ASL, CO, etc.).

Extension – shows the list of extensions assigned to a physical port carrier.

User – shows the users of extensions assigned to a physical port.

State – shows the current port state.

Description – provides associated information.

The context menu under the right button offers the following two options:

Export to CSV – export the whole table into a *.CSV file. You can use this export for stocktaking purposes and/or for contacting the 2N TELEKOMUNIKACE Technical Support if necessary.

Move to port – move quickly to the configuration of the virtual port on the selected row.


3. Virtual PortsHere is what you can find in this chapter:

3.1 BRI a PRI

3.2 Cornet

3.3 ASL

3.4 CO

3.5 GSM

3.6 SIP

3.7 SMTP

3.8 SMPP

3.9 Software and Dialler


3.1 BRI a PRI

BRIRefer to the subsection for the meaning of the virtual port.2.2 Boards

BRI virtual ports are assigned to physical ISDN ports for the Basic Rate Interface. For the hardware configuration of BRI virtual ports refer to the Virtual ports – BRI/PRImenu in the tab. A list of all BRI virtual ports is displayed on the left and a Stackwindow for the port parameter setting is available on the right. The configuration parameters are divided into logical parts.

Stack statusThis field displays information on the stack and its current status including information on the L1 or L2 states, higher error rates or loss of synchronisation.

Digital interface parameters

Interface type – cannot be selected, only shows the type of interface including bit rate information.

Interface mode – switch between the NT (Network Termination) and TE (Terminal Equipment) modes. Some basic unit positions can only be used in the NT mode. Specifically, they are basic unit board positions 6, 9 and 12 if you follow addressing as described in . A correct function requires a 2.2 Boardssoftware-hardware matching and a proper jumper setting for each board port. Figure below may serve as a guide.


Bus mode – switch between the MPT (point-to-multipoint) and PTP (point-to-point) modes. In the MPT mode you can connect up to eight terminals to one physical port. The PTP mode is mainly used for cross-connecting lines (trunks) between PBXs or for one terminal connection.

Enabled channels – activate the B–channels. If no channel is checked off, you cannot use this port for communication (it behaves as if busy).

Deactivate L1 at relax – deactivate the L1 layer on an inactive interface. The PBX automatically deactivates the layer after a timeout as defined in the Deactivation

item. Any incoming call automatically reactivates this layer.timeout

Keep L1 active – make the PBX keep L1 active on this interface without any incoming call. This option cannot be combined with the Deactivate L1 at relaxoption.

Inactive L1 as error – activate a caution about the first layer being inactive. This is indicated by a red exclamation mark on the port in the Hardware – Boardsmenu and a red text in the upper stack status field. This option may not combined with the option.Deactivate L1 at relax

Set SLIP – select to enable acceptable SLIP range Nonsynchronous as errorparameters. Use this option in the TE mode only. If the SLIP rate gets over the

level, a red exclamation mark appears on the port in the upper Hardware – menu and a red text is displayed in the upper stack status field. This Boards

error status gets changed after the SLIP rate falls below the level. The lowerinterval between these two values represents hysteresis.

Settings for BER – select to enable acceptable interface error BER as errorparameters. If the BER rate gets over the level, a red exclamation mark upperappears on the port in the menu and a red text is displayed Hardware – Boardsin the upper stack status field. This error status gets changed after the BER value falls below the level. The interval between these two values lowerrepresents hysteresis. The BER values are entered in an exponential format (e.g. 3e–5 means 3 errors in 100,000 bits). In practice, the BER may occur on a virtual port close after the cable connection. In that case, this is not an error if the BER disappears in a few minutes.

View of ISDN BRI Board Jumper Configuration (Thick Line Represents Board Front)

Specific interface parameters



Multiframe – is the first layer parameter of the So bus. For details refer to recommendation I.430.

Extended bus – activate an extended bus. With just one terminal and proper terminating impedance you can extend the PBX–terminal distance up to 1,000 metres. This parameter can be set for an NT port only.

Priority 10 – is the first layer parameter of the So bus. This parameter can be set for a TE port only.

DSS1 protocol parameters

Reverse NT/TE mode – this option refers to L3 signalling only. Check this option to make a TE port behave as an NT port (and vice versa).

Do not send time at NT – disable sending of the connection date and time information within the CONNECT message from an NT port to a TE port. Available for NT ports only.

Ignore unset explicit channel – enable call establishment without an explicitly set B–channel.

Always select B–channel – disable sending the channel identification information within the SETUP message together with channel signalling. Available for TE ports only.

Disconnect L2 if there is no call – disconnect the L2 layer on a inactive interface. The PBX automatically disconnects the layer after a timeout. An incoming call automatically reconnects the layer.

Keep L2 connected – make the PBX to keep L2 connected on this interface without any incoming call. This option may not be combined with the Disconnect

option.L2 if no call

Disconnected L2 as error – activate a caution about the second layer being disconnected. This fact is indicated by a red exclamation mark on the port in the

menu and a red text in the upper stack status field. This Hardware – Boardsoption may not be combined with the option.Disconnect L2 if no call

Outgoing numbering plan – this option allow to set up numbering plan type for called number within the SETUP message.

Terminals – is active for virtual MPT NT ports. Enter all connected ISDN terminals including their . Assign a extension to these terminals using the MSN numbers

tab. The terminal shall then identify itself as the selected extension Extensionswithin the PBX.

Caution

NetStar2N® allows only the calls encoded with to be G.711 A–law

processed on the ISDN interface (i.e. to be sent to interfaces other than ISDN). encoded incoming calls can only be sent between the G.711 µ–lawISDN interfaces via the PBX.


Digital interface diagnostic

Line state – the parameter cannot be set. It only shows the state of the first interface layer.

Number of SLIPs per minute – set the count of slips. A slip is caused by different clocks of the PBX and the active terminal. This value is updated every 6 seconds and represents a weighted average per minute.

Bit error rate per second – the BER parameter gives a count of incorrectly transferred bits during transmission. The value is updated every 6 seconds and represents a weighted average per minute.

If the port supervision is used, enable some or all of the following parameters: SNMP, , , Inactive L1 as error Disconnected L2 as error Nonsynchronous L1 as error BER as error

. If not, the port error will not be detected and the PBX will be unable to send a warning.

Expert tab

OUT parameters (PBX out)

IN parameters (PBX in)

PRIRefer to the subsection for the meaning of the virtual port.2.2 Boards

PRI virtual ports are assigned to physical ISDN ports for the Primary Rate Interface. For the hardware configuration of the PRI virtual ports refer to the Virtual ports – BRI

menu in the Stack tab. A list of all available PRI virtual ports is displayed on the /PRIleft and a window for the port parameter settings is on the right. The configuration parameters are divided into logical parts.

Stack statusThis field displays information on the stack and its current status including information on the L1 or L2 states, higher error rates or loss of synchronisation.


Interface type – the parameter cannot be configured. It only shows the type of interface including bit rate information.

Interface mode – switch between the NT (Network Termination) and TE (Terminal Equipment) modes. A correct function requires a software-hardware matching and a proper jumper setting for each ISDN board port. Figure below may serve as a guide.


Enabled channels – activate the B–channels. If no channel is checked off, you cannot use this port for communication or data transmission (it behaves as if busy). B–channels 0 and 16 cannot be used for call or data transmission under normal circumstances since they are blank. In PCMs of the 1st order are used for frame synchronisation and signalling transmission.

Deactivate L1 at relax – deactivate the L1 layer on an inactive interface. The PBX automatically deactivates the layer after a timeout as defined in the Deactivation timeout item. Any incoming call automatically reactivates this layer.

Keep L1 active – make the PBX keep L1 active on this interface without any incoming call. This option cannot be combined with the Deactivate L1 at relaxoption.

Inactive L1 as error – activate a caution about the first layer being inactive. This is indicated by a red exclamation mark on the port in the Hardware – Boardsmenu and a red text in the upper stack status field. This option may not combined with the option.Deactivate L1 at relax

Settings for SLIP – select to enable acceptable SLIP Nonsynchronous as errorrange parameters. Use this option in the TE mode only. If the SLIP rate gets over the level, a red exclamation mark appears on the port in the upper Hardware –

menu and a red text is displayed in the upper stack status field. This Boardserror status gets changed after the SLIP rate falls below the level. The lowerinterval between these two values represents hysteresis.

Settings for BER – select to enable acceptable BER range BER as errorparameters. If the BER rate gets over the level, a red exclamation mark upperappears on the port in the menu and a red text is displayed Hardware – Boardsin the upper stack status field. This error status gets changed after the BER rate value falls below the level. The interval between these two values lowerrepresents hysteresis. The BER values are entered in an exponential format (e.g. 3e–5 means 3 errors in 100,000 bits). In practice, the BER may occur on a virtual port close after the cable connection. In that case, this is not an error if the BER disappears in a few minutes.

View of ISDN PRI Board Jumper Configuration (Thick Line Represents Board Front)



Prefer CRC – enable preferring communication with the Cyclic Redundancy Check. In this mode, the PBX tries to establish connection with the CRC at first and, having failed, attempts to establish connection without the CRC.

Long haul – activate an extended bus called the Long Haul. With just one terminal and holding impedance, you can extend the distance up to 1,000 metres. This parameter can be set only on an NT port.

DSS1 protocol parameters

Reverse mode NT/TE – this option refers to L3 signalling only. Check this option to make a TE port behave as an NT port (and vice versa).

Do not send time at NT – disable sending of the connection date and time information within the CONNECT message from an NT port to a TE port. Available for NT ports only.

Ignore unset explicit channel – enable call establishment without an explicitly set B–channel.

Always select B–channel – disable sending the Channel identification information within the SETUP message with channel signalling. Available for TE ports only.

Disconnect L2 if no call – disconnect the L2 layer on an inactive interface. The PBX automatically disconnects the layer after a . An incoming call timeoutautomatically reconnects the layer.

Keep L2 connected – make the PBX keep the L2 layer connected on this interface without any incoming call. This option may not be combined with the

option.Disconnect L2 if no call

Disconnected L2 as error – activate a caution about the second layer being disconnected. This fact is indicated by a red exclamation mark on the port in the

menu and a red text in the upper stack status field. This Hardware – Boardsoption may not be combined with the option.Disconnect L2 if no call

Outgoing numbering plan – this option allow to set up numbering plan type for called number within the SETUP message.

Terminals – not applied for PRI ports.

Caution

NetStar2N® allows only the calls encoded with to be G.711 A–law

processed on the ISDN interface (i.e. to be sent to interfaces other than ISDN). encoded incoming calls can only be sent between the G.711 µ–lawISDN interfaces via the PBX.




Line state – the parameter cannot be set. It only shows the state of the first interface layer.

Number of SLIPs per minute – set the count of slips. A slip is caused by different clocks of the PBX and the active terminal. This value is updated every 6 seconds and represents a weighted average per minute.

Bit error rate per second – the BER parameter gives a count of incorrectly transferred bits during transmission. The value is updated every 6 seconds and represents a weighted average per minute.

If the port supervision is used, enable some or all of the following parameters: SNMP, , , Inactive L1 as error Disconnected L2 as error Nonsynchronous L1 as error BER as error

. If not, the port error will not be detected and the PBX will be unable to send a warning.

Expert tab

OUT parameters (PBX out)

IN parameters (PBX in)


3.2 CornetCornet is a digital virtual port for the StarPoint key phones with proprietary signalling (UPN interface). The Stack tab provides limited configuration capacities only. The parameters are divided into logical sections according to their respective functions. For the StarPoint configuration parameters refer to the Softphone subtab of the Properties tab.

Virtual port statusThe upper menu field displays information on the stack type and its current status including information on the L1 and L2 states, increased bit error rate or nonsynchronous L1.


Interface type – the parameter cannot be configured. It only shows the type of interface.

Interface mode – this parameter is set to NT and cannot be reconfigured. Thus, cross-connecting lines cannot be made using these ports.

Bus mode – this parameter is always set to PTP and cannot be reconfigured. This port is used for one terminal only.

Enabled channels – enable selected interface channels. If none is enabled, the interface cannot be used and behaves as if it was busy.

Keep L1 active – make the PBX keep the interface active automatically.

Inactive L1 as error – activate a caution about the first layer being inactive. This is indicated by a red exclamation mark on the port in the Hardware – Boardsmenu and a red text in the upper stack status field.

Set BER – select BER as error to enable acceptable BER range parameters. If the BER rate gets over the , a red exclamation mark appears on the BER error levelport in the menu and a red text is displayed in the port Hardware – Boardsstatus field. This error status gets changed after the BER value falls below the

. The interval between these two values represents hysteresis. The BER OK levelBER values are entered in an exponential format (e.g. 3e–5 means 3 errors in 100,000 bits).

Master terminal

Type – shows the type of the StarPoint terminal connected.

Firmware – shows the current firmware version of the terminal connected.

Extenders – shows information on the active extenders of the terminal connected.


Slave terminal

Type – shows the type of the StarPoint terminal connected.

Firmware – shows the current firmware version of the terminal connected.

Extenders – shows information on the active extenders of the terminal connected.


Line state – the parameter cannot be configured. It only shows the state of the first interface layer.

Number of SLIPs per minute – shows the count of SLIPs. A SLIP is caused by different clocks on the PBX and the terminal. This value is updated every 6 seconds and represents a weighted average per minute.

BER per second – the Bit Error Rate shows the count of incorrectly transferred bits on the interface during transmission. The value is updated every 6 seconds and represents a weighted average per minute.


Supported terminalsList of supperted terminals:

Optiset2N®Advance

Standard

Entry

StarPoint2N®Advance

Standard

Economy

Basic

Entry

OpenStage2N®10

15

20

30

40


3.3 ASLThe virtual port is used for connecting common analogue telephones or fax ASLmachines. This virtual port enables DTMF and pulse dialling detection and as well as DTMF or FSK using CLIP transmission. The parameters are divided into logical sections.

Stack statusThis field displays information on the stack and its current status. With an ASL virtual port you can see the following statuses:

null

config

on_hook

off_hook

error_stop

error_start_req

Figure: View of ASL Virtual Port Hardware Configuration


Line parameters

Impedance – this parameter determines the impedance of the hybrid circuit according to preset models (User, ETSI 600, Germany and Real 600).

Line model – this parameter provides further hybrid circuit parameters according to preset models EIA0 to EIA7 (e.g. EIA0 represents a 100m long line model).

Signalling type – shows the type of active state signalling. Choose from Reverse , or .polarity Tariff pulse Simple

Tariff pulse type – defines the tariff pulse sending source. Select , 12 kHz 16 kHzor .none

Incoming parameters (phone is dialling)

Call type – determine the preferred type of communication on this port. Choose one of the , , and options.Voice FAX A3.1kHz Audio 56kb Modem

DTMF dial enabled – make the carrier detect DTMF dialling from an analogue phone.

Pulse dial enabled – make the carrier detect pulse dialling from an analogue phone.

FLASH length [ms] – set the maximum time of the FLASH signal transmitted from a local phone to the PBX. The default value is 150 ms and the minimum value is 80 ms.

CPU DTMF detector enable – disable/enable DTMF detection using a detector on the board in order to save the internal PBX detectors.

Outgoing parameters (phone is ringing)

CLI broadcasting mode – define the preferred CLIP (Calling Line Identification Presentation) transmission type. The selections are , and .DTMF FSK none


3.4 COThe virtual port is an analogue virtual port for connection of a CO (central COexchange) analogue line. Since it has only a DTMF transmitter, it is unable to detect the DTMF. Therefore, route an incoming call directly to the final destination, or assign the DISA function to the virtual port to detect the DTMF symbols and route the call to the required destination. The parameters are divided into logical sections.

Basic

Stack statusThis field displays information on the stack and its current status. With a CO virtual port you can see the following statuses:

null

config

on_hook

off_hook

error_stop

error_start_req

Figure: View of CO Virtual Port Hardware Configuration


Line parameters

Impedance – determines the impedance of the hybrid coil according to preset models (User, ETSI 600, Germany and Real 600).

Line model – determines further hybrid coil parameters according to preset models EIA0 to EIA7 (e.g. EIA0 represents a 100m long line model).

Signalling type – shows the type of active state signalling. Choose one of the , or options.Reverse polarity Tariff pulse Simple

Tariff pulse type – defines the tariff pulse sending source. Choose , 12 kHz 16 kHzor .none

Dial tone

Congestion tone – select the congestion tone mask for testing purposes. The setting is available if the parametr is ticked off.Check Congestion tone

Check congestion tone – tick off this option to test the presence of the congestion tone. If the congestion tone is detected, the port passes to the On

state.hook

Outbound way parameters (from PBX)

Current detection timeout [ms] – set the time for current detection on the picked-up carrier. If no current is detected within this timeout, a failure is reported.

Dial tone timeout [ms] – set the waiting time for dialling numbers to the carrier. Select the Check dial tone option to test the dial tone presence during this time.

Dial to connect timeout [ms] – set the maximum delay time for CO line dialling. Time monitoring is renewed after every digit entered. If no digit is detected within this timeout, the connection will be regarded as terminated and a new connection attempt will be made.

Check dial tone – enable dial tone testing on the carrier for a period of time set in the parameter.Dial tone wait timeout

DTMF dial enabled – enable DTMF dialling via the port.

Pulse dial enabled – enable pulse dialling via the port.

FLASH length [ms] – set the maximum time of the FLASH transmitted from a local phone to the PBX. The default value is 150 ms and the minimum value is 80 ms.

Inbound way parameters (to PBX)

Ring pulse time [ms] – this parameter sets the minimum time of the ring signal presence needed for ring detection. If the ring time is shorter than the preset value, ringing will be ignored.

Ring pulse treshold [V] – this parameter sets the minimum ring voltage level needed for ring detection. If the ring voltage level is lower than the preset value, ringing will be ignored.

Ring pattern time [ms] – this parameter sets the minimum period of time for


Ring pattern time [ms] – this parameter sets the minimum period of time for alerting detection.

CLI reception mode – define the preferred CLIP (Calling Line Identification Presentation) reception type. Choose , or .DTMF FSK none

CLI reception timeout [ms] – set the CLI detection timeout as counted from the end of the first ring. This option is active only if the or reception DTMF FSKmode has been selected.

Polarity timeout [ms] – set the reverse polarity timeout. This option is active only if the item has been selected for this virtual port.Reverse polarity

Call reject timeout [ms] – if the PBX needs to reject an incoming CO call, it has to pick up and hang up. Use this parameter to define the timeout for this action. If the action is too short, the other party will not recognise termination.

Expert

Chipset

Chipset Type – this parameter determines the type of the chipset used. The SILABS_SI350 chipset is only supported at present.

Chipset Config – activate one of the chipset configurations created.

New config – create a new configuration for the selected chipset type.

Specific Chipset configuration

Name – set the chipset type whose configuration is being set in the section.

Caution

Do not change the above mentioned parameters unless absolutely necessary.

DCTerm – set the DC termination parameters (ringing voltage, minimum current, impedance). Hexadecimal values are used.

DAA Ctrl 5 – set further parameters for analogue line matching (on/off-hook rate, low pass filter). Hexadecimal values are used.

ACIM – set the proper impedance. Hexadecimal values are used.

Tx Gain – set the transmit gain.

Rx Gain – set the receive gain.

The table below includes the ACIM settings including meanings.


ACIM [3:0] Set AC termination

0000 00 600 Ohm

0001 01 900 Ohm

0010 02 270 Ohm + (750 Ohm || 150 nF) and 275 Ohm + (780 Ohm || 150 nF)

0011 03 220 Ohm + (820 Ohm || 120 nF) and 220 Ohm + (820 Ohm || 115 nF)

0100 04 370 Ohm + (620 Ohm || 310 nF)

0101 05 320 Ohm + (1050 Ohm || 230 nF)

0110 06 370 Ohm + (820 Ohm || 110 nF)

0111 07 275 Ohm + (780 Ohm || 150 nF)

1000 08 120 Ohm + (820 Ohm || 110 nF)

1001 09 350 Ohm + (1000 Ohm || 210 nF)

1010 0A 0 Ohm + (900 Ohm || 30 nF)

1011 0B 600 Ohm + 2.16 µF

1100 0C 900 Ohm + 1 µF

1101 0D 900 Ohm + 2.16 µF

1110 0E 600 Ohm + 1 µF

1111 0F Global complex impedance


3.5 GSMThe menu provides a list of all GSM virtual ports of the PBX. The Virtual ports – GSMparameters are divided into logical sections.

Basic

Stack statusThis field displays information on the stack and its current status.

Network selection

Net type selection – select the preferred network for module login. The following options are available:

Any

Only GSM

Prefer GSM

Only UMTS

Prefer UMTS

Roaming enabled – enable roaming for a GSM virtual port.

Manual network selection – if not checked, the SIM card tries to log into the preferred network automatically. If checked, enter the correct to Network code make the SIM card log into the selected network only. If the selected network is unreachable, the SIM card will not try to log into another network.

Network code – fill in a 5–digit international network code (e.g. T-mobile CZ=23001, O2 CZ=23002, Vodafone CZ=23003).

Network name – enter the name of the network as coded in the Network parameter.code

Cell selection – select the network cell to which the module shall/may log in.

Off – the cell is selected automatically.

Prefer selected – the module tries to log in to the cell specified in the parameter below. If unsuccessful, the module tries other available cells.

Only selected – the module only tries to log in to the cell specified in the parameter below.

Cell ID – set the network cell identifier for module login.

The following options are available under the right mouse button:

Known networks – open a dialogue with the list of known networks and their international GSM codes. The networks are arranged according to countries.


Visible networks – open a dialogue with the list of visible networks in the surroundings. By initiating the search you make your SIM card log out temporarily from the module.

Signal diagnostics

Signal measuring – enable signal level measuring for a selected carrier.

Signal monitoring – enable signal level monitoring for a selected carrier. If the signal level drops below the value specified in the parameter, a Poor signal levelred exclamation mark appears on the port in the menu and a Hardware – Boardsred text is displayed in the upper stack status field. This poor signal status gets changed after the signal level exceeds the value specified in the Good signal levelparameter. The interval between the values represents hysteresis.

Figure: View of GSM Virtual Port Hardware Configuration

GSM interface parameters


GSM interface parameters

CLI mode – enable CLI restriction for the active SIM card. The following options are available:

By SIM – the SIM card default setting is respected.

By calling number – the calling user CLI setting is respected. If CLI is allowed, the SIM card uses CLI too. If not, the SIM card's identification is restricted.

Presented – SIM CLI is always presented regardless of the SIM card or calling user settings.

Restricted – SIM CLI is always restricted regardless of the SIM card or calling user settings.

Caution

This function must be supported by the network provider. Otherwise, calls with suppressed identification are rejected with a corresponding cause.

Relax interval between calls – determine the idle period between two calls. This parameter only applies to calls going out from the PBX through the GSM carrier, not to incoming calls. During this time all outgoing calls are rejected with cause 34 – no circuit/channel available.

DID transmission in number – enable a special direct dial-in transmission function within the called number. This function is supported by some networks only.

DID separator – set a character to separate the called SIM card number and the DID (direct dial-in).

DID mode – define how to work with the DID. Select one of the following options:

By calling number – if the calling user has CLIR, the calling number is not displayed in the DID (777982494#). If not, the calling number is displayed behind the DID separator (777982494#274).

Always presented – CLI is always presented in the DID regardless of the CLI setting (777982494#274).

Restricted DID replace – the mode is similar to the By calling number mode; the only difference being that, in the case of CLIR enable, the identifier specified in the DID replace pattern parameter is displayed behind the DID separator instead of the CLI (e.g. 777982494#888).

DID replace pattern – specify the DID to be used as the caller's identification in the case of CLIR.

DID off for emergency calls – disable the use of DID for specified emergency calls while the PBX is in one of the modes as described in detail in the .3.5

menu.Přenašeč GSM v4.4

CPU DTMF detector enable – disable/enable DTMF detection using a board detector in order to save internal PBX detectors.


Send congestion tone – if this item is selected and the port requires tones, the PBX generates the congestion tone to the other party after the call end until the port receives the message or the expires. If this item is not Release 30s timeoutselected, the channel is closed practically on the telephone hang-up. If the port requires no tone, the channel is closed immediately too.

Transmit DTMF signaling via GSM – enable sending of DTMF signalling via GSM instead of a voice channel for better detection by the counterparty. The function

.is available for the GSM card with MC55 modules only – Part No. 1011708E

DTMF tone duration [ms] – define the DTMF tone duration.

Validity of SMS [s] – set validity of the SMS to be sent to the provider's network. Always round the value up. The steps are as follows: 5 minutes for up to 720 minut (12 hours), 30 minutes for 12 to 24 hours, 1 day for 1 to 30 days and 1 week for 4 to 63 weeks.

GSM modul diagnostic

Producer – provides information on the board manufacturer.

Type – provides information on the board type.

Firmware revision – the software revision of the firmware uploaded into the board.

Module IMEI – shows the detected IMEI code.

GSM network diagnostic

State – shows the current port state for detection of network login problems if any. For example, means that the SIM card requires the PIN PIN REQUESTEDcode to log in. To log in successfully, you either enter the PIN or disable PIN requesting by the SIM card.

Net type – shows the network type to which the module is logged at the moment.

Logged network – shows the international code of the network to which the SIM card is logged at the moment.

Network name – shows the name of the network into which the SIM card is logged at the moment.

Area code – displays the code of the area to which the SIM card is logged at the moment.

Cell ID – identifies the cell to which the SIM card is logged at the moment.

Cell selection state – informs whether the given module supports manual cell selection.

Signal – shows the current signal level (if activated). A low signal level may result in logout or call failure due to a high error rate.

SIM number – shows the SIM card code detected.

SMS centre number – fill in this parameter to enable SMS sending. In GSM networks, SMSs are not routed directly to the final destination, but through the provider's SMS centre. This is useful where an SMS cannot be delivered

immediately (e.g. due to target phone unavailability). The SMS centre tries to


immediately (e.g. due to target phone unavailability). The SMS centre tries to deliver this message cyclically for a preset SMS validity time. This parameter is mostly automatically detected on the SIM card (preset by the provider). If not, fill it in manually.

PIN – enter the PIN code if it is required by the SIM card and has not been entered in the menu for this SIM card.SIM – SIM cards

PUK – enter the PUK code if it is required by the SIM card and has not been entered in the menu for this SIM card.SIM – SIM cards

Phone number – this field is for information only. You can enter your SIM card telephone number for easier orientation. This parameter has no function.

Expert

AT commandsYou can add AT commands here to set the module properties. These AT commands are executed upon every PBX restart or GSM/UMTS card restart. Use the arrows in the right-hand part of the section to specify the sequence of the commands. The Timeoutcolumn sets the time during which the answer to the command entered is awaited. The columns includes a brief statement on whether or not the command was Resultsuccessful. For specific answers see the section.Answers for selected

Answers for selectedHere find an answer to the AT command selected in the left part of the screen.

Net code locksUse a special licence to restrict a module to a specific network. This restriction is

Use a special licence again to unlock the status. The lock supports up to 8 permanent.networks.

Lock mode:

Unused – the lock is not activated.

First enable – the module is locked for the first network to which it logs in.

Enable – the module may log in to the selected networks only.

Disable – the module may log in to all networks except for those selected in the section.Net code


SIM num locksUse a special licence to restrict a module to a specific SIM card. This restriction is

Use a special licence again to unlock the status. The lock supports up to 4 permanent.SIM cards.

Lock mode:

Unused – the lock is not activated.

First enable – the module is locked for the first SIM card inserted.

Enable – the selected SIM cards may log in to the module only.

Disable – all SIM cards may log in to the module except for those selected in the section.SIM num

Audio parametersHere set the audio profile for various Siemens MC55/MC55i GSM module versions.

MC55 Audio Profile – select the module version.

Tx gain – set the gain of the audio signal to be transmitted – not implemented yet.

Rx gain – set the gain of the audio signal to be received – not implemented yet.

USSD

USSD commandsThis section helps you enter the USSD commands (codes) for prepaid SIM card recharging or credit info obtaining, for example. Click to enter the required Newcommand and to repeat the last-entered command. Click to abort the Repeat Cancelcurrently executed command. View the information on the USSD command result in the window.Reply

Network name – display the name of the network to which the SIM card is logged in at the moment.

Command – display the last-entered USSD command.

State – provides information on the command processing.


3.6 SIP

SIP GatewayThe SIP Gateway virtual port is used for creating a trunk between two PBXs or connecting a PBX to the public network via a VoIP provider.

Stack statusThis field displays information on the stack and its current status.

SOCK_TCP_ERROR – the TCP socket has not been opened.

SOCK_UDP_ERROR – the UDP socket has not been opened.

CREDS_IN_ERROR – the authorisation server is unavailable.

CREDS_OUT_ERROR – the authorisation client is unavailable.

REALM_CONFLICT – the Realm collides with another port's Realm/Alias.

STUNNING – the public IP address is being obtained from the STUN server.

STUN_TIMEOUT – the STUN server is unavailable.

EXPIRED – the public IP address validity has expired.

SIP_REGISTERING – the gateway registration is in progress.

REG_TIMEOUT – the REGISTRAR server is unavailable.

REG_NOT_AUTH – the registration has not been authorised.

REG_REJECTED – the registration has been rejected with an error.

SIP server statusThis section helps identify the current connection state via the OPTIONS messages sent in a preset interval. Refer to the menu on the tab for settings. The Advanced Misc following states are available:

Unmonitored – OPTIONS sending is off.

Ready – the counterpart replied to the OPTIONS sent, connection has been established.

Not responding – no reply to the OPTIONS sent has been received.

Unknown

Local settings

Listening port – define the local gateway port to communicate with the other party.

Realm (Domain) – define the domain over which the gateway communicates. The domain and ports specified here help route calls to the gateway. The Realm(Domain) + port items are checked in the Request-URI field for incoming INVITE messages. If the setting matches the gateway SIP, the packets are routed to the


messages. If the setting matches the gateway SIP, the packets are routed to the gateway. The INVITE messages whose Request-URI items are included in the Alias field are served too.

Via/Contact – define the contents of the and headers. The following Via Contactoptions are available:

IP address – fill in the CPU IP address.

FQDN – fill in the PBX DNS HostName as entered for the PBX IP interface (e.g. on CPU board in menu – ). Hardware Boards

NAT – fill in the public IP address and NAT port for the opponent's sending of signalling messages. Packets are routed to the PBX according to the port routing and router IP address settings.

STUN – enter the STUN server address and port for finding the current address behind the NAT.

Authorisation required – enable the other party's authorisation for incoming calls. User login data are used for this purpose. All logins are always used.

Figure: View of SIP Gateway Configuration Menu

Remote SIP server parameters

Connect to – define the opponent's (provider's or other PBX's) IP address or DNS for trunk connection (call routing and registration request sending). If a port other than 5060 is to be used, it should be specified behind a colon (192.168.122.43:5071).

Protocol – specify whether to use UDP and/or TCP, or just one of these protocols for transmission. If the NAPTR (Name Authority PoinTeR) option is selected, a query to the DNS is made first and the transmission protocol is selected depending on the reply. The parameter (menu , Use DNS SRV Advanced

tab) can only be used with this setting and a suitable DNS.Miscellaneous

Use DNS SRV – if this option is selected, a query to the defined DNS is made


Use DNS SRV – if this option is selected, a query to the defined DNS is made before the INVITE message is sent. The DNS reply defines two different call routing addresses. INVITE is routed to the first address and, if no reply comes after three INVITE sending attempts, the PBX sends INVITE to the other address included in the DNS reply.

Register line – enable line registration and specify the Caller ID. If a line is not registered, no call establishing requests are sent to it.

Expiry – define the registration expiry. The final value may be defined by the other party (e.g. shorter).

Username – enter the username for login with authorisation.

Password – enter the password for login with authorisation.

Trustful IP addressesThe parameter helps you secure your PBX system against unauthorised call setup attempts via the given SIP gateway. Tick off this option to make your PBX process requests from trustworthy IP addresses only. Click on the buttons to the right of the IP address list or open the context menu in the IP address list using the right mouse button to add, remove or modify an IP address.

RTP interface

Name – shows the name of the Ethernet interface (VoIP card) used.

UDP min – define the lower limit for the UDP ports used for RTP stream sending.

UDP max – define the upper limit for the UDP ports used for RTP stream sending.

NAT – enable RTP stream routing through the NAT. If this selection is No, the opponent's RTP stream is sent to the VoIP interface. If a PBX is configured behind the NAT, one of the options in this menu has to be used for the VoIP interface to send a correct IP address to the WAN.

NAT source – if you have entered the fixed IP address in the NAT column, now fill in the NAT IP address here for RTP streaming.

NAT base – if you have entered the fixed IP address in the NAT column, now fill in the NAT port here for RTP streaming.


Codecs

Supported – here find a list of supported codecs excluding the codecs that have been selected as Allowed.

Allowed – here find a list of codecs to be used for communication on this virtual port. The context menu under the right-hand mouse button provides further

options.Codec setting

DTMF according to RFC2833 – enable DTMF transmission according to RFC2833. Upon selection, set the Payload type for DTMF transmission via a link below the name.

Fax T.38 – enable fax transmission according to the T.38 recommendation. If checked off, a link becomes available to the Advanced settings. The recommended setting is – , – and TCF Transfer Error correction Redundancy No

.compression

FAX detection – set whether shall detect fax (send re-INVITE with T.2N® NetStar

38 in SDP) for incoming or outoging fax messages only, always or never.

Figure: View of Codecs Setting Menu

Caution

All the videocodecs and audio codec G.722 are negotiated by signalling

only. does not mediate or transcode the RTP packets that 2N® NetStar

contain data encoded by any of the above mentioned codecs.


SIP ProxyThe SIP Proxy virtual port is used for connecting SIP terminals to the PBX through terminal registration. All the parameters are divided into logical sections.

BasicStack status

This field displays information on the stack and its current status.

SOCK_TCP_ERROR – the TCP socket has not been opened.

SOCK_UDP_ERROR – the UDP socket has not been opened.

CREDS_IN_ERROR – the authorisation server is unavailable.

CREDS_OUT_ERROR – the authorisation client is unavailable.

REALM_CONFLICT – the Realm collides with another port's Realm/Alias.

STUNNING – the public IP address is being obtained from the STUN server.

STUN_TIMEOUT – the STUN server is unavailable.

EXPIRED – the public IP address validity has expired.

SIP_REGISTERING – the gateway registration is in progress.

REG_TIMEOUT – the REGISTRAR server is unavailable.

REG_NOT_AUTH – the registration has not been authorised.

REG_REJECTED – the registration has been rejected with an error.

Local settings

Listening port – here fill in PBX port for the SIP Proxy – terminal communication.

Realm (Domain) – define the domain over which the gateway communicates. The domain and ports specified here help route calls to the gateway. The Realm(Domain) + port items are checked in the Request–URI field for incoming INVITE messages. If the setting matches the SIP Gateway setting, the packets are routed to the gateway. The INVITE messages whose Request–URI items are included in the Alias field are served too.

Via/Contact – define the contents of the Via and Contact headers. The following options are available:

IP address – fill in the PBX IP address.

FQD – fill in the PBX DNS HostName as entered for the PBX IP interface N(e.g. on CPU board in menu – ). Hardware Boards

NAT – fill in the public IP address and NAT port for the opponent's sending of signalling messages. Packets are routed to the PBX according to the port routing and router IP address settings.

STUN – enter the STUN server address and port for finding the current address behind the NAT.


Authorisation required – enable authorisation for all terminals. Logins and passwords of the users whose extensions are assigned to the given terminal on the tab are used for registration.Extensions

Proxy parameters

Registration validity – use this parameter to define the validity for terminal registrations. Every terminal has to send a new registration request upon expiry. The parameter range is 30 to 3,600s. The resultant registration term may be shorter than the value defined here (depending on the terminal setting).

RTP interface

Name – shows the name of the Ethernet interface used.

UDP min – define the lower limit for the UDP ports used for RTP stream sending.

UDP max – define the higher limit for the UDP ports used for RTP stream sending.

NAT – enable RTP stream routing through the NAT. If this selection is No, the opponent's RTP stream is sent to the VoIP interface. If a PBX is configured behind the NAT, one of the options in this menu has to be used for the VoIP interface to send a correct IP address to the WAN.

NAT source – if you have entered the fixed IP address in the NAT column, now fill in the NAT IP address here for RTP streaming.

NAT base – if you have entered the fixed IP address in the NAT column, now fill in the NAT port here for RTP streaming.

Codecs

Supported – here find a list of supported codecs excluding the codecs that have been selected as Allowed.

Allowed – here find a list of codecs to be used for communication on this virtual port. The context menu under the right–hand mouse button provides further

options.Codec setting

DTMF according to RFC2833 – enable DTMF transmission according to RFC2833. Upon selection, set the Payload type for DTMF transmission via a link below the name .

Fax T.38 – enable fax transmission according to the T.38 recommendation. If checked off, a link becomes available to the Advanced settings. The recommended setting is – , – and TCF Transfer Error correction Redundancy No

.compression

FAX detection – set whether shall detect fax (send re–INVITE with T.2N® NetStar

38 in SDP) for incoming or outoging fax messages only, always or never .


Caution

All the videocodecs and audio codec G.722 are negotiated by signalling

only. does not mediate or transcode the RTP packets that 2N® NetStar

contain data encoded by any of the above mentioned codecs.

TerminalsThis section is used for terminal management. If no terminal has been created, the VoIP phone cannot register to the SIP proxy. A registered phone is indicated by displaying the IP and MAC address for the connected terminal. Multiple phones may be registered to one terminal. In the case of an outgoing call, all of the registered phones are alerted until one of them answers the call. Incoming calls are identified according to the extensions that are assigned to the terminals on the tab.Extensions

Figure: View of SIP Proxy Configuration

The following options are available under the right mouse button:

Add – display a terminal adding dialogue. Select the , and Name Type MAC for the terminal.address

Edit – edit the existing records.

Remove – remove the selected record from the configuration.

Transfer MAC address to terminal – transfer the MAC address of a registered terminal device into the terminal configuration.

The handset icon colour signals the terminal state: green means relax, red indicates an active call, terminal ringing or outgoing call ringing and blue signals an unknown terminal state or a current state change.


Caution

While using the function for supported IP phones and SIP Provisioningitercoms, make sure that the terminal type and MAC address are completed correctly.

ProvisioningUse the parameters included in this tab for configuration file creation (SIP

provisioning) for some 2N StarPoint IP, Well or Yealink phones. Refer to Subs. 9.1, ® Softphone, for the list of supported IP phones.

Backup proxy – backup proxy address.

Port – backup proxy port.

Primary and secondary NTP server – server addresses for time synchronisation.

Password – web interface login password.

Advanced Settings

Figure: View of Advanced VoIP Parameter Setting Dialogue


SIP

Always mediate RTP – if this option is checked off, the RTP stream is always routed through the PBX VoIP card. If not, the RTP stream is processed outside the PBX (in the case of VoIP – VoIP connection) and the PBX is reponsible for signalling only.

Reverse RTP negotiation – define the way of codec negotiation. If this option is not checked, codecs are offered already in the message.Invite

Use short headers – enable using short headers (e.g. From = f, To = t, Via = v) for data transfer minimisation.

Do not replace +,#,* in numbers – these characters are replaced with corresponding strings %xx in numbers only if the option is not selected. If it is selected, they are sent.

Route by To header – if this option is selected, the incoming calls of the given port are routed by the header. In other cases (including the default setting!), Tothe calls are routed according to the header.Request URI

BLF by station – activate station state monitoring (according to station numbers). The user state is monitored by default (according to user numbers).

Register sender's address – used for displaying of the IP address and NAT port from which registration came of the terminal that is behind the NAT and registering with the PBX Proxy.

Send congestion tone – enable transmission of the congestion tone from the PBX or network in case the opposite subscriber hangs up.

Preserve calling name – enter the original calling name into the 'Header name' for an outgoing call via the selected SIP port to replace the phone directory name or the name entered in the complex router, for example.

Support WebSocket SIP – enable the WebSockets technology for SIP-based client communication.

Ignore History-Info and Referred-By – disable further processing of the History-Info and Referred-By headers if detected in the incoming SIP signalling.

Dedicated Registrar – enable routing registrations from gateways to another destination.

Address – the selected Registrar server address.

Scheme – set the or scheme in the To and From SIP headers. is used sip tel telfor the networks that apply the numbering plan according to the E.164 recommendation.

Caller-ID – specify whether or shall be used for caller's From P-Asserted-Identityidentification.

Min. MTU – set the packet size limit for obligatory TCP use in the UDP&TCP mode. The recommended maximum value is 1,448 bytes.


Headers

Complete domains – specify the domain to be used for the or (P-From, To PAI Asserted-Identity) headers.

Send information

P-Asserted-Identity – activate the P-Asserted-Identity header in the INVITE message for CLIR transmission. The opponent can thus obtain CLI information even if its CLIR (Calling Line Identification Restriction) is enabled. By default, CLIR is enabled on the SIP Gateway port (header active) and disabled on the SIP Proxy port (header inactive).

P-Preferred-Identity – activate the P-Preferred-Identity header in INVITE messages. This header is used for transmission of the number (ID) of the user to be redirected. Also, enable the or Redirecting number Facilityparameter on the – tab.Properties Customer

Complete users – modify the ID to be sent in the header.P-Asserted-Identity

AliasesThis section enables you to define further Realms (Domains) to be accepted. Those incoming calls (their INVITEs) will be accepted to the port whose Request-URI field shall match the SIP gateway or SIP proxy Domain and Alias settings.

RTP

DSP – use this section for transferred data optimisation. When enabled, packets are not sent uselessly when the user does not speak. VAD stands for Voice Activity Detector.

VAD off

VAD according to G.729 Annex B

VAD light

Generate comfort-noise – generate some noise into the call. Since analogue line users are used to some background noise, similar noise is simulated here for their comfort.

Mask lost packets – activate lost packet masking optimisation.

RTCP – Nastavuje interval odesílání RTCP paketů.


QoSThe section helps you set the packet outgoing parameters to determine TOS/DiffServthe packet processing priority in network elements.

SIP – set the hexadecimal value of the SIP packet priority.

RTP – set the hexadecimal value of the RTP packet priority.

Default values – reset the default values for both the parameters. The default settings are optimum for voice transmission.

Echo suppressionUse this tab to enable variable echo suppression modes.

Suppression off

Profile G.168 8 ms

Profile G.168 16 ms

Profile G.168 32 ms

Profile G.168 64 ms

Profile G.168 128 ms

Delay [ms]

Adaptive suppression

Non-linear processing

Re-use coefficients

Automatic control

Jitter bufferUse this tab to enable variable jitter suppression modes.

Type – select one of the following jitter buffer types:

Fixed – the jitter buffer settings are fixed; the set delay value is constant.

Adaptive – the jitter buffer settings may vary depending on the network conditions and variable packet delays.

Short-run adaptive – the jitter buffer settings are fixed; the set delay value is constant . If, however, packets are discarded or the delays exceed the upper/lower limit, the jitter buffer will adapt to the current network conditions.

Lower limit [ms]

Upper limit [ms]

Threshold


Non-managed network – jitter buffer settings for networks with constantly varying transmission conditions. The settings are based on the Adaptivejitter buffer, but the setting changes are not made in steps.

Delay [ms]

Depth [ms]

MiscellaneousIn-call mark receiving

Mode – set the supported DTMF receiving method for calls.

Generating of INFO message

DTMF – select one of the two DTMF transmission modes via SIP INFO. The modes use different formats for the DTMF transmitting message.

KeepAlive

Interval – defines the KeepAlive packet sending interval. The default value is 10 s.

STUN server

The STUN server allows the NAT clients (i.e. computers behind the firewall) to set up calls with a VoIP provider outside the local network.

Host – enter the STUN server address (IP or domain name). The address is used in case the STUN IP method is selected in the RTP interface configuration. By default, is selected.stunserver.org

Port – define the port to be used for the STUN server. The default value is port .3478

Others

Use DNS SRV – if this option is selected, a query to the defined DNS is made before the INVITE message is sent. The DNS reply defines two different call routing addresses. INVITE is routed to the first address and, if no reply comes after three INVITE sending attempts, the PBX sends INVITE to the other address included in the DNS reply.

Outbound gateway – if this option is selected, the and Contact Record routeheaders are ignored in replies and packets are routed directly to the address specified in the field.Address

http://stunserver.org


3.7 SMTP

SMTPThe is an Internet protocol used for e-mail Simple Mail Transfer Protocol (SMTP)transmissions across the Internet between clients and the server. The protocol provides deliveries via a direct sender-receiver connection.

E-mail messages can be sent via the SMTP client and received via the SMTP server in the NetStar PBX. See below for details. E-mail messages are routed by the PBX in the same way as SMS messages, i.e. using such objects as text routers and the SMS routingtab.

SMTP Clients

2N® NetStar provides more than one Ethernet interface. For communication with the

SMTP server, however, the PBX always uses the CPU Ethernet interface. In the Virtual menu you can create SMTP clients to log into the SMTP servers and send ports – SMTP

e-mail messages.

Figure: View of SMTP Client Configuration

Network interface – choose the network interface to be used for SMTP communication with the server. In this version you can use the CPU network interface only.

Outgoing mail server – enter the IP address of the SMTP server. If you use the DNS server, you can also use the domain name of your SMTP server.

Port – define the port to be used for communication with the SMTP server. Typically, port 25 is used.

E-mail address – is used for identification of incoming messages within the SMTP server. Without a correct setup, the SMTP server will probably reject all connection establishment requests!


Authentication – choose the type of authentication for access to your e-mail account on the selected SMTP server.

None

Plain

Login

Digest_MD5

Cram_MD5

Account name – provide the name of the e-mail account registered by the selected SMTP server. Required by all the above mentioned methods.

Password – set the account access password required by all the above mentioned methods.

SMTP Server (SMTPD)The server processes incoming e-mail messages.SMTP

Port – set the port on which incoming e-mails for this SMTP server are awaited. Two PBX SMTP servers may not have one and the same port.

Queue length – set the count of e-mail messages to be queued and subsequently processed by the server (routed to final destinations in the PBX or resent to another interface). If you set 1, the server will not receive an e-mail processing request until it completes the preceding one.

Authorised group – authorise incoming e-mails. The following options are available:

Without authorisation – incoming e-mails are accepted without authorisation.

Any – the e-mails matching any PBX user login are accepted.

Group of users – the e-mails matching user logins from a certain user group are accepted. If a superior group is selected, all the subgroup users are included too.

Figure: View of SMTP Server Configuration


SMPPThe is a network protocol designed for SMS Short Message Peer-to-peer Protocoltransmission. It is primarily used for bulk transmission between the provider's SMS centre and the client (called ESME – External Short Message Entity).

Stack statusThe upper menu field displays information on the stack (communication protocol) type, connection state and current port status.

Connection state

Idle

Error

Connecting...

Logging...

Logged in

Login failed

Last state

O.K.

Disconnected

SMPP error by login (Invalid response)

SMPPNACK error

Response timeout

Invalid password

Invalid username

Figure: View of SMPP Configuration


Connection type – select one of the following modes:

Transciever – the port receives and sends SMS messages.

Receiver – the port is connected to the server waiting for SMS messages. The attempt to send an SMS to the server is rejected by the PBX.

Transmitter – the port only sends SMS messages to the server. The connection state is and login occurs when an SMS is to be sent.Idle

Server address – set the IP address of the server (SMSC) to which the client shall connect. You can use the server domain name too for a DNS server.

Port – set the number of the port to be used for server communication.

User name – set the user name for server communication authentication. The maximum length is 15 characters (limited by the SMPP standard).

Password – set the user password. The maximum length is 8 characters (limited by the SMPP standard).

Address range – specify the range of addresses of potential addressees, used optionally by the server for SMS sending.

System type – specify the type of client if the server connection requires so (optional parameter).


3.9 Software and DiallerThe port and are objects used for easy verification of the PBX Software Diallerconfiguration, especially the routing rules for calls and SMS.

When a software port is added, a Dialler (software terminal) is created automatically. As its functions are based on a key phone, its controls and menus are almost identical with the key phone. Similarly, you can assign a station to the software port on the

tab and make outgoing and incoming calls like with a standard key phone.Stations

Caution

The calls between the Dialler and any other PBX port are only connected on the signalling level: as the speech channel is not allocated, the speech signal cannot be transmitted.

Caution

If you, during an active call, pass into another menu (except for Trace), the call will be terminated. Use the function on the Eject to windowupper toolbar to keep the active call.

Figure: Software Port Options

Port type

Admin

No login

Display width – define the Dialler display width in characters.

Display height – define the Dialler display height in rows.

Keys count – define the count of programmable Dialler keys. Having assigned a station to the Dialler, configure the keys in the fall-down menu on the Properties

tab.Softphone


Figure: Dialler

The Dialler figure can be divided into the following three sections:

Display – display the following in the default state (upon station login): station number, current profile, date and time.

Control keys

represents the Dialler handset. Press the handset to start a call with the previously dialled number, or receive a call.

is the HandsFree key, which has the same function as the handset key in the Dialler.

helps you hold an active call or return to a call on hold.

and helps you move across the Dialler menu.

is used for confirmation. Press this key to select a function, receive a call or send the dialled number.

is used for return from a menu or call rejection.

Phone keys – set the phone key count on the tab of the respective Stack Software port. Set the phone key function in the fall-down menu on Propertiesthe tab.Softphone



IntroductionThe menu helps you configure all virtual port types and virtual ports. In Virtual portsthe menu you can see all virtual ports regardless of their type. For Virtual ports – Alleasier orientation, the virtual ports are arranged according to port types and also colour-distinguished according to the stack type. To display a selected virtual port type use the submenus. By default, the following colours are assigned to Virtual portsvirtual ports: , , , , , , , DSS1 BRI DSS1 PRI CORNET ASL CO GSM SIP Proxy SIP Gateway, , , , . These settings can be changed within SMTP client SMTP server Software SMPPthe application setting as described in Chapter .1. About Application

Creating Virtual PortsBy default, the database contains two basic virtual port types – and Default IN Default

. The virtual ports are created automatically in the menu. You OUT Hardware – Boardscan create more virtual ports and virtual port types manually using the following context menu options:

Add virtual port type – create a new virtual port type. To assign a virtual port to a new virtual port type use the function or the parameter on Drag&Drop Typethe tab.Basic

Add virtual port – initiate a dialogue box for adding a new virtual port. Enter the virtual port name and choose the stack type from the available submenu list. The offerings depend on which menu you use. The menu includes a Virtual ports – Alllist of all stack types, but the menu provides a list of Virtual ports – CornetCornet stack types only. Manually created virtual ports are not assigned to physical ports automatically. They have to be assigned manually using the

. Hardware – Boards

Delete – delete a selected virtual port or virtual port type.

Rename – rename a selected virtual port or virtual port type.

Copy – make a copy of the selected virtual port or virtual port type retaining its settings (only the items that may not be identical are changed).

Assign name to unchanged according to physical port – rename all the virtual ports that have not been renamed yet according to the physical ports they are assigned to.

Assign name to all according to physical port – rename all virtual ports according to the physical ports they are assigned to.

Set parameters as Default IN – this option is only available for the port type and helps you set all parameters of a new port type quickly according to Default IN.

Set parameters as Default OUT – this option is only available for the port type and helps you set all parameters of a new port type quickly according to Default OUT .


Export SIP terminals – export the list of terminals of all Proxy ports into a text or csv file.

The structure of the list of exported SIP terminals

proxy (virtual port) name;terminal name;terminal type;MAC address;IP address;content of the UserAgent header

Moving records using the mouse, also called , has been implemented in drag & dropthis menu for an easier transfer of existing virtual ports between the virtual port types.

The subsection below provides a description of the virtual port and virtual port type tabs. All the tabs and parameters defined below are common for all virtual port types. Some parameters or tabs are omitted in some virtual ports because they have no sense there.

BasicThe tab includes the following parameters:Basic

Name according to the physic port – rename a virtual port according to the physical port to which it is assigned. The name consists of the stack name and hardware address in the square brackets. In the event of a manual name change, the option keeps automatically unchecked.

Type – assign a virtual port to a specific virtual port type, which represents another hierarchical level for some parameters.

Enable call without extension – enable/disable answering of incoming calls without the CLI. This parameter is enabled by default. For example, it can be used where a terminal is connected to a certain physical port and no extension has been assigned to the virtual port.

Internal numbering plan – set the Calling Line Identification (CLI) subtype to . is typically set for internal ports and for external ports. If is Internal YES NO YES

selected, no CLI normalisation is made and Mobility Extension terminals are not recognized.

Call on port is accounted – insert an indicator (a = accounted) in the accounting sentence that is used for charging outgoing calls through this port. The accounting sentences are thus easily traceable by the accounting software.


CLI section

Identification tab – assign an identification table to a virtual port. Choose any of the tables available in the menu. The selected Routing – Identification tablesidentification table is used for changing the Calling Line Identification (CLI) for outgoing calls through the corresponding virtual port.

Add prefix for external CLI – assign a prefix to the virtual port as defined in the menu. The prefix is then added to the Calling Global data – Global parameters

Line Identification for all external subtypes, but does not influence number assignment to a phone directory name. The prefix addition facilitates CallBacks for the virtual ports that do not support the number subtype (analog lines, SIP).

Keep number subtypeThe parameter defines the final subtype of incoming and outgoing numbers for a virtual port, including the Calling Line Identification (CLI) and Called Party Number (CPN).

You can set the following:

Incoming CLIP – set whether or not an incoming CLI shall be retained.

Incoming CPN – set whether or not an incoming CPN shall be retained.

Outgoing CLIP – set whether or not an outgoing CLI shall be retained.

Outgoing CPN – set whether or not an incoming CPN shall be retained.

Meanings of set values:

Default – settings from higher levels can be taken over.

Replace unknown – the numbers are only normalised that arrive in the PBX with the subtype. The other subtypes are retained. Normalising Unknowntakes place as defined in the menu.Localisation

Replace always – all incoming numbers are normalised.

Retain – no number is normalised. The numbers are further processed with the subtype they arrive in the PBX with.

AutoClip routersThis section is used for assigning a selected AutoClip router to a virtual port. Assign the AutoClip routers for calls and messages separately but you can use one and the same AutoClip router. For details on AutoClip routers refer to Subs. 7.7 AutoClip router.

Calls – assign an AutoClip router for saving records on outgoing calls. To make the function work, assign the to the calling user in the AutoClip parameters

menu on the user or use group level. To assign the Routing – Users & groups use the menu.AutoClip parameters Global data – Autoclip parameters

Messages – assign an AutoClip router for saving records on outgoing SMS. To


Messages – assign an AutoClip router for saving records on outgoing SMS. To make the function work, assign the to the calling user in the AutoClip parameters

menu on the user or use group level. To assign the Routing – Users & Groups use the menu. One and AutoClip parameters Global data – AutoClip parameters

the same AutoClip router can be used both for SMS and calls.

Cause mappingIn this section, you can specify your own sets of causes to be used for signalling. To do this, use the menu. You can set a Global data – Causes – Cause mapping tablestranslation of a certain event into the given interface (SIP, GSM a ISDN) in the cause mapping table. Use these sets only if the predefined cause translations are inconvenient.

CP to stack – define a specific cause translation for changes from the PBX to the virtual port. Hence, it is a change of a specific internal cause into any stack cause.

Stack to CP – define a specific cause translation for changes from the virtual port to the PBX. Hence, it is a change of any stack cause into a specific internal cause.

You can choose a mapping table for each direction, the use of a table assigned disableto a virtual port type or use to enable the default table.Default

Name information sendingThe settings in this section are primarily intended for the SIP Gateway port. Select one of the following two parameters to insert the caller's name in the From field of the SIP INVITE message.

Find name in group phone book – select a group whose phone directory shall be used for matching the calling number and a name in the phone directory.

Insert calling station name – define whether the calling stattion name shall be added to the outgoing INVITE message.

Call admission controlThis section helps you limit the count of calls via a selected port. The following parameters are available:

Internal call count – limit the count of internal calls. When a call comes in and goes out via one and the same port, the parameter rises by 2.Internal call count

External call count – limit the count of external calls. When a calls comes in via the selected port and goes out via another, the parameter is External call countincreased by 1 for both the ports.


Meaning of the options:

Default – default value 0 = unlimited.

Unlimited – unlimited count of calls.

Limit to – set the maximum call count.

Own channel countDisplay the count of voice channels that can be served by the virtual port.

Licences neededIn this section, you can check easily whether the or Mobility Extension Call Recordinglicence is required on the virtual port. If a licence is required yet absent or insufficient, it is in red letters here. If a licence is valid, the text is blue highlighted.Valid licence

Used sources for call admission controlThis section displays how many internal/external calls are being made via the selected port at a moment.

PropertiesThe tab consists of a number of subtabs, which are described in a separate Propertieschapter. This tab is exceptional because almost all of its parameters obey the fall-down hierarchy. For the hierarchy and parameter details refer to the Setting Propertiessubsection.

Progress InfoThe parameters in this tab help you enable/disable progress tones that are to be played back to the user. In some cases, the progress tones are generated by the PBX, in others they are transmitted from the network. The final effect depends not only on the user's virtual port, but also on a combination of the user's virtual port (mostly an internal PBX port) setup and the other party's virtual port (a public network port or another internal PBX port).

The port generates the network progress info into the opposite port:

Dial tone – the port is a source of the network dial tone for the opposite port if the latter requires the dial tone.

Alert tone – the port is a source of the network alert tone for the opposite port if the latter requires the alert tone.

Disconnect tone – the port is a source of the network disconnect/congestion/busy tone for the opposite port if the latter requires the disconnect tone.


Setting options

Default – provide fall-down to the next level (virtual port type).

Yes – enable use.

No – disable use.

Conditionally – if a tone is signalled by the network, the generated tone is played. If a tone is not signalled, the internal PBX tone is played. The Reset

section is applied to this setting only. This setting is conditionrecommended especially for the SIP Gateway virtual port.

The port requests progress info from the PBX or the opposite port:

Dial tone – the port requests the dial tone from the PBX or the opposite port that generates the dial tone.

Alert tone – the port requests the alert tone from the PBX or the opposite port that generates the alert tone.

Disconnect tone – the port requests the disconnect tone from the PBX or the opposite port that generates the disconnect tone.

Setting options


Yes – enable use.

No – disable use.

Reset condition – enable playing of some PBX tones and some network tones for one call.

Parameters

Alert resets condition – an incoming Alerting message resets the tone-generating condition and signalling of the played tone is awaited again.

Connect resets condition – an incoming Connect message resets the tone-generating condition and signalling of the played tone is awaited again.

Disconnect resets condition – an incoming Disconnect message resets the progress tone condition and signalling of the played tone is awaited again.

Setting options


Yes – enable use.

No – disable use.


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The following examples are given for easier comprehension:

Suppose that user 's phone is connected to an internal PBX port. Set the dial Atone request for this port for user to hear the dial tone after picking up the Aphone. If user makes a call to user connected to another internal PBX port A Band you want user to hear the alert tone, set the alert tone request for user 's A Aport too.

Suppose that your PBX is connected to a public or private network that generates progress info. If you want the calling user to hear the alert tone, set the port used for calling into the public or private network to generate the alert tone and the user extension port to request the alert tone. Otherwise, the calling user would not hear the alert tone during outgoing calls. The disconnect tone is mostly generated by own PBX.

Outgoing call to PSTN with conditioned generating

Suppose that a call is going out to the PSTN, which generates the dialtone only. However, you want the user to hear the ringing and disconnect tones too. Therefore, set the in the section and the Conditioned option Generate tone Alert

parameter at least in the section for all the three resets condition Reset conditiontones for the port used for the outgoing call. Suppose that the three-tone requirement is set for the user's internal extension. Thus, when a CO line is seized, the user is played the PSTN dialtone first and then the internal PBX ringing tone after the dialling and ringing start (Alerting state signalling), because the PSTN generates no ringing tone. When the PSTN subscriber hangs up, the user is played the internal PBX disconnect tone.

Terminate call when PROGRESS_IND received – here enable call termination on selected ports in case the oppopnent signals progress tone playing during call setup. Moreover, you can define the call rejection cause. Two basic modes are defined for call setup termination:

Before alerting – when the PROGRESS_IND message comes before the opponent's alert signalling.

During alerting – when the PROGRESS_IND message comes after the opponent's alert signalling.


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OverlapOverlap is one of the Called Party Number (CPN) sending methods. If enabled, the CPN is not transmitted all in a SETUP message, but digit-by-digit in an INFO message.

The setup consists of the following parameters:

Overlap sending – enable overlap sending in the port-to-PBX direction. It is primarily used for ISDN virtual ports.

Overlap receiving – has not been implemented yet. The selection is inactive.

Overlap dialling – has not been implemented yet. The selection is inactive.

First digit timeout [ms] – set the first digit dialling timeout starting at the moment of the microtelephone pick-up. When it expires, the user cannot go on dialling, obtains the disconnect tone and the call connection is terminated. The default timeout value is 14 s.

Next digit timeout [ms] – this parameter sets the next digit dialling timeout restarting after each digit received. When it expires, the call establishment begins. The default timeout value is 6 s. To initiate call establishing before the timeout, push the button.#

ExtensionsThe tab provides a list of extensions assigned to the virtual port. There are Extensionsthree forms of the tab depending on the virtual port type:

With the , , and virtual ports, the tab structure BRI PRI SIP Gateway SIP Proxyrespects the presence of a terminal. To create a terminal, use the tab. HWTerminals are used for authorisation, MSN numbers and extension assignment. The tab consists of three parts. The first window from the left Extensionsincludes a list of terminals assigned to the virtual port. If you have not created any terminal, you can use the one. The central window provides a list of Defaultextensions assigned to the selected terminal. The field on the right-hand side of the menu helps you select a extension to be active among multiple extensions assigned to the terminal. You can make calls to all of the extensions, but all outgoing calls from this terminal are identified as the active extension and accounted to this active extension too (except for the Private call from my

service).extension

With the virtual port, the situation is similar. The difference is that Cornetextensions are assigned as Master or Slave terminals because you can connect just one terminal to a Cornet port. If multiple extensions are assigned to this virtual port, specify the active extension (one for Master and one for Slave).

With the , and virtual ports, the situation is the simplest. The tab has ASL CO GSMtwo parts only. One is used for extension assignment and the other for active extension specification.


Free Minutes/SMSThe tab helps you set free minutes and SMS for a selected virtual port.

Select tariff rateClick on the button to display a dialogue and select one of the Set free minutes/SMStariff rates as defined in the menu. In addition, you can Accounting and tariff ratesassign here a setting to the selected virtual port tariff rate as defined earlier for any other virtual port. To change the tariff rate if necessary, use the option. Used tariff rateIf you do so, you will lose all data saved on free minutes with the given tariff rate via this virtual port. To cancel the virtual port tariff rate, push the Cancel free minutes/SMSbutton.

Free minutes/SMS settingsOnce a tariff rate is selected, the tariff rate credit rows are displayed in this section. Click on a row to display a setting dialogue for the count of free minutes, SMS messages and other credit parameters for the given virtual port. See the figure below for the dialogue.


The table includes columns with the following meanings:

Credit name – the credit name as defined during tariff rate creation.

Free minutes for month – the column includes the count of free minutes per month for the given virtual port. This count is credited to the given virtual port at the beginning of the accounting period. If the free minute count changes within a month, the port credit is not increased until the beginning of the next accounting period unless provided otherwise in the setting dialogue.

Free minutes for this month – the column shows the current count of free minutes to be used in this month. The value includes free minutes transferred from the previous accouting period if any.

Spent minutes – display the current count of minutes spent in the accounting period.

Free SMS for month – the column includes the count of free SMS messages per month for the given virtual port. This count is credited to the given virtual port at the beginning of the accounting period. If the free SMS count changes within a month, the port credit is not increased until the beginning of the next accounting period unless provided otherwise in the setting dialogue.

Free SMS for this month – the column shows the current count of SMS messages to be used in this month. The value includes free SMS transferred from the previous accouting period if any.

Spent SMS – display the current count of SMS sent in the accounting period.

Account – set the accounting frequency, i.e,. the accounting period length. On this date, the free minute and SMS counts are increased according to the selected transfer mode. The mimimum values are set in the Free minutes for month a Free SMS for month columns.

Mode – select the method of transfer of old free minutes into the next accounting period.

Do not transfer – no free minutes and/or SMS are transferred.

First use new – old free minutes and SMS are transferred but new ones are used first. Unused units older than one month are not transferred.

First use transferred – old free minutes and SMS are transferred and new ones are not used until these old units have been exhausted. Unused units older than one month are not transferred.


FilesThe menu displays the current files with records of calls via the selected virtual port or virtual port type. The menu consists of a simple table with five columns with the following meanings:

Name – name of the locked file.

File type – type of the file.

Created – the moment of file creation.

Valid for – the file locking time, in other words a file storing time in a physical storage. When this time elapses, the file will be deleted.

Size – size of the file.

CLIP Scheme – scheme of the calling number.

CLIP Type – type of the calling number.

CLIP Number/URI – number or URI of the calling subscriber.

CPN Scheme – scheme of the called number.

CPN Type – type of the called number.

CPN Number/URI – number or URI of the called subscriber.

Moreover, the context menu provides the following record handling options:

Save – transfer the file from a storage to the PC.

Listen – play the selected file.

Remove – remove the selected file from a storage.

Remove all – delete all files from a selected storage.

StackThe tab is described in S. depending on the stack type.Stack 3. Virtual Ports


4. SIMHere is what you can find in this chapter:

4.1 SIM cards

4.1 SIM cardsThe menu includes a list of all PBX SIM cards. This menu is Virtual ports – GSM – SIMopened automatically whenever the SIM card is inserted in the PBX and the parameters filled in by the user (e.g. PIN) are used automatically for any future system detection of the SIM card. The menu includes two tabs.

Basic

Card serial number – this parameter shows the SIM card identification code

detected. uses this code automatically for SIM card 2N® NetStar Admin

identification in the list to the left.

PIN – Personal Identification Number – insert the SIM card PIN if requested to be shared by the user and the authentication system. If the PIN is requested yet not entered or entered incorrectly, the SIM will not be logged into the network.

PUK – Personal Unblocking Key – insert the PUK to unblock the SIM card in case you have entered three incorrect values of the PIN code.

SMS centre number – enable SMS sending. In GSM networks, SMS messages are not routed directly to the final destination, but through the provider's SMS centre. This is useful where an SMS cannot be delivered immediately (e.g. due to target phone unavailability). The SMS centre tries to deliver this message cyclically for a preset SMS validity time. This parameter is mostly automatically detected on the SIM card (preset by the provider). If not, fill it in manually.

Phone number – this field is for information only. You can enter your SIM card telephone number for easier orientation. This parameter has no function.

Free Minutes/SMSUse the tab to set the count of free minutes and SMS mesages via the selected SIM card. Refer to Free minutes/SMS in Subs. for details on 3.10 Virtual Port Optionscontrols and tables.


5. NetworkHere is what you can find in this chapter:

5.1 Network Interface

5.2 Routing table

5.3 Service Settings





5.3.5 DHCP server


5.3.7 API

5.4 Dohledové služby



5.5 DB Conncetors


5.1 Network InterfaceThe menu helps you manage all network interfaces Network – Network Interfacesavailable in the PBX. In addition to the CPU interface, there are Ethernet interfaces of VoIP boards. The bit rate of all the interfaces is 10/100 Mbit/s. These interfaces are used for communication with the PBX and SMTP clients, for signalling and RTP streams of VoIP calls. Having been opened, the menu Network – Network interfacesdisplays a list of Ethernet interfaces of the PBX on the left and selected interface settings on the right. With the CPU interface, the options are as follows:

Get IP address from DHCP server – enable obtaining IP settings from the DHCP server automatically. In this case the following sections are inactive.

Use following IP address – enable the following static IP address and DNS server setting sections.

IP address – define the static IP address for this interface.

Subnet mask – define the subnet bit mask.

Default gateway – define the IP address of the router or PC through which the PBX communicates outside the LAN.

DNS server addresses

Preferred server – define the IP address of the primary DNS server.

Spare server – define the IP address of the secondary DNS server.

DNS HostName – define the PBX Host Name.

DNS Domain – define the PBX Domain Name.

Description – this field is for information only.

Producer – this field is for information only.

Tip

You will lose connection with the PBX whenever you change the IP address. We recommend you to change the IP address via the console menu on the serial interface.


5.2 Routing tableThe routing table includes records on network routes. When the CPU address is set, two default static records are created. Additional records can be added and removed via the context menu.

Figure: Routing Table with Default Static Records

Warning

Make sure that the settings are correct to avoid the PBX connection error.


5.3 Service SettingsHere is what you can find in this section:





5.3.5 DHCP server


5.3.7 API

5.3.1 Time Synchronisation (NTP)The menu helps you define the NTP server to be used for time synchronisation by the PBX. After checking the option in the upper menu part, enter the IP address or domain name of the existing NTP server into the field under the checkbox. After saving the data, the PBX will try to synchronise time with the preset NTP server. The result of this action is always shown in the field together with information Synchronisation resultabout the next planned synchronisation attempt.

Figure: View of Time Synchronisation Menu


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What Is TFTP?The is a very simple file transfer protocol, containing just Trivial File Transfer Protocolbasic FTP functions. The TFTP works over the connectionless protocol. A single UDPfile can only be transferred through one connection. A single packet is only present in the network during communication. Having sent a packet, the program waits for confirmation and only then sends another one. Due to such simplification, the protocol provides just a to lines with a high latency. The TFTP uses low transmission rate port

.69

It has some restrictions and differences compared with the FTP:

Directories cannot be browsed through.

No user login and/or password entering is possible.

Can be used for data reading and/or remote writing only.

Supports the following three different transfer modes:

netascii – for an ASCII text with modifications from the Telnet protocol

octet – for raw binary 8–bit data

mail – for e-mail sending; this mode should not be used any longer

The maximum size of the file to be transferred is 32 MB.

TFTP in 2N ® NetStar

In , the TFTP storage is used as a root directory for the TFTP server where 2N® NetStar

the files provided by the TFTP server (which is part of every unit) to 2N® NetStar

TFTP clients are located. The typical TFTP client is an IP phone, which requests configuration, a phone directory or new firmware from the TFTP server. The TFTP server searches this directory and provides the file, if found, to the client.


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Using TFTP in 2N ® NetStar

The TFTP is used for downloading the following files from :2N® NetStar

gs_phonebook.xml, which contains the compatible phone book. GrandStream Set the source phone directory in the menu. SIP phonebooks

tftpPhoneBook.xml, which contains the compatible phone 2N® StarPoint IP Txx

book. Set the source phone directory in the menu . SIP phonebooks

y0000000000xx.cfg and , plus , which <MAC_address>.cfg contactData1.xml

contain the configuration and phone directory, respectively, for the 2N®

phones.StarPoint IP Txx

hipv-common.xml and l configuration files for the hipv-MAC_address.xm 2N®

intercom.Helios IP Vario

ConfigurationThe context menu provides the following options:

Refresh – refresh the root storage for updated view.

Delete – remove a file from the root storage.

Rename – rename a file within the root storage.

Add file – add a PC file to the root storage.

Save file – save a root storage file into your PC.

The meanings of the table columns are as follows:

Name – display the file name within the root storage.

Size – display the size of the file added.

Changed – display the date and time of the last file update.

Attributes – display additional information on the file.

Example for 2N ® StarPoint IP T28Log in to the telephone web interface (default login data: admin, admin) and move to the tab. Here select the middle link in the Phone directory Remote phone directoryupper part. Enter the following string into one of the fields: tftp://PBX_IP_address

. Save the data. Go to the /tftpPhoneBook.xml Users – Phone directories – SIP phone

directories menu in and select the phone directory source. Now push the 2N® NetStar

directory access button on your phone to download the directory from the PBX.

tftp://PBX_IP_address/tftpPhoneBook.xml

tftp://PBX_IP_address/tftpPhoneBook.xml


5.3.3 TCP/IP Communication PortThe menu is used for management of ports via which you TCP/IP Communication portcan access your PBX. Basically, you can only or a port in this menu, Add Removeenabling/disabling the authorisation requirement. It is only port 6992 that requires authorisation after initialisation.

Figure: View of TCP/IP Communication Port Menu Configuration

If the PBX is accessed via a password-protected port, the tab for direct Databaseconfiguration is not displayed for the user or administrator by default. To display it, assign the Read and Write rights to the user or administrator using the Users – User

menu, or use a PBX port without authentication. In such case, however, you rightsexpose your PBX to the risk of unauthorised access. The database access is unnecessary for common configuration needs and should be granted to experienced technicians only.

As shown in figure above, you can define more parameters for each port (except for authorisation requirement).

KeepAlive packets – enable/disable sending of keep-alive packets on a port to keep communication.

HeartBeat interval – define the time interval between the keep-alive packets.



Communication PortsPorts 22 (SSH) and 23 (TELNET) are closed for security reasons in the default configuration in the firmware version 2.3.0 and later. To open them use a console or this configuration tool. Enter the "root" login for Telnet or SSH. The password is not defined by default. You are strongly recommended to set the password! The password setting command syntax is as follows:

passwd root <Enter>

Change password for root

New password: <Enter>

Retype password: <Enter>

The password will be changed regardless of the original password. If you forget your password, you can change it any time in the same way. All you have to know is your console login name.

Use the "Admin" login for the console and password "2n" (by default).

Warning

If you, despite recommendations, use the TELNET and SSH protocols for login to the NetStar PBX, any software warranty provided by the manufacturer shall be null and void. The system access is logged and intended for servicing purposes only.


Menu SystemTo open or close a port using the configuration tool, go to the Network – Service

menu and use the or options. The setting – System services Enabled Disabledmeanings of the options are as follows:

Internal server of Assistant – open or close access to the Assistant web server (user web application).

Telnet server – open or close access to the system via the Telnet protocol.

SSH server – open or close access to the system via the SSH protocol.

Trace level – enable the Linux level for trace message displaying. This option is especially useful if you need to supervise the PBX remotely with a poor connection.

All – all available trace messages. Basic settings.

None – discontinue all trace messages.

Call mng – messages related to call routing by the PBX.

Stack – just messages related to call processing on the PBX interfaces (GSM, ISDN, SIP, analogue).

HEAP – write out the PBX memory load messages in special development firmware versions only.


5.3.5 DHCP serverThe is used exclusively for assigning IP addresses and other parameters DHCP serverto SIP terminals with the specified MAC address in NetStar.

The menu consists of two basic sections. Look left for a field to set the ranges of subnet IP addresses to be assigned and right for more parameter settings for the selected subnet.

SubnetThe context menu provides the following options:

Add new subnet – display a dialogue to define the required subnet parameters, see below.

Change range – edit the existing subnet range. It has the same function as a double click on the selected subnet.

Delete subnet – delete the selected subnet from configuration.

You can specify the following in this dialogue:

IP address range

Subnet mask

Default gateway

Preferred DNS server

Backup DNS server

The first obligatory step is to enter the range of IP addresses. The remaining parameters need not be filled in immediately if you unselect them. You can edit the parameters later in the subsection.Subnet properties

Subnet Options


Subnet OptionsThe context menu provides the following options:

Add subnet mask – the option is only active for subnets with no mask defined so far.

Add default gateway – the option is only active for subnets with no default gateway defined so far.

Add DNS server(s) – the option is only active for subnets with no DNS defined.

Edit value – edit the existing values. It has the same function as a double click on the selected parameter.

Remove option – remove a parameter from the selected subnet configuration.



What Is a Directory ServiceDirectory service is an application running on the Directory server, which collects and provides information on the named and frequently accessed objects that seldom change. The information is stored in the form of tree-structured records on the directory server. The Lightweight Directory Access Protocol (LDAP), working on the client-server principle, is a convenient tool for storing and accessing data on the directory server. The LDAP also includes client authentication.

LDAP in 2N ® NetStarThe Directory service menu consists of two main sections. A list of available directory servers is located to the left and server parameters are to the right. Click the context menu in the server list with the right-hand mouse button to create, rename or delete a server. The LDAP server configuration includes:

Figure: View of LDAP Settings

Settings

User – use this item for user authentication in server communication. Enter without the domain.

Password – enter the user password.


LDAP Address – enter the IP address or domain name of the LDAP server to which the PBX gets connected.

Domain – enter the whole domain including the highest order domain ( , tel-2n.cze.g.).

Port – enter the port number for directory server communication. The default port is 389.

Authentication – select one of the following three authentication protocols for user authentication in server communication:

Simple – user name (DN – Distinguished Name) and password based authentication.

KERBEROSv5 – Kerberos based authentication.

Address – set the KDC server (Key Distribution Centre) address for user authentication.

Port – set the KDC server port. The default value is 88.

NTLMv2

PartitionsUse the context menu to add or remove an organisational unit – phone book.

Phonebook – set the PBX phone book (directory) to which the records obtained from the LDAP server shall be stored.

Suffix – define the search area or directory level from which synchronisation with the LDAP server shall be made. Start with the lowest level.

Example:

Suppose there is a structure on the LDAP server ( domain) including a group tel-2n.cz(2N) and subgroups (Development, Sales, etc.). Enter the suffix ou=Development,

to download the contacts of the Sales subgroup.ou=2N,dc=tel-2n,dc=cz

Phone type – define the type of the record with which synchronisation with the LDAP server shall be made.

Caution

The records that are not included in the LDAP server are deleted from the assigned phone book during synchronisation.

Synchronisation resultUse the menu to schedule synchronisation. Add the Global data – Scheduled tasks

event and set the synchronisation time. View the LDAP synchronisationsynchronisation result in the lower part of the screen.

http://tel-2n.cz

http://tel-2n.cz


5.3.7 APIThis menu helps you define the TCP ports for (Application Programming APIInterface) communication via the XML and HTTP protocols. Use the context menu to modify the port and maximum count of clients that can communicate via the interface at the same time.

Figure: API Default Settings

Caution

One port can only be assigned to each protocol type.


5.4 Dohledové službyHere is what you can find in this section:



5.4.1 Event ReporterFind the Event reporter in the menu. Network – Supervision services – Event reporterHere you can set the basic rules for sending info SMS on system parts. This function is subject to licence!

Event – set the event type to be SMS-reported. Choose one of the following options:

PBX restart – PBX restart notification.

PBX keepalive – sending of KeepAlive messages for PBX operation check. Set the KeepAlive sending period in the Scheduled events in the Global

menu. data

Port ready – notification of virtual port function reactivation.

Port busy – notification of virtual port occupation. The BRI and PRI ports are considered busy whenever all channels have been occupied. This function cannot be used for the SIP or SMTP port.

Port error – virtual port error notification.

Storage error – notification of a functionless physical storage (due to network failure, e.g.).

Storage full – notification of a functionless logical storage due to overfilling.

No call port credit – low credit notification for a defined virtual port.

No call SIM credit – low credit notification for a defined SIM card.

No call terminal credit – not implemented yet.

No call user credit – low credit notification for a defined user.

No SMS port credit – notification of SMS limit exhaustion for a defined port.

No SMS SIM credit – notification of SMS limit exhaustion for a defined SIM card.

No SMS terminal credit – not implemented yet.

No SMS user credit – low credit notification for a defined user.

State of status control object Error – notification of the status control object transition to the Error state.

State of status control object OK – notification of the status control object transition to the OK state .


Set the report sending hysteresis to avoid excessive report [s] – set the report sending hysteresis to avoid excessive report sending due to repeated port changes. The states to which a hysteresis timeout applies are reported at once (within one report) after the timeout.

Report on Active NotificationDefine how to report an active event:

Message – specify the SMS text to be sent whenever the selected event occurs (Port error, Storage full, etc.), i.e. when the event is active. If the field is Messagenot completed, the SMS will not be sent. In addition to standard texts, the following dynamic strings can be entered:

%n – name – enter the name of the event reporter that recorded the event.

%d – date – enter the PBX date and time valid at the instant of event recording.

%k – key – enter the name of the port to which the event relates.

%v – value – enter the event value. At present, there is no event to meet this parameter.

Relay action – this option is not available until the port is selected in the Used block. Choose the required relay action:relay

Switch on – close the relay of the below-specified port whenever some of the conditions defined in the is met.Event parameters

Switch off – open the relay of the below-specified port whenever some of the conditions defined in the is met.Event parameters

Positive pulse – close the relay of the below specified port for a period defined in the parameter whenever some of the Relay pulse widthconditions defined in the is met.Event parameters

Negative pulse – open the relay of the below specified port for a period defined in the parameter whenever some of the Relay pulse widthconditions defined in the is met.Event parameters

Relay pulse width – if above is enabled, set the Positive pulse/ Negative pulse pulse width in milliseconds.

Report on Inactive NotificationDefine how to report an inactive event:

Message – specify the SMS text to be sent whenever the selected event ceases to exist (Port error, Storage full, etc.), i.e. when the event is inactive. If the

field is not completed, the SMS will not be sent. In addition to standard Messagetexts, dynamic strings can be entered (see above).

Relay action – this option is not available until the port is selected in the Used block. Choose the required relay action.relay

Relay pulse width – if in the above mentioned Positive pulse/ Negative pulse parameter is enabled, set the pulse width in miliseconds.


Parameter Evaluation

Active notify evaluation – specify under which conditions the message shall be sent. The following options are available:

Independent of parameter – messages on active/inactive events are sent for each parameter separately, independently of the states of the other parameters.

At least one parameter active – messages are sent for each parameter with active event separately, independently of the states of the other parameters. If the string is used in the SMS text, the SMS always %kcontains a list of parameters with active event. The inactive event message is not sent until the event ceases to exist for all the parameters.

All parameters active – this selection represents a logical AND of all the selected parameters. The message is sent only if the event condition is met for all of the selected parameters. Messages on inactive events are sent for each parameter separately, independently of the states of the other parameters. If the string is used in the SMS text, the SMS always %kcontains a list of parameters with active event.

Deglitch active change shorter than [s] – restrict the evaluation of active events in the case of abrupt changes. If the selected event exists for a time period shorter than as defined here, the event will not be evaluated as active.

Deglitch inactive change shorter than [s] – restrict the evaluation of inactive events in the case of abrupt changes. If the selected event stops for a time period shorter than as defined here, the event will not be evaluated as inactive.


Figure: View of Event Reporter Settings


Send as User

User – define the user to be presented as the message author.

Send to User

User – define the user to which the message shall be sent.

Save to User – enable/disable saving messages to the user regardless of the user settings, or respecting the user settings.

SIP extensions – enable/disable resending messages to user SIP extensions regardless of the user settings, or respecting the user settings (According to stations).

Email extensions – enable/disable resending messages to user email extensions regardless of the user settings, or respecting the user settings (According to stations).

Mobility Extensions – enable/disable resending messages to user external extensions regardless of the user settings, or respecting the user settings (According to stations).

SNMP

– specify the SNMP user for notifications. The SNMP block is not Notificationavailable yet.

Used Relay

Port – specify the port whose relay is to be closed whenever some of the conditions defined in the is met.Event parameters

Event ParametersThe block is accessible if one of the above mentioned options (PBX restart, Port ready, Port error, etc.) is selected in the . A survey of available Event parametersobjects related to the event is to the left and a list of objects currently monitored by the given Event reporter is to the right. Use the arrows to move the objects from one side to the other.

ExampleRefer to Figure 1 for GSM port trunk occupation. If all the GSM ports are occupied, the addressee is sent an "All ports occupied" SMS. When some of the GSM ports get released (the selected event is inactive), the addressee is sent the "Port %k is free" SMS, where %k represents the list of available ports.


HTTP Commands

NetStar2N® allows you to close/open the defined relay via the HTTP commands on

default port 8088. Change the port in the menu. Network – Network settings – API Enter the HTTP command as follows:

http://ns_address:port/httpAPI.xml?port=1&relay=ON&pulselen=1000

where

port is the port ID BIO

relay can be (closed) or (open)ON OFF

pulselen defines the closing/opening pulse time in milliseconds. The parameter is optional.

Example:Close relay:

http://192.168.100.100:8088/httpAPI.xml?port=1&relay=ON

Open relay:

http://192.168.100.100:8088/httpAPI.xml?port=1&relay=OFF

Close relay for 3 seconds:

http://192.168.100.100:8088/httpAPI.xml?port=1&relay=ON&pulselen=3000

Caution

The HTTP command is Lower/Upper Case sensitive. Keep the format to avoid the HTTP command failure.



What is SNMP?The is part of the Internet protocol Simple Network Management Protocol (SNMP)suite as defined by the Internet Engineering Task Force (IETF). The SNMP is used in network management systems for data acquisition and network monitoring for administration purposes. It consists of a set of network management standards, including the Application Layer protocol, a database schema and a set of data objects. The SNMP is available in three versions. Compared with the first version, version 2 is enhanced with authentication and version 3 with encryption. A majority of today's devices support the SNMP version two.

There are two sides in the SNMP communication – the monitoring one and the monitored one. These parts can run on separate physical devices or within one piece of equipment. The monitored side is often called and the monitoring side Agent

. The monitoring side flexibly collects information on the system state. The Manager sends requests to the , mostly requesting some system state Manager Agent

information. The provides responses to the . The Agent Manager Agent-Managercommunication is often marked as an .SNMP operation

The or is an identifier used for explicit identification of each value OID Object Identifierin the SNMP communication. The OID is composed of a dot-separated sequence of numbers where each dot represents one level of the OID tree structure. The numerical identifications in subtrees are not unique and that is why the OID is always sent as a whole string. Each company and each of its SNMP supporting devices has an international OID of its own.

The or is used for translation of OID strings into a MIB Management Information Basemore comprehensible text. The can be extended to include more MIB database MIB

.files

UsersThe SNMP v3 is a user oriented communication protocol. The user created in this part of configuration corresponds to the (User Security Model) in the SNMP v3 and to USM

in the other versions. In addition to standard options , and Community Add Delete user, the option is available, which helps introduce the default SNMP Rename Default

setting, including creation of the user, right line and filters public Unrestricted Internetand .NetStar Traps

Authentication – define the password and way of encryption for authentication.

Protocol – use the or methods to secure your password.MD5 SHA

Password – enter the user password.

Privacy – define the password and way of encryption for data transmission.


Privacy – define the password and way of encryption for data transmission.

Protocol – use the or methods to secure your transmission.DES AES

Password – enter the encryption password.

Access – assign rights to a selected user using the list of available rights.

Rights

Name – define the name of the right to be created. This name is displayed in the selection.Users

Context – use a text string to identify the SNMP module within the client address. This parameter need not be filled in.

Full match context – enable requirement of full match including context. It is mostly unnecessary.

Security model – choose either a specific security model (SNMP v1, SNMP v2c, USM = SNMP v3) or the option. Any selection has to be supported by Whateverthe other party too since no communication feedback is available.

Minimum security level – the parameter offers three different models:

Authentication and privacy

Without authentication and privacy

Authentication only

Read filter – set the Read filter by choosing an item from the list of available filters on the tab. The filter restricts access to the PBX information for Filtersselected users.

Write filter – set the Write filter by choosing an item from the list of available filters on the tab. The filter restricts writing within the PBX for selected Filtersusers.

Notify filter – set the Notify filter by choosing an item from the list of available filters on the tab. The filter restricts notifications from the PBX for Filtersselected users.

Figure: View of SNMP User Right Setting Menu

FiltersThe tab displays the list of filters created. The and filters Filters Internet NetStar Trapsare created by default. Use the context menu to add, edit and derive the filters.

OID root – set the OID tree root to be used as a base for filter setting. You can view the OID structure in a tree or an alphabetical list.


Exception – change the meaning of a filter rule. If this option is not checked, the defined OID subtree is used. If it is checked, the use of a subtree from this row is denied. With this parameter you can specify that the whole section 2.1 will be used except for subsection 2.1.3.

OID subtree – choose restrictions for a subtree. If the row is empty, the whole of the above specified OID root is used. Every filter should have one rule at least, even an empty row. It is because the filter compares rules with the subtrees instead of the OID root. If some subtrees overlap, the most common (the shortest OID) rule is applied.

MIB files

Add – add a selected MIB file to the MIB database.

Delete – delete a selected MIB file.

Recompile – recompile a selected MIB file.

File – this column shows the path to the MIB file source. This path is relevant for the option.Recompile

Status – this column shows the current status of a MIB file. The options are , and . The MIB file statuses are also indicated Compiled Not compiled Not found

by the icons on the line beginnings as shown in Figure 2.

Additional information – display additional information.

Figure: View of MIB File Management Section

Default Notify filter options according to RFC3415

Internet access:

subtree 1.3.6.1

Restricted access:

System – subtree 1.3.6.1.2.1.1 according to RFC3918

SNMP – subtree 1.3.6.1.2.1.11 according to RFC3918

snmpEngine – subtree 1.3.6.1.6.3.10.2.1 according to RFC3411

snmpMPDStats – subtree 1.3.6.1.6.3.11.2.1 according to RFC3412

usmStats – subtree 1.3.6.1.6.3.15.1.1 according to RFC3414

When you choose the Internet filter, the following traps are transmitted upon the PBX restart:


1.

2.

3.

4.

5.

ColdStart is sent notifying of the PBX restart or switch-off.

Information is sent on all L1 Active ports, i.e. Cornet, BRI and PRI.

Information is sent on all CO and ASL ports.

Information is sent on all successfully logged-in GSM ports with SIM cards.

Finally, information on error statuses and their elimination is sent: DSS1 deactivation or higher BER or SLIP (according to setting), Cornet deactivation (according to setting), ASL – error, CO – error, GSM – logout or bad signal (according to setting).

AnswerIn this tab specify the ports and the client from which the PBX is able to receive requests.

Peer port – here the PBX expects the SNMP requests and acknowledgements. The default SNMP port is 161.

From this client address only – lock request receiving from a selected IP address or domain name.

Figure: View of Listening Port Setting Section

Notification

Client address – define the client's IP address or domain name to which notifications are filtered as mentioned below are sent.

Client port – define the client port to which notifications are sent.

Used local port – specify the PBX port to be used for sending notifications if necessary. And for receiving info request confirmations. If this option is disabled, the port is selected randomly.

Notification type – select the type of notification to be used. For SNMP v1, Traps may be selected only, for higher versions info requests are also available.

Trap – is an SNMP message sent to the client about an event that should be notified. The message does not require acknowledgement.

Inform request – is an SNMP message sent to the client about an event that should be notified. Unlike traps, inform requests can be resent if undelivered within the acknowledgement timeout.

Repeat – define the count of notification sending attempts.

In interval – define the time interval during which confirmation is awaited from the client.


Version – specify the notification coding type according to the SNMP version used.

SNMP v1

SNMP v2c

SNMP v3

Figure: View of Notification Configuration Menu

User/Community – define the SNMP user that corresponds to the for SNMP USMv3 and for the other versions.Community

Filter – define the Notify filter. The longer the root and subtree OID, the stricter the filter.

Security level – this parameter can be used for SNMP v3 only and defines the notification security level. Choose one of the following options:

Authentication and privacy

Without authentication and privacy

Authentication only

Context – use a text string to identify a SNMP module within the client address. This parameter need not be filled in.


1.

2.

5.5 DB ConncetorsThe menu helps you set communication with the External Network – DB connectors Routing Machine (ERM server), which can work in the following two modes:

Routing – partially replaces or complements the internal routing mechanisms of

the PBX. Based on a call/SMS routing request, a query is sent to the 2N® NetStar

ERM server. If a matching record is found in the ERM database table, the ERM server returns a response specifying a parameter for further call or SMS routing via the PBX.

Names – is used as an external phone directory. Based on a CLIP-name assignment request, the PBX sends a query to the ERM server. If a matching record is found in the ERM phone directory, the ERM server returns the calling subscriber's name.

The DB connectors created are assigned to External routers ( Routing – External ), which, in the mode, route calls/SMS to the selected destination routers Routing

according to the parameter returned by the ERM server. In the mode, they Namesroute calls directly to the default destination of the External router (the router rows are not applied in this mode).

Use the context menu to the left to add, rename and remove the DB connectors. There two sections to the right: the upper section sets the DB connector properties and the lower one sets parameters for the DB connector – ERM server communication.


Figure: View of DB Connector Settings

DB connector name – set the DB connector name.

DB connector type – select the DB connector type.

Answer timeout – set the time period during which the PBX shall wait for the ERM server reply. If no response comes within this timeout, the call is routed to the default destination in the External router.

Cache by – select how to store the ERM server replies. If a matching record is found in the cache, no query is sent to the ERM server from the PBX. Records can be stored according to the calling/called subscriber. Record storing is disabled by default.

Maximum number of records in cache – set the maximum count of records to be stored.

Valid record time in cache – set the validity for the record to be stored in the cache.

Actual count of records in cache – display the current count of records stored. Click to delete all the records.Clear cache

Port – set the port number for PBX – ERM server communication.

Type – define the DB connector mode:

Routing – for call/SMS routing via the PBX.

Names – for name assignment to CLIP.

Check IP address – select this option and complete the to Checked IP addresscommunicate with the ERM server with this IP address only.

User name – enter the user name for user authentication in the ERM server communication.

Password – enter the user password as set in the ERM server.

Connection state – monitor the ERM server connection state. Click to Disconnectdisconnect from the ERM server temporarily.


6. Global dataHere is what you can find in this chapter:


6.2 Emergency Calls

6.3 Localisation

6.4 Licences

6.5 Language Packages

6.6 Services

6.7 Conference Rooms

6.8 Active Conferences

6.9 Progress Tones

6.10 Ring Tones


6.12 Storage Manager

6.13 Scheduled Tasks

6.14 Status Control Parameters

6.15 DTMF

6.16 Causes


6.16.2 User Causes


6.17 Time Parameters



6.17.3 Holidays

6.18 Assistant





Disable all new callsTick off the parameter to switch the PBX into a mode in which no new calls can be made but active calls are not forcibly terminated. Trying to set up a call, the user fails being played a defined message. This function is useful for servicing purposes.

Switch on ME modeUse this option to switch your PBX into the Mobility Extension mode, which is specifically used whenever the PBX is connected as a gateway between another PBX and various types of private or public networks. When this mode is active, all the Flash patterns and DTMF characters are sent directly to the opposite port of the PBX, which does not respond to them. This function has nothing to do with the Mobility Extension used for authorised external extensions!

Unselected as missedThis option sets the way of displaying missed calls. It refers to cases when an incoming call is routed to a group of extensions or to one user with multiple extensions and is answered by one of these extensions. If this option is not checked, missed calls are not displayed. If it is checked, missed calls are displayed at all extensions except for the one that has answered it.

Generate phone directories from usersUse this option to define the way of automatic generation of phone directories using the list of users or extensions in the Users – Phone directories – Group generatedmenu. If this option is checked off, the phone directory is filled with user names and respective internal numbers. If not, the phone directory is filled with extension names and respective numbers.

Repeat destinationsWith firmware v. 2.7.0 and higher, you can repeatedly route a call to one and the same object. This is used, for example, for new routing in a bundle if the Repeat cyclicallyparameter is enabled.

Timeout – define the time interval after which the same call can be routed to the selected object. If routed before the timeout end, the call is ignored by the PBX.

Count – define the count of routing repetitions via one object for the given call.


Count – define the count of routing repetitions via one object for the given call. When this count is completed, call routing is terminated in the next routing attempt to this object.

Note

An unduly low timeout value may result in a considerable PBX overload due to call deadlock.

Example:

Figure: View of Global Parameter Configuration Menu

Global prefixesGlobal prefixes are primarily used for and virtual ports for easier Analogue VoIPdialling (CallBacks) even to public networks from the list of missed calls. The prefix is not added where the CLI has the subtype. Assign the respective prefixes to Internalthe virtual ports using the included on the tab. Added prefix for external CLIP Basic


Unlike the frequently used identification table, this option is applied close before departure to the selected port, i.e. after pairing with the telephone directory. It would be necessary to keep different phone directories for digital, analogue and SIP virtual ports if this option were not used.

Add prefix when dialling via CTI – enable dialling prefixes also for CTI calls, i.e. calls from the Tray icon, Assistant and-or Communicator. In order to set up a call, the PBX checks the length of the called number against the Numbering plan

parameter in the Localisation menu. If the called number is longer, the lengthprefix matching the calling user group is dialled.

Add prefix also when dialling from Assistant – enable dialling prefixes also from the Assistant. The key fact for whether to enable or disable this item is whether or not the prefixes are included in the phone directories. If so, do not check this option off to avoid double prefix dialling. You can select this option only if the preceding one is enabled.

The meanings of the tab columns are as follows:

Prefix name – define the prefix name to be used in other menus for identification.

Prefix – define the prefix to precede the calling party identification.

Group of users – define the group of users to use the given prefix. If no group is selected, the prefix is valid for all groups of the PBX.

Visible in Assistant – display a prefix within the application. If it is not checked, the prefix is not available for use.

Example

Suppose a call is coming from a public network extension with the number 777123456. The call is routed through the PBX to the user Karel Furst, who belongs to user group 'Skupina 1'. His VoIP phone is registered to the SIP proxy, which has been assigned prefix PRI GTS (Figure 1) in the parameter. If the Added prefix for external CLIPnumber 777123456 is found in the phone book, the calling user name is sent to the terminal including the calling user number and the added prefix 51, i.e. 51777123456. To call this number back, the user Karel Furst can dial it directly from the list of missed calls (or received calls). The call is correctly and least-cost routed to the appropriate external port.

BillingSimple AOC – this function helps you modify the billing rercords of the PBX. It allows you to simply distribute costs of such functions as call forwarding, Mobility Extension (bidirectionally) and CallBack. You are recommended to tick off the parameter before exporting data from the PBX at the latest. This simplification applies to CDRs created by firmware versions 3.0.0 and higher. The internal subscriber numbers are crucial for billing in these versions.


RecordingRecord format .wav – select this option to store the call records in the .wav format. The default format is .alaw.

Miscellaneous

Maximum simultaneous recordings – set the maximum count of calls to be recorded at the same time.

Caution

Setting a value higher than 20 (default value) for the Maximum simultanous recordings may lead to a considerable load and malfunction of the PBX.

Maximum number of detectors for contact centres – set the count of allocated

DTMF detectors for the .2N® Contact Centre Solution


6.2 Emergency CallsThe menu helps you route emergency calls properly when the PBX is Emergency Calls in one of the pre-defined emergency modes. Of course, this setting does not solve the PBX error states. In error states, analogue CO lines and an analogue telephone connected to the corresponding port of the same card can be used, for example. If the card is not powered, these ports are disconnected and you can make PSTN calls via the card directly.

List of emergency numbers – specify all necessary emergency numbers. Enter the numbers into one row using the comma separator. The count of numbers is unlimited.

Set exceptional situations – define how the PBX should process calls other than the emergency ones. A call rejection cause and a voice message are defined for each contingency.

Licence expired – the PBX licence has expired.

Emergency mode – PBX emergency mode.

Disable calls – the Disable new calls option from the Global parameters menu has been activated. No new calls may be set up but active calls are not terminated (the PBX is waiting for the users to hang up).

Destination for emergency calls in exceptional situation – define a virtual port or bundle of ports to be used for emergency call routing.

Figure: View of Possible Emergency Call Menu Configuration


6.3 Localisation

Destination selectionIn this field enter the numbers and prefixes according to the international numbering plan. This subsequently facilitates normalisation of incoming and outgoing numbers and call routing:

Destination – choose a (country) from the list and the appropriate Localisationcountry code and access codes will be assigned automatically. The settings can be changed manually if needed.

Number – this number represents the country code within the international numbering plan. For example, the Czech Republic has number 420 and Slovakia 421.

Prefixes – this prefix represents access codes into the international telephone network. By default it is 00 and + for the GSM network.

Figure: Basic PBX Localisation Setting Menu

Local settingsLike the option, the tab helps you define the national International Local calls possibleparameters:

Number – represents the national access code (area code). For example, the town of Bratislava, Slovakia, has the area code 2.

Prefixes – represents the access codes into the national telephone network. The default value is 0.


Normalise CLIP

Normalise CLIP – cut automatically the Calling Party Number (CPN) to the shortest known format according to the CLIP routing setting. If this Localisationoption is not checked, you have to route incoming calls to the requested destinations via the CPN routers. As a matter of fact, this setting means that numbers +421XXX, 00421XXX, 0XXX and XXX are identical in terms of routing.

Number plan length – define the PBX numbering plan length. The setting affects number normalising.

The settings included in this chapter are particularly utilised by the Initial Wizard for the first system login in the on-line mode with a new database.


6.4 Licences

Licence FilesThis section provides a list of installed licence files including basic descriptions. Here you can install, uninstall or download the licence files to your computer. The field consists of several columns with the following meanings:

Serial number of CPU – shows the CPU serial number.

File – shows the absolute path to a licence file within the system data space.

State – shows the current state of a licence file within the system (e.g. , Loaded, , etc.).Not loaded Bad CPU

E1 ports – shows the count of licensed ports for ISDN PRI.

E1 channels – shows the count of licensed channels for ISDN PRI.

SIP terminals – shows the count of licensed terminals for VoIP telephones that are necessary for logging your VoIP extension to the SIP proxy.

ME – shows the count of licensed external extensions (ME – Mobility Extensions). To enable the parameter on the tab on any of the Transfer Properties – MEhierarchical levels, you need more licences. Setting this parameter to on the YESuser level needs as many licences as many extensions the user has. Setting this parameter to on the carrier level needs as many licences are there are YESextensions logged to the carriers of the selected type.

Figure: Example of Three–Licence 2N NetStar®


LicencesThis part displays a well-arranged table showing details on a selected licence file. The field consists of several columns with the following meanings:

Feature – shows the type of a licensed service, interface or object within the system.

Type – defines a licence within its type.

Licensed – shows the count of licensed channels, terminals or service accesses.

Requested – the currently requested count of channels, terminals or service accesses. The red-highlighted rows indicate a lack of licences.

If no licence is available, the PBX works in the trial mode. After the trial licence expiry (800 hours), the system is blocked and will not work until the relevant licence is installed.

Figure: View of Licence Features Table


Most Important LicencesThe survey below includes the most important licences including their function descriptions.

SIP terminal – shows the count of licensed terminals for VoIP phones. You cannot log in a VoIP extension to the SIP proxy without a terminal.

Mobility Extension user – shows the count of Mobility Extension licences (external extensions). They are necessary for enabling the parameter on Transferthe tab on some of the hierarchical levels. Setting this parameter Properties – MEto on the user level needs as many licences as there are user extensions Yes(excluding external extensions). Setting this parameter to on the carrier Yestype level needs ME licences for all extensions logged in to the carriers of this type.

CallBack user – the CallBack licence shows the count of extensions that are allowed to use and that are currently using the function.

Conference subscriber – the licence shows the highest count of conference participants during the PBX operation instead of the current count of participants for the Requested column. The function helps identify the need for licence expansion.

Conference rooms – shows the count of licensed and currently existing conference rooms. The licence is also allocated to the rooms that are currently inactive due to a time condition.

VoiceMail user – gives the count of users to whom the VoiceMail function may be assigned and the count of users who are currently using the VoiceMail function.

Modem – the modem licence shows no count. The function is either licensed or unlicensed.

Event reporter – the Event reporter licence shows no count. The function is either licensed or unlicensed.

Call recording – the count of recording users or channels is licensed. One licence is allocated to one virtual port channel or one station of an authorised user. This means that 30 licences are needed to enable recording over the whole ISDN PRI port. If, for example, your licence is limited to 10, calls via 10 channels of this port will only be recorded.


6.5 Language PackagesThe list of available language packages finds itself in the Global data – Language

menu. In addition to default packages, new languages packages can be packagesinstalled here. A language package consists of progress tones, service messages and StarPoint key telephone menus.

You can create a language package easily using any of the existing packages. Open the file (a common text file). Change the into the number Language.ini Language IDcorresponding to the required localisation. Now translate the file rows into the selected language. To add a message, record one and name it exactly as specified in the English/Czech or any other package. Finally, zip all the files into a folder (hungary.zip, e.g.).

To install a new language package, click on the right-hand mouse button and select the option in the context menu. Then choose the path to the Install language packagepacked file. To uninstall a language package, select the package and use the Uninstall

option with the right-hand mouse button. The language package language packagetable consists of four columns.

It is unnecessary to define a storage for the language package in this menu as the storage set in the menu is used automatically. Global data – Storage Manager

Figure: View of Language Package Adding Menu

Column meanings:

Name – display the name of the language package. The default packages are named after their respective languages.

Storage – define the path to the package storage within the system data space. means the directory and means the Built-in /opt/netstar Internal /data/netstar

directory. Together with the column, this parameter gives the absolute Directorypath to the storage.

Status – display the package installation status.

Directory – define the path to the package storage within the system data space. Together with the column, it gives the absolute path to the storage.Storage


6.6 Services

Service Division

The services supported by the PBX can be divided into four groups: , 2N® NetStar User

, and .Users VoiceMail Extension Others

User – this group consists of user call forwarding settings, including PIN changing, user bundle login and so on.

Users VoiceMail – this group contains services related to call forwarding to the user VoiceMail, recording, playing and deleting user VoiceMail welcome notes and other services.

Extension – this group consists of extension forwarding settings, extension ringing settings for user calls, extension bundle and carrier logins, private calls, CallBack to a extension, call takeover from a extension and so on.

Others – this group consists of all the remaining services – progress tones recording, PBX date and time setting, CallBack to a number, global setup, call assumption from a group, conference calls, connection to calls, profile activation and so on.

Service SettingThe menu displays a list of available services on the let. To Global data – Servicescreate the default list use the option in the context menu. Set the selected Defaultservice on the right-hand side of the menu. Programmable service attributes are added to each service such as progress tones, setting status messages, PIN activation, alert time and default routing destination. If a service requires the PIN, assign the PIN to the calling user to avoid service unavailability!

To activate a service, dial the service prefix into the routing table according to the called number. For the default service router refer to the in the Services Routing –

menu as shown below: Routers


w

Figure: View of Service Call Routing Menu

Or, activate a service by sending an SMS. This method can only be used for groups of services that need not be active during extension setting (e.g. call forwarding, PIN/user changing, profile activation, etc.).

Description of Selected ServicesFor some services, more parameters should be defined in addition to progress tones or messages. Below are some of them:

Private callHere set the destination type in the field to (the calling user Destination Nothingsettings are used) or (select a router).Router

Call parkingHere define the . The default value is set to 180s. The parked Maximum parking timeuser hears the Music on Hold. After the time limit, the parking place is cleared and the call returns to the extension that parked it before. The parked user hears the alert tone.


Set presenceYou can use an SMS message (not a call) to set the presence text. The SMS is routed to the text router where the text section is removed that is used for SMS routing to the service. The default code is . The rest of the text is used as a new Set presence *61presence text of the SMS sending user. If the user has an active profile at the time of sending, the presence text is assigned to the profile.

Add to conferenceUse this service to add held subscribers to a conference. Two extensions at least have to be on hold for a correct function (speech slots are used). Now use the Add to

service. The default code is**0#*. The two held extensions and the caller conferencenow join the conference.

Call to conference roomUse this service to call a conference for a defined conference room. Having dialled the service code ( by default), you are invited to enter the conference room access *2#code. Enter the code and then a 'hash'. If the calling user has the right to call together the conference room subscribers, the other users are called subsequently. For more information refer to Subs. .6.7 Conference Rooms

6.7 Conference RoomsFor conference room settings refer to the menu. Use Global data – Conference roomsthis menu to configure the conference rooms and define the authorised users. This function is subject to licence, so make sure that you have the required count of licences for operating all of your conference rooms.

Basic

Access code – is used for distinguishing your conference rooms within the service. Therefore, assign a unique access code to each conference room.

Time condition – define a time interval to limit the conference room use. If no time condition is defined (– – –), the conference room is accessible continuously.

Maximum time alert [s] – define the maximum ringing time for each of the extensions called together within a conference room. After this time interval, ringing to unanswered extensions is terminated. The default value is 180s.

Licensed – if this checkbox is ticked off, the conference room is licensed and may be used. If not, check the count of licences in the Global data – Licences menu and purchase new licences or delete unused conference rooms as necessary.


Access only for enumerated – lock the conference room and give access to selected users only. Unauthorised users are denied access to the conference room.

Unknown dials others – assign the conference-calling right to a user that is not included in the conference room.

Figure: Conference Room Settings

Tones

Welcome to conference – this tone is played to the user after the user's joining the conference called by the selected conference room (handset pick-up).

Notice on entering – this tone is played to the conference participants after joining of the user that was not dialled during the conference calling or got out of the conference and is now trying to rejoin the conference room.

Alone in conference – this tone is played to the user that remains alone in the conference room (no other extension is even ringing).

Alone with alerted – this tone is played to the user that is the first or only to answer during ringing to the conference room users. As soon as another user answers the phone, the tone is disconnected.


Assistant

Visible in Assistant – enable displaying of the particular conference room in the Assistant.

Group – with the Assistant display enabled, specify here the user group for which the conference room should be visible. To make it visible for all user groups, select (– – –).

Destination for addressesUse this parameter to define the routing destination if the address specified in the Conference subscribers block is used for conference call set-up. If the Default destination type is selected and the calling party is an address, you cannot call the specified addresses but can call the extensions and users that are dialled directly. If the calling party is a extension or user, you can call the addresses too (routing From port of the calling party is used).

Type – define the destination type for the address.

Id – define the specific destination of the selected type.

Destination for addresses (messages)Set the destination for routing bulk SMS messages if sent to the address specified in the . The SMS can only be sent to the parties to the conference Parties to conferencefrom the web application. A user may send an SMS on condition that the Assistant user is assigned the conference room rights in the Assistant and the conference room is visible in the Assistant.

Type – define the destination type for the address.

Id – define the specific destination of the selected type.

Parties to conferenceIn this configuration section, you can specify users, extensions or just telephone numbers including properties for a conference room. For this purpose, seven columns with the following meanings are available:

Destination type – select a user, extension or address. The following columns are available or not depending on your selection.

Destination – define a extension or user.

Schema – enter the Number or URI scheme for the address.


Prefix – select the required prefix from the list of prefixes defined in the Global data – Global parameters menu for the address. Use this prefix for dialling the given conference subscriber.

Number/URI – enter a specific Number of URI for the address.

Dials others – set whether the selected user has the right to call a conference for the selected conference room.

Is dialled – set whether the selected user shall be dialled or not during conference set-up.

Mute – set only listening-in for the user. The other parties to the conference do not hear the user.

Destroy when created – enable the right to terminate a conference for the user that called the conference or was the first to call the conference room.

Active usersThis tab includes an on-line list of all active conference participants. The extension name (if available in the telephone directory), scheme (Number/URI), number type (national, internal, ...) and the number are defined here for each of them. You can mute/unmute each user in the Mute column. Such setting is applicable until the next change or call end.

6.8 Active ConferencesThis menu displays all currently active conferences in the PBX. You cannot configure the parameters except for the Mute option, which helps you mute/unmute a conference participant immediately.

The active conference rooms and conferences are displayed to the left. The conference name always includes the name of user who called the conference. Select an active conference to display its participants.

Figure: List of Active Conferences


6.9 Progress Tones

IntroductionProgress is a general name for all tones and announcements injected into the speech channel by the PBX. When a new database has been created, the PBX provides a set of default progress tones depending on the language packages installed. The basic set can be extended to include own (user recorded) files and tones, or external audio inputs (e.g. mp3 player) can be connected. The menu is logically divided into tabs.

Figure: View of Progress Tones Menu Tabs

Progress ListThe progress tones represent the highest level of tone and message processing. They are ready-made tones and announcements that are played back to the user by the PBX. Each progress tone has to contain one source at least.

Progress listThe Progress list tab displays all progress tones available in your PBX including those created by the user. The following functions can be used through the context menu:

Add – add a new progress tone.

Rename – rename a selected progress.

Delete – delete a selected progress.

Delete all – delete all progresses.

Add default progresses – update the default progress set preserving the changes made in other default and user progress tones.

Restore default progresses – restore the default set of progress tones without changing or deleting the user created progress tones.

Information about progress

Name – the parameter shows the name of a selected progress tone and cannot be configured in this section.

Number – is the progress tone number to be used for user progress tone recording, playing and deleting services.


Allow progress sharing – enable sharing of a selected progress tone by multiple users with the aim to save the PBX internal sources (players) during increased traffic. Since the progress tone is played back from the current position in this mode, this function is mostly used for the progress tones that need not necessarily be played back from the beginning (e.g. Music on Hold).

Language – select the progress tone language version.

Play – play back a selected progress tone.

Stop – stop playing a selected progress tone.

Progress configuration

Action – here choose one of the listed commands to define the meaning of the row.

Repeat – set the count of repetitions from the last Repeat command, or from the beginning of the progress tone till this moment. Set the count of repetitions in the Repetitions column. If the parameter is set to 2, the sequence is played once and then repeated twice.

Repeat from beginning – set the count of repetitions from the beginning of the progress tone till this moment. Set the count of repetitions in the Repetitions column. If the parameter is set to 2, the sequence is played once and then repeated twice.

Pause – create delays between the progress tones. Do not define the delay in the column until you have set the or Duration [ms] Play Play progressrow.

Play sound – play a selected element for the period of time as set in the column. If you set 0, the element will be played till the end.Duration [ms]

Play sequence – play other progress tones. For the progress to be played refer to the columns. The other columns are unused.Progress

Play parameter – play the parameter selected in the Priority/Sequencecolumn,/Parameter

Off – use this option for the progress tones played within a notification of a new incoming call. Fill in the column to set the tone off-time. If Durationrepetition is enabled, then the selected tone is replayed after re-connection. The is an example of this type of tone. To add this Queue Alerttone to your list, click on .Add default progresses

PrioritySequence/Parameter – the setting options depend on the value.Action

Priority – define the priority of sources for the row set by the Playcommand. If a source is unavailable at the moment, another source with a lower priority is used.

Sequence – define the progress (sequence) to be played if is set to Action.Play sequence

Parameter – define the parameter to be played if is set to Action Play .parameter

Package file/Parameter format – set the progress source file or the parameter type.


Progress – define the progress tone to be played within the Play progresscommand.

Own file – define the source file as listed on the tab to be played within Own filesthe command.Play

Tone – define the source tone as listed on the tab to be played within the Tones command.Play

Input (AUX in) – define the source audio input as listed on the tab Audio inputsto be played within the command.Play

Repetitions – define the count of repetitions for the command. If you Repeatselect 0, the progress tone is played all around. If you select 1, the sequence is played once.

Duration – define the duration for the or commands.Play Pause

To change the order of the progress rows use the arrows in the right-hand part of the menu. To add new rows to a certain position use the and Insert ahead selected Insert

options. The selection adds a record after the last one.behind selected Add

Language Pack Files

Language pack file listThis section shows all available language package files that can be used as sources for progress tones. Save these files to your local PC disk if necessary using the context menu.

Related progress listThis section gives a list of all the progress tones that use the above selected file. You can use all the context menu functions as available on the tab.Progress list

Other sectionsThe and sections are common for Information about progress Progress configurationall tabs and their parameters.


Own Files

Own filesThis section shows all the files uploaded by the user to the PBX that can be used as progress tone sources. The voice message must have the following format: wav 8kHz,

. You can also use the service as described in the User Manual for 8bit, mono, aLawrecording. The context menu of this section provides the following functions:

Add – add a record. The record is then used as a progress tone source. It has no file after creation. It has to be uploaded via the section.Own files source

Rename – rename a selected record.

Delete – delete a selected record.

Delete record, keep file – delete a selected record while keeping its uploaded file in the PBX.

Backup file to local disk – download a file of a selected record to your local disk. First select the file to be saved in the NetStar data space and then enter the name and storage on your local disk.

Own files sourcesWithin this section you can upload a file of your own assigning it to the created record in the section. The context menu of this section provides the following Own files listfunctions:

Add – upload a file with announcement for a selected record.

Add record for existing file – create a record for an existing file.

Delete – delete a file of a selected record.

Delete record, keep file – delete a selected record while keeping its uploaded file in the PBX.

Related progress listThis section displays a list of all the progress tones that use the above selected own file. You can use all the context menu functions as available on the tab.Progress list



Tones

TonesThe section displays all the tones of the PBX that can be used as progress tone Tonessources. The context menu of this section provides the following functions:

Add – add a new tone.

Rename – rename a selected tone.

Delete – delete a selected tone.

Delete all – delete all tones.

Derive – create a copy of a selected tone.

Add default tones – complete the list of default tones while preserving any changes in the existing tones.

Restore default tones – complete the list of default tones restoring the default values of all the changed tones.

Tone configurationIn this section you can configure a tone using a three-column table as follows:

Language – define the language version for each tone row. Thus, you can create different forms of a tone for different languages.

Action – set one of the following actions for each row:

425Hz – play a tone with the frequency of 425Hz. Set the tone duration in the column.Duration [ms]/Repetitions

Repeat – repeat rows from the beginning to this row. Set the count of repetitions in the column. If you select 0, the Duration [ms]/Repetitionssection is played back all around. If you select 1, the sequence is played just once.

Silent – define the delays between the function rows. When used on 425Hzthe first row, this function has no meaning.

Duration [ms]/Repeatitions – enter the duration for the , and 425Hz Silent functions or the count of repetitions for the function.Through Repeat

Through – has the same function as , yet you can hear the line sounds if Silentany.

Related progress listThis section provides a list of all the progress tones that use the above selected tone. You can use all the context menu functions as available on the tab.Progress list

Other sections



Audio Inputs

Audio inputsThe section displays all the audio inputs of the PBX that can be used as Audio Inputsprogress tone sources. The context menu of this section provides the following functions:

Add – add a new audio input.

Rename – rename a selected audio input.

Delete – delete a selected audio input.

Audio input sourcesIn this section you can assign a virtual port of the Audio/IO/Relay board to a selected Audio input. For each port define the language to be used for the input.

Related progress listThis section provides a list of all the progress tones that use the above selected audio input. You can use all the context menu functions as available on the tab.Progress list



6.10 Ring TonesTo set the ringing tones use the menu. Each ringing tone Global data – Ring patternsconsists of a ring pattern and Cornet tune. Some terminals are unable to change the ring tune and use the ring pattern only. See a list of available ring patterns on the left. You can add, remove or rename the ring patterns using the context menu. During database creation, default progress tone patterns are created that can be edited or removed as necessary. To restore the default settings without removing the tones created by you, use the option. To restore the default settings and remove all Updatethe ring tones, use the option in the context menu. To create the ring tones, Defaultuse the following parameters:

Figure: View of Ring Tone Configuration Menu

Repeat – make a tone pattern being played repeatedly. If this option is not checked, the tone pattern is used only once.

Cornet tune – assign a ring tone tune to a selected ring pattern in the StarPoint key telephones.

BRI signal – set the ring tone signalling for the ISDN terminals that support multi-tune ringing.

Delay – set the ring pattern using two columns and four rows. Set the and ON columns for each row to be used. represents the ring current time and OFF ON represents the row resumption delay. Both the parameters are set in OFF

milliseconds.




AutoClip RoutingAutoClip routing is used for routing of incoming calls and SMS messages in NetStar mainly through the carriers that do not transfer the PBX CLI. For example, an outgoing call via the GSM carrier identifies itself as a SIM card number assigned to a port, not as a calling user. For these cases, the information on outgoing calls and messages is saved into AutoClip routing tables, which help find the originally calling user and route the incoming call or SMS to this user. For more details on AutoClip routing refer to Subs. .7.7 AutoClip router

AutoClip ParametersYou can save records on outgoing calls/messages including user defined parameters into the AutoClip table with the aid of AutoClip parameters. To define the AutoClip parameters, use the menu. The menu is divided Global data – AutoClip parametersinto two parts. A list of available AutoClip parameter sets is on the left. Here add, remove or rename the sets using the context menu. On the right, you can configure the parameters of a selected AutoClip parameter set.

Figure: View of AutoClip Parameter Setting Menu Used for Saving

Name – is the name of the AutoClip parameter set.

Number – is the number of the AutoClip parameter set. It has no function in the current firmware, but is ready for later use.

Store:

Missed – store only records on unanswered calls (including rejected) in the AutoClip router.

Answered – store only records on answered calls (signalling connections) in the AutoClip router.

Both – store records on all calls made (answered, missed, rejected) in the AutoClip router.

Mark record as used:


Mark record as used:

After alerting – a record is marked as used after the originally calling user extension starts ringing (Alerting message).

After active – a record is marked as used after the originally calling user receives an incoming call (Active message).

Action after record call/message use:

None – no action is done after use and the record may be reused for the next matching call(s)/message(s) (until its validity has expired).

Restart timeout – the record validity is restarted after use and the record may be reused for the next matching call(s)/message(s) (until its validity has expired).

Delete record – the record is deleted after use.

Time [mins] – set the validity period for each record of the AutoClip router. When it is checked, the given record has an unlimited validity.

6.12 Storage ManagerFind the in the menu. This menu Storage Manager Global data – Storage managerhelps you define all storages necessary for the PBX operation and services. In addition to classical internal storages (such as DATA, NAND), you can map network disks and MMC cards, which make the usable space almost unlimited and provide access to such services as call recording, for example.

Logical StoragesLogical storages represent the basic storing units for all PBX services and functions. You can add logical storages to the PBX from a pre-defined set but cannot create logic storages of your own. Logical storages themselves have no reserved data space. Hence, you have to map one physical storage at least to each logical storage, such as the internal memory, MMC card or network disk (CIFS – Common Internet File S.).

Right-hand button context menu actions:

Expand all – unfold the logical storage tree into a view of the physical storages.

Collapse all – fold the tree structure into a view of the logical storages.

Add logic storage – select and add a pre-defined logical storage.

Add physical storage – add a physical storage to the currently selected logical storage. You can only add physical storages as created in the Physical storagessection.

Remove – remove the selected physical storage or logical storage including all of its physical storages.

Remove all – remove all logical and thus all physical storages from this section.

Default – reset the current structure of logical/physical storages to the default values. All the physical storages added are ignored and remain in the structure.


Figure: Mapped Storage Setting Options

Properties

Strategy – select how to choose physical storages for the given logical storage. A linear strategy is only available at present. The selection is active when you click on any of the logical storages. You can change priority of the physical storages by Drag&Drop function.

Linear – data are stored in a sequence starting from the first physical storage. When the first storage is full, the next physical storage in the sequence is used.

Move register of files to storage – move a register of records and VoiceMail messages to a storage in a separate file instead of the PBX database. If, however, is selected too for the given physical storage, the register will Shift files remain in the PBX database. Files (records and VoiceMail messages) are always moved to a storage with the highest priority or the first available storage without the selection located above the storage from which the files Shift filesare to be moved.

Ukázka záznamu externí evidence pro nahrávání hovor

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Ukázka záznamu externí evidence pro hlasovou poštu

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Keep files removed from register – delete the removed files from the register but


Keep files removed from register – delete the removed files from the register but keep them in the physical storage.

Delete empty directories – when the file validity expires, the respective directory will be automatically deleted too.

Mapping – the section is active when you click on any of the physical storages and includes the following parameters.

Status – this parameter displays the current status of the selected physical storage within a specific logical storage, including whether the storage is available or full.

Read only – this parameter disables file editing for the given physical storage. The files can be read only.

Shift files – set the given physical storage as temporary. The files stored here are not archived but shifted to a storage with a higher priority. If this option is enabled, the parameter cannot Move register of files to storagebe applied for the selected storage. For example, in the event of network storage failure, files are stored in the next storage in the sequence and the file register is stored in the PBX database. When the network storage becomes available again, the files and file registers are moved to it.

Caution

All database records created before version 4.1.x are deleted when the option is selected. If the Shift files Move register of

parameter is disabled, files are deleted from files to storagethe storage too.

Path – this parameter defines the path to a directory within the logical storage mapped. This accelerates physical storage mapping as you just search the selected directory instead of the entire structure. Make sure that the selected directory really exists in the specified location.

Usable space – the parameter specifies some space in the currently selected physical storage to be occupied by the given logical storage. It is because one physical storage may be used by multiple logical storages (by defining the used directory, e.g.), each of which may be assigned a different data space size. Set to leave the space unlimited on this level.-1

Used – this non-editable parameter informs of the current occupation of the space reserved for the selected physical storage.


Caution

Where a physical storage is used for multiple logical storages, create subdirectories and define the path on the Setting tab to apply the quotas necessary for the given physical storage.

Example:

Suppose network storage is used for both recording calls and server02 VoiceMail. While setting quotas, create a call recording folder ( ) and voicerecVoiceMail folder ( ) in the network storage and define the paths for voicemail

and in the configuration tool (see the figure above).voicerec voicemail

QuotasTable columns:

Subject type – this column displays the subject type to which the row relates. Choose User, Group, Virtual port, or Virtual port type.

Subject – defines a subject of the above selected type.

Usage quota – defines the usable space for the given subject within the logical storage data space (for all its physical storages).

Item size – defines the maximum file size for the given subject.

Number of items – defines the maximum count of files to be stored for the given subject within the logical storage data space.

Item life [s] – sets the time for which the file stored for the given subject shall be kept in the logical storage data space.

Delete oldest achieving quota – enables deletion of the oldest user files within the logical storage data space (when the file retaining time expires).


List of filesIf you are on the logical storage level, you can see all files contained in the corresponding physical storages. If you select a physical storage, you can only see the files saved in the particular physical storage.

Right-hand button context menu actions:

Re-read view – you can refresh the current file list within the logical storage data space.

Remove – use this option to remove the selected file.

Rename – use this option to rename the selected file.

Create directory – use this option to create a directory within the data space of the currently used physical storage.

Import file – add a file from a PC to the currently used physical storage.

Export file – load a file from the logical storage data space into a PC.

Table columns:

Name – file name.

Size – file size.

Changed – last file revision date and time.

Attributes – additional information on the file to be used by the system.

List of locked filesRight-hand button context menu actions:

Save – load a file from a storage to a PC.

Listen – play the selected file.

Remove – remove the selected file from the storage.

Remove all – delete all files from the selected storage.

Table columns:

Name – name of a locked file.

Created – file creation date/time.

Validity – file locking time, or file lifetime in the physical storage. The file will be deleted when this time expires.

Size – file size.

Subject type – subject type to which the file belongs.

Subject – file owner.

Media – type of the memory connected (MMC, USB, ...). The function has not .been implemented yet


Physical StoragesRight-hand button context menu actions:

Add – add a row for your own physical storage mapping.

Rename – rename the selected physical storage.

Remove – remove the selected physical storage.

Remove all – remove all physical storages defined.

Default – reset the default physical storages for the PBX.

Table columns:

Name – name of the physical storage.

Type – basic type of the physical storage. Choose , , or Built-in Network.Removable

Access point – define the path to the storage.

Removable or built-in – a set of pre-defined paths to specific parts of the internal data space or the MMC card slot.

Network – define the path to the shared space of the network disk as for classical sharing (e.g. \\192.168.22.164/netstar_storage).

Usage quota – define the total space to be used by a physical storage for all of its PBX functions. When the limit is exceeded, the physical storage will be put out of operation.

Network type – choose either or . Used for network Microsoft Windows Nfsconnections only.

Login – set the login for connection to the shared space on the network disk. Used for network connections only.

Password – set the password for connection to the shared space on the network disk. Used for network connections only.

Connection attempt in [s] – define the intervals in which the PBX attempts to get connected to the given storage shall be made. If you set , the storage -1function will not be checked.


Detected StoragesTable columns:

Physical storage – name of the physical storage.

Type – type of the physical storage (Built-in, Network, Removable).

Access point – path to the storage as described in the Physical storages above.

Usage – amount of data saved in the physical storage.

Free size – current free space in the physical storage.

Total size – total amount of data space in the physical storage.

State – current state of the physical storage.

Root path – root directory of the physical storage.

6.13 Scheduled TasksThis menu helps you schedule your database backup, PBX restart, UMTS board restart or KeepAlive sending for PBX operation information. Click in the context menu to Add add an event and select the event type, name and repetition mode in the subsequent window.

Database backupThis option helps you schedule your database backup intervals easily, especially in the case of incidental data loss or configuration changes. The database is stored in the physical storage defined in the Storage manager in preset intervals. The storage can be a MMC card or a shared directory on a network disk. The database is stored with a timestamp designating the storing date/time and current firmware version for later use.


LDAP synchronisationThis option sets synchronisation with the LDAP server specified in the Network –

menu. Synchronise with a selected LDAP Service settings – Directory service (LDAP) server or all servers at one time.

PBX restartThe deferred PBX restart is useful, for example, when it is impossible to restart the PBX immediately after firmware upgrade as it is used by the users.

UMTS boards restartSchedule the UMTS card restart via this option.

PBX KeepAliveThe KeepAlive messages help monitor the PBX operation. The KeepAlive settings include the Event reporter, which helps send the KeepAlive messages. Refer to the

menu for the Event reporter settings: set the Network – Supervision services PBX event type for the object created. keepalive The Event reporter function is subject to

licence.

The following options are available in the Scheduler menu:

Type – select the event to be scheduled.

Name – set the name of the event to be scheduled.

Schedule task – select a repetition frequency for the event to be scheduled.

Not scheduled – the event is not scheduled in this mode.

Daily – specify in how many days the event shall be repeated.

Weekly – specify in how many weeks and on which days the event shall be repeated.

Monthly – specify on which days or days of which weeks of the selected month(s) the event shall be repeated.

Once only – select a date on which the event shall be executed.

At time – define the time at which the event shall be executed on the selected day. You are advised to select a time value off the working hours (in which the PBX is heavily loaded with user calls) except for the KeepAlive messages.


Figure: Daily Database BackUp Configuration Example

Do action immediately – make the selected action be executed immediately upon the button press.

6.14 Status Control ParametersThe menu helps you define various states of Global data – Status Control parametersthe Status Control objects, which can be created in the Routing – Routing objects –

menu. Figure 1 shows an overview of programmable Status Control objects parameters. Use the context menu under your right mouse button to add and remove items.


Figure: View of Status Control Parameters

Meaning of the columns:

State – define the Status Control object state: and .Unready, Ready, OK Error

State name – use this optional parameter to facilitate state identification in the case of a high number of states.

Message text – enter a text to be compared with the message text sent to the Status Control object. If a match is found, the appropriate state is selected for the Status Control object.

Called number – enter a number to be compared with the called number. If a match is found, the appropriate state is selected for the Status Control object.

Tone after set – set the tone to be played to the calling subscriber in the case of a Status Control object state change. If the tone is not set, the call is not answered in the Status Control object (the call does not pass to CONNECT), but the state change is made.

State colour – assign one of the preset colours to each state. When the Status Control object state changes, the colour of the respective Status Control object

will change too in the Operator menu in the application. If 2N® NetStar Assistant

one state is defined more times with different colours in the table, the colour of the first state in the sequence will be used for all identical states.

Note

You must be assigned the rights in order to Operator administration

display the Operator menu in the . Refer to the 2N® NetStar Assistant

menu for details.Users – User rights

Click on the column header to arrange the rows upwards/downwards according to the name or numerical value in the cell.


6.15 DTMFRefer to the for DTMF profile settings. Select a profile for Global parameters – DTMFDTMF detection in this menu. Click in the context menu to add the default DefaultDTMF profiles.

6.16 CausesHere is what you can find in this section:


6.16.2 User Causes


6.16.1 Cause ObjectsUse the menu to create sets of causes to be used for modifying bundle Cause Objectsparameters. The menu is divided into two parts. You can add, remove and rename objects to the left and edit the selected objects to the right. The following options are available to the right:

Name – name of the selected cause object.

Respond to – specify the object's behaviour with respect to the causes entered:

Unspecified – the object shall respond to all causes unspecified in the .Cause

Specified – the object shall only respond to the causes specified in the .Cause

Cause – select one of the pre-defined causes. In addition to common causes, you can use specific PBX causes or user causes defined in the Global data – Causes –

menu. See below for some specific causes. The following options User causes are available under the right-hand mouse button:

Add – add a row.

Remove – remove the selected row.

Remove all – remove all rows all at once.

Specific causes

Invalid licence – the cause warns that the licence is invalid.

Low credit – the cause warns that the virtual port credit has been exhausted.

Recording not ready – the cause notifies a call recording error due to inaccessibility or unavailability of the storages mapped, for example.


6.16.2 User CausesUse the menu to add user causes to be used within other objects if User Causesnecessary (Cause objects or Cause mapping tables, e.g.). The following options are available under the right-hand mouse button:

Add – add a row.

Remove – remove the selected row.

Remove all – remove all rows all at once.

The table consists of two columns with the following meanings:

Assigned Id – the columns shows the Id that is automatically assigned to this user cause and used by the PBX.

Desription of cause – the column defines the user description of the cause, which replaces the Cause ID in other menus.

6.16.3 Cause Mapping TablesUse the to specify changes in selected causes. By assigning Cause Mapping Tablescauses to different types of virtual ports you can present identical causes in a different way on the PBX interfaces. To assign a mapping table to a virtual port use the Basic tag for the particular virtual port. You can also specify in which direction the mapping table should be used. One and the same table can be used for different interfaces and both directions at the same time. Hence, an internal caue can be translated into a cause towards the ISDN, SIP or GSM interface and also in the opposite direction.

The menu has two sections. You can add, remove and rename mapping tables to the left. The right-hand section includes two parameters and the mapping table.

Name – name of the selected mapping table.

Mask stack cause – disable displaying of the original cause in the PBX trace for the whole mapping table. In that case, the trace displays .Type: None


Mapping TableThe context menu helps you add, remove and remove all rows of the table. The table consists of twelve columns and an unlimited number of rows. The sequence of rows is irrelevant unless there are two rows with an identical cause and different settings. In that case, the earlier-added row is applied (the one higher in the configuration).

Cause – choose one of the pre-defined PBX causes. Here user causes are applied.

Mask – disable displaying of the original cause in the PBX trace for a selected mapping table row.

Q.850 value – enter the particular cause value according to Q.850 to be assigned to the cause in the given row.

Valid – enable translation for an ISDN stack.

Q.850 location – define the Location value to be used in DSS1 for specification of the network or user from which the cause is coming. For the acceptable values see the table below.

Decimal value Meaning

0 User

1 Private network serving the local user

2 Public network serving the local user

3 Transit network

4 Public network serving the remote user

5 Private network serving the remote user

7 International network

10 Network beyond interworking point

Test – this option relates to column Q.850 loc and is used in the inbound direction (Stack to CP). If it is not checked off, column Q.850 loc need not match and the row is recognised according to Q.850 val. If it is checked off, both the values have to match.

Set – this option relates to column Q.850 loc and is used in the outbound direction (CP to Stack). If it is not checked off, column Q.850 val is and column Q.850 loc is not included in the outgoing message. Otherwise, the message includes both the values.

GSM type – define the GSM message type.

GSM value – enter a GSM cause value to be assigned to the cause in the given


GSM value – enter a GSM cause value to be assigned to the cause in the given row.

Valid – enable translation for a GSM stack.

SIP value – choose one of the available causes for the given row.

Valid – enable translation for a SIP stack.

6.17 Time ParametersHere is what you can find in this section:



6.17.3 Holidays

6.17.1 Date and TimeUse the menu to find the current date and time of your PBX including Date and Timethe time zone. The figure below shows a basic view of the menu. The Date and Timedate format is and time is displayed in the format.year/month/day 24–hour

Figure: View of Date and Time Setting Menu

Push the button to display a dialogue box as shown in the figure Set date and timebelow. Select a calendar item or use the arrows in this window to change the date. Type the day/year values to set the date.

To set time, type the values or use the arrows. Standard 0–23 hour and 0–59 minute/second limitations are applied.

Choose a time zone from the list of time zones.


Figure: View of PBX Date and Time Setting Dialogue

6.17.2 Time ConditionsTo define the time conditions use the Global Data – Time Parameters – Time Conditionsmenu. The menu is divided into two parts. A list of available time conditions is to the left and can be created, removed or renamed here via the context menu. To the right you can compile the time conditions. A time condition can consist of several simpler rules that are added up. You can specify, add, remove or edit the selected time condition rules in the context menu.

Figure: Basic View of Time Condition Menu

First select the time condition rule to be added or modified using the or Add Editoptions. A dialogue box as shown in the figure below is displayed for you to select the rules.


1.

2.

a.

b.

c.

d.

e.

Figure: Part of Time Condition Editing Menu

With regard to a complexity of time conditions, the following time condition setting rules have been defined:

Parameters that optionally define the absolute time limits (i.e. beginning and end) have been introduced. The time interval has to obey the limitation if applicable regardless of any other settings (including ). To Interval negationdefine the time limit, use the and checkboxes and and fields From To Date Timein the upper part of the time limit setting window.

The other fields except for define a repeat rule for each part of Interval negationthe definition. An interval is valid for a selected time point if:

Holiday is not checked or the selected time point represents any of the defined holidays;

No weekday is checked or the selected time point represents a checked weekday; and simultaneously

no day is checked or the selected time point represents a day within the specified day range;

no month is checked or the selected time point represents a month within the specified month range;

no time is checked or the selected time point represents some time within the specified time range;

If the is checked, rule 2) is inverted. The limits described in Interval negation


If the is checked, rule 2) is inverted. The limits described in Interval negationitem 1) are not affected.

Below, the parameter criteria are specified:

If defined, the and options have to contain valid day and time data. If From Toyou set the and parameters at the same time, then the value may From To Fromnot be greater than the value;To

If the holiday option is selected, no other options may be checked except for ;Interval negation

If the day option is selected, the and parameters must range between 1–From To31 (cum) and the value may not be greater than the value;From To

If the month option is selected, the and parameters have to range From Tobetween 1–12 (cum) and the value may not be greater than the value;From To

If the time option is selected, the and parameters have to From Hour To Hourrange between 0–23 (cum) and the and parameters have From Minute To Minuteto be in the range of 0–59 (cum). The composed time parameter (Hour + FromMinute) may not be greater than the composed time parameter (Hour + ToMinute).

The time conditions can be used for call routing or user profile switching.

6.17.3 HolidaysTo define holidays and important days use the Global Data – Time Parameters –

menu. The menu is divided into two parts. A list of available holidays is to the Holidaysleft and the setting options are to the right. To add a holiday, choose the option Addin the context menu. Then choose a day in the calendar to the right. You can define holidays for the current year or select a holiday that repeats periodically using the

item below the calendar. The holidays are not arranged alphabetically Valid every yearbut according to their dates. You can also load the holiday list from a predefined file via the option. You can remove and rename holidays as necessary.Update from file

Figure: View of Holiday Adding Menu Used by Time Conditions

6.18 Assistant


6.18 AssistantHere is what you can find in this section:




What is 2N ® Assistant?

The is a web application for user account supervision. The web server 2N® Assistant

for this application can be run from a PBX or an external computer. The web server version has to be the same as that of the PBX firmware. In the menu you can Assistantfind three submenus for an easy Assistant managing and active session monitoring.

Administration SettingsThe menu provides the following basic application Assistant – Administration settingssettings:

Confirm deleting – enable confirmation of record removing from the call history. If this option is checked, the user is asked for confirmation before removing a record.

Default language – select the application language from a list. Currently, the list includes three languages – Czech, English and Finnish.

Image directory – select one of the predefined image sets.

CSS style file name – set the CSS style to be used for the application.

Maximum user session time [min] – set the logout timeout for an inactive user.

Hide progress tones – hide the possibility to set progress tones in Assistant.

Figure: View of Assistant Web Server Setting Menu



6.18.2 Uživatelská relaceIn the submenu you can find the list of all active sessions. Assistant – User relationsThere are three columns in the list with the following meanings:

Username – shows identification of each user session within the database.

Session ID – shows the user that corresponds to a specific session.

Last access time – shows the last user activity time in a specific session.


7. RoutingHere is what you can find in this chapter:

7.1 Routers

7.2 External Routers

7.3 Complex Routers

7.4 Switch Routers

7.5 Routing Objects

7.5.1 Bundles

7.5.2 ACD groups



7.5.5 Ring Group

7.5.6 Ring Tables

7.5.7 Modems

7.5.8 Sets



7.5.11 CallBack




7.7 AutoClip router

7.1 Routers

RouterThe is a set of rules used for incoming call routing via the PBX. The routers are routerdefined in the menu, which consists of two windows: the list of Routing – Routersavailable switch routers to the left and the configurable router parameters to the right, The context menu in the left part of the menu provides the following options:

Add – open a router adding window and enter the router name and type. After creation, the router types are colour distinguished for convenience. Choose any of the following router types:

Called number – add a router t+hat routes calls according to the CPN.

Calling number – add a router that routes calls according to the CLI.

Called number type – add a router that routes calls according to the called


Called number type – add a router that routes calls according to the called number subtype (CPN subtype), i.e. Internal, Local, National, International or Unknown.

Calling number type – add a router that routes calls according to the calling number subtype (CLI subtype), i.e. Internal, Local, National, International or Unknown.

Call type – add a router that routes incoming calls according to the call type, i.e. voice, fax or data calls.

Port – add a router that routes calls according to the incoming carrier.

Text – add a router that routes incoming SMS messages according to the text.

Delete – delete a selected router. If the router is not empty (has some rows), you will be asked for confirmation. If you delete a router, all the associated data are deleted too.

Delete all – delete all of the created routers.

Rename – rename a selected router. If you fill in an already used name, you have to change it or abort renaming.

Add router group – add a new group of routers.

Add router subgroup – add a subgroup to the currently selected group of routers.

Move to root level – move the selected object to the highest level beyond all groups and subgroups created.

Move group content – move the group/subgroup content to another router group/subgroup.

Default – delete all the current routers and create new default routers according to the router list. These new routers are automatically filled with services, users and extensions.

Default from file – this option has a similar function as Default, but in this case you can choose a file of your own for creating new routers.

Update – update the currently used routers including settings.

Update router – fill a router with services, users or extensions. If the given records already exist in the router, they are ignored, if not, they are added to the router end.

Update from file – this option has a similar function as , but in this case Updateyou can choose a source file of your own. The existing routers are not deleted but completed with missing records.

Export to file – back up all routers including records in the .xml file format.

Export router to file – use this option to back up the currently selected router in the .xml file format.

Copy router – make a copy of the currently selected router. All you have to do is enter a new name. The router copy contains the same data as the original router (including the default destination).

Show objects routed to router – open a side window to see the list of all objects to be routed to the selected router. This function helps you check the PBX routing settings.


Expand all – open the whole structure of groups and subgroups with routers easily.

Collapse all – close the whole structure of groups and subgroups with routers easily.

Moving records using the mouse, also called , has been implemented in drag & dropthis menu for easier moving of existing routers, or router groups and subgroups.

Call RoutingCall routing is executed similarly in all router types. First, the row that matches the incoming information (CPN/CLI, CPN/CLI subtype, call type, incoming carrier or SMS text) is found and then the rule specified therein is applied. In the case of prefix congruence the following columns are applied and in the case of absolute congruence of all criteria the higher row is preferred. To change the row priority use the arrows on the right-hand side of the screen. To insert rows with a certain priority use the Insert

and options. The option is used for adding ahead selected Insert behind selected Adda record behind the currently last one (i.e. the lowest priority record). Add copy of rowand are also useful functions, with which you can Add copy of row to another router add a selected row to a router of the same type or with the same column header. Some router types also enable to change the number or SMS text used for routing. The subsections below describe all available router types and their configurable parameters.

By called numberThis router is based on called number (CPN) routing. The router consists of twelve columns with the following meanings:

Prefix – set a part or the whole of the called number. When this prefix matches the incoming CPN, this row can be used for routing. In this column you can use all digits, characters and letters , , , , which can also be DTMF dialled. *, #, + A B C DThe question mark ( ) can substitute any digit (or character), but not the whole ?number (or prefix). Therefore, to substitute all of the three-figure prefixes , xyzyou have to use three question marks, i.e. . Generally, the character is used ??? *for services like the character, which is also used for dialling end signalling.#

Digits after – this column provides details on the called number length for a row (the prefix can be followed by a different count of digits). This number sets how many digits are to be awaited after the prefix before the call is routed to another destination according to the preset rule.

"0" – no more digits are awaited.

">0" – the process waits for a given count of digits (characters).

"–" – the dash indicates an unknown length of the called number. Dialling should be terminated by adding a or by the timeout expiry.#


In the case of an 'unknown length' of the called number, the call is routed immediately upon prefix recognition and the following digits are transmitted to the destination according to the rule (generally to another router or to the public network). Otherwise, the call is not routed until the whole number has been dialled (according to the preset prefix and count of expected digits, so the number need not be complete at all). Therefore, remember to sort prefixes from the longest to the shortest ones while using the 'collision routing'.

The called number can also be changed in this router type. Having passed through the router, the call can be routed to another router of the same type where, however, it is routed according to the number modified by the preceding router. Use the following columns for CPN changing:

Remove from beginning – define the count of digits to be removed from the called number beginning.

Add to beginning – fill in the string to be added to the called number begin. Use this column only if the called number length is other than '–' (dash) in the Digits

column. Doing this use the following symbols:after

Number – means digits, letters A, B, C, D and characters *, #, +.

, – the comma means waiting for one second.

p(X) – represents the count of seconds of waiting. This instruction is Xequivalent to entering an X number of commas.

t – determines whether the preset number will be dialled after connection to a voice channel ( used), or whether dialling will be delayed before tconnection ( unused).t

Remove from end – define the count of digits to be removed from the called number end. Use this column only if the called number length is other than '–' (dash) in the column.Digits after

Add to end – fill in the string to be added to the called number end. Use this column only if the called number length is other than '–' (dash) in the Digits aftercolumn. Doing this use the following symbols:

Number – means digits, letters A, B, C, D and characters *, #, +.

, – the comma means waiting for one second.

p(X) – represents the count of seconds of waiting. This instruction is Xequivalent to entering an X number of commas.

t – determines whether the preset number will be dialled after connection to a voice channel ( used), or whether dialling will be delayed before tconnection ( unused).t

Scheme – change the called number scheme to or . The default Number URIvalue of this column is .Preserve

Subtype – select the called number subtype as , , , Internal Local National or . The default value of this column is .International Unknown Preserve


1.

2.

Examples

The instruction means that after the other party answers the call, you t1p(5)3,,*6dial digit 1, wait for five seconds, dial digit 3, wait for two seconds and, finally, dial * and digit 6.

The instruction means that digit 1 is dialled followed by a one-1,2,,3p(3)456second delay, then digit 2 is dialled followed by a two-second delay, digit 3 is dialled followed by a three-second delay and, finally, digits 4, 5 and 6 are dialled.

Destination type – set the type of destination to which an incoming call should be routed. Choose an item from the list of available PBX routing objects. There are three options in the column that need more explanation:

Default – route the incoming call directly to the next routing level (if any). It is generally used for sorting objects into sets. With the Default option, the incoming call is routed back to the superior set and the next set item is used.

Disabled – terminate the incoming call routing immediately. The calling user will hear the congestion tone.

Origin – return a modified number from the given router back to the incoming port (through which it came to the PBX).

Destination – select a destination within the above-selected destination type.

Tone – define the tone to be played to the calling user after prefix dialling in the case of overlap sending. The tone is played after dialling end only in case the called number length has not been defined and a router is the next destination.

Time condition – set a time condition for each router row. The routing rule is valid only during the time condition validity period. Time conditions help you create sophisticated routing schemes according to time. You can route a call to different destinations for the same incoming conditions (except for time).

Default destination – if no match is found in the Prefix column, the call is routed as defined in this option (located below the routing rule table):

Type – set the type of destination to which an incoming call is to be routed. Choose an item from the list of all available PBX routing objects.

Id – select the destination of the above-selected destination type.

By calling numberThis router is based on routing according to the calling number (CLI). The router consists of twelve columns with the same meanings as the case is in the By called number router. The only difference lies in that the CPN prefix is used and no instructions for delayed dialling are included. All completed changes affect the resultant CLI!


By called number subtypeThis router is based on routing according to the called number subtype (CPN subtype). The called party number subtype is the only parameter that comes into the router and cannot be changed there. The router consists of five columns with the following meanings:

Subtype – is a part of the identification to be used for call routing. You can set five subtypes:

Internal – represents an internal phone number specified by the PBX administrator.

Local – represents a private network phone number in the local format.

National – represents a public network number in the national format with prefixes.

International – represents a public network phone number in the international format with prefixes.

Unknown – an unknown number format relating to none of the above mentioned subtypes.

Destination type – this column sets the type of destination to which an incoming is routed by this rule. Choose an item from the list of all available PBX routing objects. In this column you can find three options that need more explanation:

Default – route the incoming call to the next routing level (if any). It is generally used for sorting objects into sets. With the Default option, the incoming call is routed back to the superior set and the next set item is used.

Disabled – terminate the incoming call routing immediately. The calling user will hear the congestion tone.

Origin – return a modified number from the given router back to the incoming port (through which it came to the PBX).

Destination – select a destination of the above-selected destination type.

Tone – define the tone to be played to the calling user after prefix dialling in case that a router is the next destination.


Default destination – if no match is found in the column, the call is routed Prefixas defined in this option (located below the routing rule table):

Type – set the type of destination to which an incoming call is to be routed. Choose an item from the list of all available PBX routing objects.

Id – select the destination of the above-selected destination type.


By calling number subtypeThis router is based on routing according to the calling number subtype (CLI subtype). The router consists of five columns with the same meanings as the case is with the By called number subtype. The only difference lies in that the CLI subtype is the only parameter coming into the router and cannot be changed there. The calling party number subtype changes made in this router are only used for routing and not for call identification.

By call typeThis router is based on routing according to the call type (voice, data, video, etc.). All the columns have the same meanings as the case is with the By called number subtype except for the first one. The first column defines the call type. When a preset call type is recognised, the call is routed to the preset destination.

By portThis router is based on routing according to the incoming port (the call comes into the PBX through this port). All the columns have the same meanings as the case is with the By called number subtype except for the first one. The first column defines the port. When a preset port is recognised as the incoming port, the call is routed to the preset destination.

Message RoutingThe last router type is an SMS router, which routes SMS messages according to their texts. This router can also be created in the menu and cannot be Routing – Routersused for call routing. It consists of five columns with the following meanings:

Prefix – use this column to enter a text string to be recognised at the SMS beginning. After recognition, the SMS message is routed through the PBX according to the preset rule.

Replace – edit SMS messages. You can either replace the existing text with another one or insert instructions with the following meanings:

%c – insert the sender number (CLI).

%l – insert the receiver number (CPN).

%se – erase the whole text of any length. If you leave the column Replaceempty, it is translated as 'Don't change the incoming text'.

%sr(B,E) – insert the original string omitting the first number of Bcharacters and the last E number of characters.

%ss("STRING",X,N) – find the appearance of the in the X–th STRINGincoming SMS message. From this point on, you can leave letters of the Nincoming text deleting all the others. By setting to zero, you insert the Nwhole text (from the mentioned point to the message end).


%sm(B,L) – from the character of the SMS message on, you can insert B–th characters of the original text. By setting to zero, you insert the rest of L L

the text.

Destination type – set the type of destination to which an incoming SMS message is routed by this rule. Choose one of the destinations used for SMS routing only. In this column you can find two options that need more explanation:

Default – route the incoming SMS to the next routing level (if any). It is generally used for sorting objects into sets. With the option, the Defaultincoming SMS is routed back to the superior set and the next set item is used.

Disabled – terminate the incoming SMS routing immediately. The SMS message will not be delivered!

Destination Id – this column sets a destination of the above-selected destination type.


Default destination – if no match is found with any of the preset strings, the SMS message is routed as defined in this option (located below the routing rule table):

Default type – set the type of destination to which an incoming SMS message is to be routed. Choose only one of the destinations that can be used for SMS routing.

Default Id – set a destination of the above-selected destination type.


1.

2.

7.2 External RoutersThe can be used in the following two ways:External router

For calls and SMS, where the External Routing Machine (ERM server) routing

partially replaces or complements the internal routing mechanisms of the 2N®

PBX. Based on a call/SMS routing request, a query is sent to the ERM NetStarserver. If a matching record is found in the ERM database table, the ERM server returns a response specifying a parameter for further call or SMS routing in the External router.

For , where the ERM server is used as an external phone assigning namesdirectory. Based on a CLIP-name assignment request, the PBX sends a query to the ERM server. If a matching record is found in the ERM phone directory, the ERM server returns the calling subscriber's name.

The DB connectors, used for setting the ERM server communication, are inseparable part of the external routers. Refer to for details. Network – DB connectors

The menu consists of two windows: the list of created Routing – External routersexternal routers is to the left and external router configuration to the right. The context menu displayed in the left part offers the following options:

Add – open a router adding window and enter the router name.

Delete – delete the selected router. This deletion automatically removes all database links to this object.

Delete all – delete all routers in the menu. You will be asked to confirm the action before deleting.

Rename – rename the selected router. If you enter an existing name, you will be warned and no change will be made.

Add router group – add a group of routers.

Add router subgroup – add a subgroup to the selected group of routers.

Move to root level – move the selected object to a higher level beyond all created groups and subgroups.

Move group content – move the group/subgroup content to another group/subgroup of routers.

Copy router – make a copy of the selected router. Enter a new name. The router copy contains identical records as the original one (including the default destination).

Show objects routed to router – activate a side window including listing of all objects that are routed to the selected router. This function helps you check the PBX routing settings.

Expand all – expand the whole router group/subgroup structure easily.

Collapse all – collapse the whole router group/subgroup structure easily.


RoutingCall/SMS routing via the External router is similar to standard routing. The only difference is the use of a . If the parameter value returned by the ERM Parameterserver matches the value in any of the External router rows, routing to the set destination is executed.

Figure: View of External Router Settings

DB connector – select a DB connector to be used for communication with the ERM server. The External router cannot work without a DB connector assigned.

Parameter – set a string of characters to be compared with the string returned buy the ERM server. Alphanumeric characters can be used.

Destination type – set the type of destination to which the call shall be routed. All the PBX routing objects are available (if created) plus three options in which the destination is not obvious at first sight:

Default – the call routing will jump to the next routing level if any. This option is primarily used for assigning objects to sets. Select this option to return the call from the router to the superior set and routing to the next item of the set follows.

Disabled – terminate the call routing process. The calling subscriber gets the busy tone.

Origin – return the modified number from the router to the original port.

Destination – select a destination of the above selected type.

Time condition – assign a time condition to each router row to make the row valid within the set time period only. Time conditions help you create rather sophisticated, time-dependent call routing schemes. Calls can thus be routed to different destinations at different times despite identical input conditions.

Default destination – if no match is found in the with the value Parameter returned by the ERM server, the call is routed according to the Default destination settings (below the routing rule table):

Type – set the destination type. All the PBX routing objects are available (if created).

Id – select a destination of the selected type.


1.

2.

3.


NamesThe router rows are not applied in this mode and the call is routed directly to the External router default destination.

7.3 Complex RoutersThe menu provides a complex solution to routing calls via Routing – Complex routersthe PBX. This object is subject to licence!

Functionally, the menu is divided into three sections:

Route when

Changes

Route to

The first section defines the parameters according to which a call/SMS is to be routed, the second section changes the call parameters and the third one selects the final call destination. Tick off one or more parameters in the or blocks to Route when Changes

make the settings of the first and second sections. Click to select or remove all

the parameters. Use the and arrows to set the sequence of the parameters.

Additional sections:

Hide sections – hide the highlighted marking of the , and Route when Changessections.Route to

Show comments – display the column. This column allows you to Comment enter a note to the row without affecting call/SMS routing.

Name information sending – add information on the name during call routing. This setting is used, for example, for SIP communication, where an item from the selected phone directory is entered into the field in the and Name From P-

headers.Asserted-Identity

Find name in group phonebook – select a phone directory for adding CLIP information.

Find name – select whether the calling or called subscriber's phone directory shall be used.

Insert calling extension name – add the calling extension name to the name information.

Route WhenHaving created a router for each row added, specify the input parameters for call routing. The following options are available:


Called scheme and type – route the call according to the CPN scheme (phone number/URI) and type.

Called prefix – route the call according to the called number (CPN).

Calling scheme and type – route the call according to the CLI scheme (phone number/URI) and type.

Calling prefix – route the call according to the calling number (CLI).

Port type – route the call according to the original port type.

Port – route the call according to the original port .

Group – route the call according to the group to which the call initiating user is assigned .

User – route the call according to the call initiating user.

Extension type – route the call according to the call initiating extension type.

Extension – route the call according to the call initiating extension .

Call type – route the call according to the call type .

Text – route the SMS according to the SMS text.

Facility scheme and type – route the call according to the facility number scheme and type.

Facility prefix – route the call according to the facility number (used for call billing).

Redirecting scheme and type – route the call according to the redirecting number scheme and type.

Redirecting prefix – route the call according to the redirecting number (used for call billing).

While selecting the , , , or Called p refix Calling p refix Text Facility p refix Redirecting pparameters, you can either define the prefix/text to be used for call/SMS routing refix

in the row, or click to display an auxiliary window with extended settings. Select a value from a list of preset values to set the other parameters. The extended setting window is identical for the above mentioned parameters and includes the following options:

Format – set the prefix format: digits or text strings (see below).

Test – check the prefix format for a match with the input value.

Input – enter a value to be compared with the value.Format

Result – display the result of the Input – Format comparison.

Help buttons

Arbitrary number of chars – insert a text string in the prefix format.[?]

Specific number of chars (X) – insert a text string in the prefix format, [X]where is the value.X Settings for buttons

Clear prefix – delete the content.Format


Figure: Route When Setting Window

Note

If there are more call routing parameters than one, you are recommended, for convenience, to reclick the selected parameters,

combine all the parameters into a single field Route when

and just click to modify a routing rule. The window displays all the parameters included in the section.Route when

Figure: Route When Setting Window

Advanced information – display the current settings of the section instead Route when

of in the router row.


ExampleThe prefix format is set in the figure in such a manner Route When Setting Window that the router row is applied to all CPNs with in the second position. The prefix 2 format defines that one arbitrary character must precede digit . The count of 2characters following digit is not defined.2

ChangesThe section specifies the call parameters that are to be modified before Changesrouting a call to the destination. The following options are available:

Called info format – change the called number (CPN).

Called scheme and type – change the CPN scheme (phone number/URI) and type (Internal, Local, ...).

Facility info format – change the facility number (used for call billing).

Calling info format – change the calling number (CLI).

Calling scheme and type – change the CLI scheme (phone number/URI) and type (Internal, Local, ...).

Calling attributes – change the CLI attributes (Numbering plan, Authentication and Presentation).

Text – change the text message contents.

Call type – change the call type (Voice, 3.1k Audio, ...).

Control of FACILITY – decide whether or not the facility settings on the tab shall be applied to the given router row. Choose (ignore Properties Disabled

setting), or .Enabled No change

Control of REDIRECTING – decide whether or not the redirecting settings on the tab shall be applied to the given router row . Choose (ignore Properties Disabled

setting) , or .Enabled No change

Next numbers permitted

Tone when called match

Terminate on progress – decide whether or not the Terminate call when received port settings on the tab shall be applied to the PROGRESS_IND Progress Info

given router row . Choose (ignore port setting) , (apply port Disabled Enabled setting) or for the and parameters.No change Before alerting When alerting

Facility scheme and type – change the facility number scheme and type.

Redirecting prefix – change the redirecting number (used for call billing).

Redirecting scheme and type – change the redirecting number scheme and type.

Put first to queue – set priority queue processing for a call. An incoming priority call is thus put on the first place of the active queue to a busy destination. Multiple priority calls are included in a 'priority' queue and processed one by one depending on their arrival time.


Name – change the name information during call routing. This setting is used for SIP communication, where the field is changed in the and Name From P-

headers.Asserted-Identity

Recording mark – decide whether or not the recording settings on the tab shall be applied to the given router row . Choose (ignore Properties Disabled

setting) , or .Enabled No change

Like with , some parameters have extended setting windows: Route when Called info , , , , and format Facility info format Calling info format Text Redirecting info format Name

. The other parameters are set by selecting a value from a list of preset values again, The extended window provides the following options:Changes

Figure: Changes Setting Window

Format – set the parameter changing format (of the CLI info in our case; see figure above).Changes Setting Window

Test – check whether the Format has changed as required by entering input values.


Help buttons – set the format of the parameter to be changed. Select an item (Calling, Called, ...) in the button table that shall affect the parameter to be changed (Calling info format in our case). Select a row to define which item characters are to be inserted in the parameter to be changed. Click the appropriate button to enter a string into the field, which modifies the Formatparameter into the required format. The strings can be entered sequentially, the count of strings is unlimited. If the parameter includes specific values, check Test

the resulting format of the parameter changed. Click to save the

defined format into configuration or click to delete the format.

Insert a whole – insert the whole item into the parameter to be changed.

Insert a part – insert a selected item part into the parameter to be changed. Click the button to display a window (see below) for you to specify the part to be inserted: starting from a certain character from the beginning/end, or before/after a certain occurrence of the specified character.

Figure: Format Inserting Window for Parameter to Be Changed

Remove X chars from beginning – remove the defined count of characters from the beginning and insert the rest of characters.

Remove X chars from ending – remove the defined count of characters from the end and insert the rest of characters.

From beginning to X-th character – insert all characters placed before the X-th character from the beginning.

From X-th to ending – insert all characters placed after the X-th character incl. to the end.

First X characters – insert the first X characters of the item.

Last X characters – insert the last X characters of the item.


Note

Like in the section, reclick the selected parameters and Route when

combine all the parameters into a single field for Changesconvenience.

Figure: Parameter Changing Window

Route ToThe last complex router section helps you select the final destination for a call and set the router row time validity via the following options:

Destination type – set the type of destination to which the call shall be routed. All the PBX routing objects are available (if created) plus three options in which the destination is not obvious at first sight:







Default destination – if no match is found in any router row in the , Route whenthe call is routed according to the Default destination settings (below the routing rule table):



7.4 Switch Routers

The helps you change call/SMS routing via using a service Switch router NetStar2N®called . Dial or send an SMS to the service and choose a switch router Set switch routerand one of its predefined parameters to specify the call/SMS destination. This object is subject to licence!

The menu consists of two windows: the list of created switch Routers – Switch routersrouters is to the left and the configurable switch router parameters to the right, The context menu in the left part of the menu provides the following options:

Add – open a router adding window and enter the router name.

Delete – delete the selected router. This deletion automatically removes all database links to this object.

Delete all – delete all routers in the menu. You will be asked to confirm the action before deleting.

Rename – rename the selected router. If you enter an existing name, you will be warned and no change will be made.

Add router group – add a group of routers.

Add router subgroup – add a subgroup to the selected group of routers.

Move to root level – move the selected object to a higher level beyond all created groups and subgroups.

Move group content – move the group/subgroup content to another group/subgroup of routers.

Copy router – make a copy of the selected router. Enter a new name. The router copy contains identical records as the original one (including the default destination).

Show objects routed to router – activate a side window including listing of all objects that are routed to the selected router. This function helps you check the PBX routing settings.

Expand all – expand the whole router group/subgroup structure easily.

Collapse all – collapse the whole router group/subgroup structure easily.

Call Routing


Call RoutingCall and SMS routing via the switch router is similar to standard routing. The only difference is the use of a as the routing input. Use the Parameter Set switch router service to select a switch router and its Parameter. Use the service Get switch router to identify the currently active row of the switch router. This service informs the calling subscriber of the set active row by playing the respective . You can Info tone

configure the switch router via the application too (refer to the 2N® NetStar Assistant

manual for details).2N® NetStar Assistant

Figure: View of Switch Router Settings

Router number – enter the router identifier. The number is entered into the service during router selection.

Active row – display the active parameter of the switch router.

Show comments – display the column. This column allows you to Comment

enter a note to a row without affecting call/SMS routing . The 2N® NetStar

displays comments automatically at the switch router rows.Assistant

Assistant – use this block to set the switch router with respect to the 2N®

.NetStar Assistant

Visible in Assistant – enable switch router displaying in the application. If disabled, the selected switch router is not available in the application and cannot be worked with.

Group – set a group/subgroup whose users can work with the switch router via the application. It holds true that if the given group/subgroup has more subgroups, the switch router is offered to the users of the set group/subgroup and not to the users of their subgroups.

Parameter – set a string of characters for router row identification during router setting by the service. characters can be entered only.Numeric

Destination type – set the type of destination to which the call shall be routed. All the PBX routing objects are available (if created) plus three options in which the destination is not obvious at first sight :

Default – the call routing will jump to the next routing level if any. This






Info tone – select a tone to be played to the user to identify the active row in the service.Get switch router

Caution

Make sure that the parameter is set to make the Info tone Get switch service work properly.router


Default destination – if no match is found in the column, the call is Parameter routed according to the Default destination settings (below the routing rule table):




7.5 Routing ObjectsHere is what you can find in this section:

7.5.1 Bundles

7.5.2 ACD groups



7.5.5 Ring Group

7.5.6 Ring Tables

7.5.7 Modems

7.5.8 Sets



7.5.11 CallBack



7.5.1 Bundles

BundleThe is a routing object that enables to route an incoming call to one (or all) of bundlethe objects specified in the bundle. Choosing an object within a bundle depends on the selected strategy. The fact that an routing object is busy need not necessarily lead to routing termination. The call can be routed to another routing object either upon a busy router recognition or after a timeout as preset. For bundle parameters and their usage see below.

Bundle SettingsBundles can be configured in the menu. A list of Routing – Routing objects – Bundlesavailable bundles is displayed to the left. Add, delete or rename bundles using the context menu. You can also create predefined bundles with the Default option. The parameters of the selected bundle are shown to the right. The figure below shows a possible bundle configuration.


Figure: View of Bundle Configuration Menu – Basic

The above mentioned menu consists of the following parameters:

Allocation strategy – select the way of object choosing within a selected bundle. Choose , , or .Linear Cyclic All By credit

Linear strategy – an incoming call is always routed to the first bundle row. If this object is busy or unavailable, the call is routed to the next row or terminated (as preset).

Cyclic strategy – an incoming call is routed to the bundle row that comes immediately after the one used for the previous routing to this bundle. If this object is busy or unavailable, the call is routed to the next row or terminated (as preset).

All – an incoming call is routed to all objects at the same time. Basically, the strategy substitutes the ring group function. The main difference, however, is that stations and users can login to a bundle using a service.

By credit – this strategy is intended for credit-monitored bundles with virtual ports. An incoming call is routed to the virtual port of the bundle with the currently highest count of free minutes. If there are more rows with the same count of free minutes, the sequence of rows in the bundle is respected. If a row object is busy or unavailable, the call is routed to the next row, or terminated.


Queue on bundle – enable queuing of incoming calls into bundles. Depending on the strategy selected, the queue on the bundle is active either upon the first or the second passage through the bundle. With the strategy, the queue is All forced for all destinations. With the and strategies, the destinations Linear Cyclic are dialled according to the active strategy and the queue is disabled. The first attempt to call all the bundle destinations is followed by active queue routing to all the destinations that did not return causes 21 – Call Reject or 18 – No user responding and/or the cause defined in the parameter.Cause object for queue

Access number – enter the bundle number to be used for identification in the and services.Bundle login Bundle logout

Bundle conduct

Cause object – select one of the cause objects as pre-defined in the Global data menu. These objects represent a set of causes to be – Causes – Cause objects

responded to by the bundle. When one of the cause objects has been selected, the and options are disabled automatically.Respond to busy Respond to reject

Cause object for queue – route incoming calls with queue to all the bundle destinations except for those which returned the cause defined in the selected cause object.

Respond to busy – route an incoming call to the next row in case the object is busy. This increases the successful routing rate. Here cause 17 – User Busy – is relevant only.

Respond to reject – route an incoming call to the next row if rejected by the called user. Here cause 21 – Call reject is relevant only.

Route to next row at no answer – proceed to the next bundle row in case the call is not answered within the timeout defined in the or No answer timeout [s]under causes 18 – No user responding – and 19 – No answer from user.

Let ring the last call – if this option is checked off and Route to next row no is used, the incoming call is not routed to the Default destination after answer

routing to the last unused object, but it rings at the last destination. The last unused object need not be the object from the last row of the bundle.

Repeat destinations – if this option is checked off and proceeding to the next row on busy is applied, the call is not routed to the default destination after the last bundle record, but routing starts from the first bundle record again. Firstmeans the within this incoming call. A timeout for repeating First useddestinations and acceptable count of repetitions is defined in the Global parameters for convenience and deadlock protection. If a call is routed to a destination before the timeout expires, the destination pretends to be unavailable.

Default alert tonesIn this section select variable alert tones for specific situations.

Normal – set the alert tone to be used in all situations except for the two cases mentioned below.


Queued – set the alert tone to be used for routing to a extension with an active queue. The incoming call has to be queued, otherwise the normal alert tone is used.

No-port extension – set the alert tone to be used for routing to the user with a no-carrier extension (external extension). Such user has to be assigned an external extension and another internal (active) extension at least. Otherwise, you will hear another alert tone.

Default destinationHere select the destination to be used whenever the call is rejected on the last bundle destination, the next row proceeding timeout expires, or the call cannot be delivered for any other reasons (all destinations are busy or logged out). Default destination routing depends on the settings.Bundle conduct

In addition to the above mentioned parameters, an object adding table is available. The table consists of two columns with the following meanings:

Destination type – select the type of the routing object to be used for incoming call and SMS routing. Define the extension, user, carrier, set, ring group, another bundle, ringing table and VoiceMail, or disable the selected line. Remember that a call is answered immediately when routed to the VoiceMail. Also keep in mind that if an SMS is routed to the bundle, the ringing table and ring group object rows are not applied!

Destination Id – set an object of the selected type.

Disable logout – disable user or station logout from a bundle using the Logout from bundle service. Any logout attempt is notified as an error.

Advanced settings

Send CLIP – this option serves as a quick CLIP identification table. The coming identification is changed into the set format. Use to set two ID Send asdisplaying modes: select to display the as the CLIP, but Display Number/URIstore the original calling subscriber ID in the user call history. Select to Forcechange both the CLIP phone display and the user call history record. Select

to choose Number/URI, and to set the number subtype (Unknown, Scheme TypeInternal, Local, National or International).

Force facility – refers to the called number. It is used in DSS1 messages for communication with Ericsson exchanges for billing purposes. Again, set the

(Number or URI), (Unknown, Internal, Local, National, Scheme SubtypeInternational) and (specific number or address).Number/URI

Force redirecting – refers to the called number. It is used in DSS1 messages for communication with Nokia exchanges for billing purposes. Again, set the Scheme(Number or URI), (Unknown, Internal, Local, National, International) and Subtype

(specific number or address).Number/URI

Assistant – set a bundle with respect to the Assistant user application.

Visible in Assistant – display a bundle within the application. If it is not checked, the bundle is not available for use.

Group – select a group or subgroup of users who are allowed to work with


Group – select a group or subgroup of users who are allowed to work with a selected bundle within the application. If the selected group (or subgroup) contains subgroups, the bundle is available only to the users who are assigned directly to the group (or subgroup) to which the bundle is assigned.

Accounting group – enable adding the selected group number to the accounting sentence for a selected object for later cost distribution purposes.

Figure: View of Switch Router Settings

Service Login to BundleThe services have been enhanced with the option to Station/User Login to bundlespecify the bundle position to which the station/user will be assigned. If a '0' is selected for the bundle position or no selection has been made, the station/user is placed last in the bundle (as before). Selecting a '1' means the first position, '2' means the second, '3' the third, and so on. Refer to the example below for illustration.


ExampleSuppose you want to log in a station to the third position of bundle 151. Dial the service access number *64 from the station and enter the four-digit user PIN (1111, e.g.) when requested so. Now you will be asked to dial the bundle number. Dial 151 and press * for confirmation. Then dial the required bundle position for your station, i.e. 3, and press for dialling end. The service has been completed successfully and your #station is logged in as the third in the bundle.

Note

Refer to the User Manual for details on the Login to bundle service.

7.5.2 ACD groups

ACD GroupThe is a routing object that allows incoming calls to be routed to one or all ACD groupactive specified users. User selection for ACD group routing is determined by the strategy selected. If the ACD group user is busy or not responding, the call is routed to the next user as soon as the preceding user is detected busy or the preset timeout elapses depending on the setting. If all the ACD group users are busy, the call is put in a queue. Use the VIP list or the complex router (Put first to queue) to arrange the queue. The ACD group also allows you to log in/out users. This object is subject to licence!

ACD Group SettingsUse the menu to configure the ACD groups. Routing – Routing objects – ACD groups The list of created ACD groups is to the left. Use the context menu to add, remove and rename the ACD groups. Having selected an ACD group, display the group parameters to the right.

The ACD group settings include the following options:

Allocation strategy – select the call routing strategy for the ACD group users: , or .Linear Cyclic All

Linear strategy – an incoming call is always routed to the first ACD group row. If this user is busy or unavailable, the call is routed to the next ACD group user.


Cyclic strategy – an incoming call is routed to the user that comes immediately after the last-called user (the call must pass the active state). If this user is busy or unavailable, the call is routed to the next ACD group user.

All – an incoming call is routed to all users at the same time.

Access number – set a numeric identifier for the ACD group, which is required for the ACD group log in/out services.

ACD group conductThe menu contains the following parameters:

Wait for active operator – allow the incoming call to remain in the active user queue for the time period defined in the parameter. The Maximum queuing timecall will be terminated after this timeout. The will not be Default destinationapplied in this case. Having been rejected with the CALL REJECT cause by all the ACD group users, the call is put in the queue after the last rejection.

Route to next row at no answer – enable transition to the next user in case the call is not answered within the timeout defined in the or if No-answer timeout [s]causes 18 – No user responding and 19 – No answer from user occur.

Redial interval – set the interval for repeated call attempts to the ACD group users if the call was put in the queue due to busy users or rejection with USER BUSY.

Maximum time in queue – set a time period after which the queued call is routed to the or terminated if the is ticked Default destination Wait for active operatoroff.

Default alert tonesThis menu helps you set playing of various alert tones in certain situations.

Normal – set the alert tone for all situations except for the following two.

Queued – set the alert tone for the users joining the queue.

No port extension – set the alert tone for call routing to a user with a no-port extension to which the call is routed with at least one internal extension.

Default destinationHere select the destination that will be used if none of the specified users is active or there is no user in the ACD group. It is also applied when the call is rejected (CALL REJECT) by all the ACD group users. The will not be applied of the Default destination

parameter is ticked off.Wait for active operator

2N TELEKOMUNIKACE a.s., www.2n.cz 244/6251.

LicencesDisplay the state of the ACD group licence. The ACD group can be used for call routing via the PBX only if the licence is valid.

Service login to ACD groupAdd a user to the ACD group via the PBX configuration tool only. Set the following parameters for each user:

Default – a user can log in to all the ACD groups where this option is enabled for the user.

Active – informs that the given user is logged in to the ACD group and calls are routed to it. Change the setting either via the PBX configuration tool or a service. A call is not routed to an active user only if a state other than is Onlineselected.

Use the following services to log in/out a user:

Log in to ACD group – having dialled the service, you will be asked to enter the number of the ACD group to which you want to log in (confirm the dialling end with a #).

Log out from ACD group – having dialled the service, you will be asked to enter the number of the ACD group from which you want to log out (confirm the dialling end with a #).

Log in to all ACD groups – having dialled the service, you will be logged in to all the ACD groups where is selected for you.Default

Log out from all ACD groups – having dialled the service, you will be logged out from all the ACD groups where is selected for you.Default

VIPUse the VIP tab to arrange calls in the queue via the phone directories and process VIP calls preferentially. If the calling number is found in any of the specified phone directories, the call is put on a position that corresponds to the phone book level. After a timeout, the calling subscriber passes to a higher priority level. Refer to the example and figure below for details.

Example:

Suppose that calls are coming to an ACD group. As all the ACD group users are busy, the calls are queued as follows:


1.

2.

3.

The callers whose numbers are not included in any of the VIP lists are put in the end of the queue behind the VIP list users according to their call arrival times. After the (600 seconds), the caller passes to the level of the Transition timeout

caller. After another 300 seconds, the caller moves to the queue level VIP list Cof the caller from the and so on.VIP list B

The callers from the precede the callers that are not included in any of VIP list Cthe VIP lists and follow the and users. After the VIP list B A Transition timeout (300 seconds), the caller passes to the level of the caller. After another VIP list B 120 seconds, the caller moves to the queue level of the caller from the . VIP list AAs there is no higher priority level, the calls are now processed depending on the time spent in the queue.

The callers from the precede the callers that are not included in any of VIP list Bthe VIP lists and the users and follow the users. After the VIP list C VIP list A

(120 seconds), the caller passes to the level of the Transition timeout VIP list A caller. As there is no higher priority level, the calls are now processed depending on the time spent in the queue .

Note

The VIP callers are entered into the General phone books of the PBX.

Advanced Settings





Force redirecting – refers to the called number. It is used in DSS1 messages for communication with Nokia exchanges for billing purposes. Again, set the Scheme(Number or URI), (Unknown, Internal, Local, National, International) and Subtype


Assistant – set up a ACD group with respect to the Assistant user application.

Visible in Assistant – display a ACD group within the application. If it is not checked, the ACD group is not available for use.


Group – select a group or subgroup of users who are allowed to work with a selected ACD group within the application. If the selected group (or subgroup) contains subgroups, the ACD group is available only to the users who are assigned directly to the group (or subgroup) to which the ACD group is assigned.

Accounting group – enable adding of the selected group number to the accounting sentence for a selected object for later cost distribution purposes.


DISAThe (Direct Inward System Access) routing object is used for automatic call DISAacceptance by the PBX with a subsequent DTMF transfer option and playing of the selected tone. In conjunction with suitable routers, you can create the IVR structure. This routing object is particularly suitable for GSM and CO virtual ports where you have to answer incoming calls 'Manually' to give the calling user an opportunity to influence routing (these virtual ports do not support the dial-in option).

DISA SettingTo configure the DISA routing object use the Routing – Routing objects – DISA/IVR

menu. A list of available DISA objects is displayed to the left. Add, delete or objectsrename the DISA objects using the context menu. Moreover, the following three options are available:

Default – add three basic DISA modes – DISA_DEN, DISA_NOC, DISA_ME.

Update – update the existing default DISA services.

Select one of the DISA objects to display its configuration to the right. of the StrategyDISA objects is the first parameter to be configured. Select a strategy to determine the behaviour of the whole DISA routing object: or . Each strategy Immediate Alertingis assigned different DIS object parameters.

Immediate strategyThis strategy represents a common DISA concept. When a call comes to the port, it is answered, the DTMF detector is connected and the selected progress tone is played to the calling user. The numbers to be dialled are searched in the preset router. If no digit is detected before the timeout expiry, the call is routed to the selected default destination. The DTMF detector is active only in the period between the call answer and the timeout end.


Figure: Example of DISA Object Configuration with Immediate Strategy

The menu consists of the following parameters:

Tone – choose a suitable progress tone from the list. Add progress tones and messages of your own in the menu in Subs. if desired.6.9 Progress Tones

Destination after DTMF dial – set a router to be used for call routing via the PBX after the DTMF dial.

DTMF – set whether the DTMF detector should be allocated for the DISA. The count of DTMF detectors is determined by available hardware profile. If this option is disabled, the following three parameters are unavailable for configuration:

Timeout [s] – set the DIS object waiting time for DTMF. If you set ' ', the 0whole message will be played and the call will be routed to the Default destination. In this case, do not select the in the endless message Tone parameter.

Type – set the router type for call routing.

Id – select a router of the selected type.

Default destination – set the object to which the call will be routed if no DTMF dial is detected within the timeout.

Type – set the destination type for call routing.


Alerting strategyThis strategy represents a new DISA concept. When a call comes to the port, it is immediately routed to the preset and this destination is being Alerting destinationalerted till the end of timeout. The timeout is set by the parameter. The call is Timeoutnot answered during the timeout and the calling user hears the alert tone from the


network. After the timeout, the call is answered, but only towards the calling user, who is played the predefined progress tone. The Alerting destination is still being alerted. If the DTMF option is checked, the DTMF detector is connected after the timeout and remains active until the end of routing (answer, reject, etc.). The DTMF digits are transferred into the router selected in the option.Destination after DTMF dial

Figure: Example of DISA Object Configuration with Alerting Strategy


Alerting destination – set the alerting destination parameters.

Type – set the destination type for call routing.


Timeout [s] – set the timeout after which the preset message is played. The '0' selection is not suitable for the strategy since the call Alertingwould be answered immediately, which is undesirable in this strategy.

Tone – select a message from the list of active progress tones of the PBX.

Destination after DTMF dial – set a router to be used for call routing via the PBX after the DTMF dial.

DTMF – set whether the DTMF detector shall be allocated for this DISA object.

Type – set the router type for call routing.

Id – select a router of the selected type.

What Is a Contact Centre?

The is a routing object working together with the Contact centre 2N® Contact Centre

external application. The main purpose of the application is to route incoming Solution

calls to to one or all logged users (agents) of the Contact centre 2N® NetStar


depending on the strategy selected. In case all the Contact centre agents are busy, the

call is put in a queue. The also offers detailed call 2N® Contact Centre Solution

statistics.

The application communicates with via the XML API interface and 2N® NetStar

provides a user-friendly web interface.

Configuration

The Contact centres are set automatically when the project is created in the 2N®

. The NS Admin configuration tool allows you to set the Contact Centre Solutionfollowing two parameters only:

Default destination – is applied only if the call is routed to the Contact centre project outside the working hours (as set in the external application) or if no responsible agent is logged in.

Destination at connection loss – when the – 2N® NetStar 2N® Contact Centre

connection fails, the call is routed to the set destination.Solution

Caution

Use only the application to add, rename and 2N® Contact Centre Solution

delete the contact centres (projects). Removing or renaming an object via the NS Admin configuration tool may result in a Contact centre failure or data loss.

7.5.5 Ring Group

Ring GroupThe is a routing object that is used for routing an incoming call or SMS Ring groupmessage to more destinations at the same time. When the call is answered, the other destinations stop ringing and display the Missed call message. For more information refer to the item in Subs. .Unselected as missed 6.1 Global Parameters

The ring groups are also used as user groups for taking over calls. The users who miss their calls due to absence may use the and Take over from group Take over from my

selections. For these purposes, the ring group has to contain extensions only!group


Ring Group SettingTo set the ring groups use the menu. A list of Routing – Routing objects – Ring groupsavailable ring groups is displayed to the left. Add, delete or rename the ring groups using the context menu. Moreover, the following options are available:

Default – add default ring groups. Rings groups are added to any group or subgroup that contains a user.

Update – update the currently selected default ring group.

Update all – update the contents of all default ring groups at once.

If you choose one of the ring groups, you can set its parameters on the right-hand side of the menu.


Number – is used as a ring group identification for taking over calls. If this number is not filled in, the ring group cannot be used for the Take over from

and services. Use the to group Take over from my group Default alert tonesselect specific alert tones for specific situations.

Default alert tones – in this section you can set different alert tones for specific situations.

Normal – set the alert tone that is used in all situations except for the two cases mentioned below.

Queued – set the alert tone to be used for routing to a extension with an active queue. The incoming call has to be queued, otherwise the normal alert tone is used.

No-port extension – set the alert tone for routing to a no-port extension user (external extension). An external extension and one internal extension (active) at least have to be assigned to the user. Otherwise you will hear the alert tone.

In addition to the above-mentioned parameters, an object adding table is available. The table consists of two columns with the following meanings:

Destination type – select the type of the routing object to be used for incoming call routing. Define the extension, user, virtual port, set, ring group, bundle, ringing table and such objects as DISA, VoiceMail and service, or disable the selected line. Remember that a call is answered immediately when routed to the DISA (Immediate), VoiceMail and service and thus it makes no sense to add other objects to the ring group!

Destination – select an object of the selected type.


Figure: View of Ring Group Configuration Menu – Basic

Advanced setting


to choose Number/URI, and to set the number subtype (Unknown, Scheme TypeInternal, Local, National or International) .



Force forwarding – refers to the called number. It is used in DSS1 messages for communication with Nokia exchanges for billing purposes. Again, set the Scheme(Number or URI), (Unknown, Internal, Local, National, International) and Subtype


Assistant – set the selected ring group with respect to the Assistant user application.

Visible in Assistant – display a ring group within the application. If it is not checked, the ring group is not available for use.

Group – set a group of users to be able to work with the selected ring group within the application. Assign ring groups to so-called root groups only (never to subgroups). The assigned ring group is available to all the root group and subgroup users.

Accounting group – enable adding the required group number into the accouting sentence for a selected object for later cost distribution purposes.


Figure: View of Ring Group Configuration Menu – Advanced

7.5.6 Ring Tables

Ring TableThe is a routing object used for sequential routing of incoming calls to Ring tablemultiple objects, thus combining the advantages of a bundle and a ring group. The incoming call routing obeys predefined rules, which are always searched from the beginning. If an incoming call is answered by the destination to which it has been routed, the ring table routing process is terminated.

Ring Table SettingTo set a ring table use the menu. A list of Routing – Routing objects – Ring tablesavailable ring tables is displayed to the left. Add, delete or rename the ring tables using the context menu. The configuration menu of a selected ring table is displayed to the right, providing the following parameters:

Default alert tones – define specific alert tones for specific situations.


Normal – set the alert tone to be used in all situations except for the two cases mentioned below.

Queued – set the alert tone to be used for extension routing if one of the following commands is used.

Route with queue – the selected alert tone is used regardless of the setting at the final destination.Queue

Route – the selected alert tone is used only if the parameter is Queueenabled at the final destination.

No-port extension – set the alert tone to be used for routing to a no-port extension user (external extension). Such user has to be assigned an external extension and one internal (active) extension at least. Otherwise, you will hear the alert tone.

Figure: View of Ring Table Configuration Menu – Basic

The most important part of the ring table setup is the table located in the bottom part of the menu. Use this table to define the call routing rules. For this purpose, you can combine a few commands, which can be divided into three logical groups according to function.

Routing – these commands determine the object to which an incoming call will be routed:

Route – route an incoming call to the object defined in the remaining table columns. First select an object type and then an object of the selected type. Choose the extension, user, carrier, set, ring group, bundle, another ring table, AutoClip router and also such objects as DISA, VoiceMail and service. Remember that a call is answered immediately when routed to the DISA (Immediate), VoiceMail and service and it makes no sense to add other objects to the ring group!

Route with queue – route an incoming call to the object defined in the remaining table columns. If the object is busy, the incoming call is queued regardless of the setting for the object.Queue


End of routing – these commands terminate call routing to the object to which the call was routed using the or commands.Route Route with queue

Do not route – terminate routing to an object. Be sure to terminate call routing only to the object to which the call has been routed to by the ring table. For example, you cannot terminate routing to a user extension if you have routed the call to a user.

Do not route all – terminate all active routing settings in the ring table.

Waiting – these commands are used for setting the time intervals between the routing commands and routing termination commands. To specify the time period, use the second column of the table.

Wait – set the timeout for proceeding to the next row of the table. The timeout is not applied if the previous command has routed the incoming call to a busy destination and the call has been rejected or queued. In this case, the routing proceeds immediately to the next row. If 0 is used, the PBX waits for an indefinite period of time and the next row is only used in the event of busy destination or call rejection.

Wait always – set the timeout for proceeding to the next row of the table. The incoming call is not routed to the next row before the timeout expiry. If 0 is used, the PBX does not wait and immediately proceeds to the next row (such row has no sense).

Wait with queue – set the timeout for proceeding to the next row of the table. The timeout is not used if the previous command has routed the incoming call to a busy destination and the call has been rejected (not queued). In this case, the routing proceeds immediately to the next row. If, however, the call has been queued, the routing waits for a preset timeout or busy object answer (whatever comes first) before proceeding to the next row. If '0' is used, the PBX waits for an indefinite period of time and the next row is only used in the event of busy destination or call rejection.

The last command cannot be included in any of the above-mentioned groups.

None – has the same function as an empty row (= no function).

Advanced settings






Force forwarding – refers to the called number. It is used in DSS1 messages for communication with Nokia exchanges for billing purposes. Again, set the Scheme(Number or URI), (Unknown, Internal, Local, National, International) and Subtype


Accounting group – enable adding the required group number into the accounting sentence for a selected object for later cost distribution purposes.

Figure: View of Ring Table Configuration Menu – Advanced

7.5.7 Modems

Modem ConnectionModem connection is used for remote PBX access where no TCP/IP connection is available. A modem also provides remote access to the database and enables to receive current system traces via the TraceView application. Modem access, however, is considerably limited by a low data rate and thus is not recommended for the Localisation where the TCP/IP access can be used. The current NetStar PBX firmware version supports the . The figure below shows an ISDN modem with protocol X.75example of modem configuration for remote access to the PBX.


Figure: View of PBX Remote Access Modem Configuration

Connection SettingTo enable modem connection, select the required modem (if unavailable, it is probably not installed in the PC) in the (see the figure below) and the Connection parametersnumber to be dialled for PBX communication setup. This number must then be routed to the routing object created in the menu.Routing – Routing objects – Modems

Figure: Remote Access Modem Connection Settings


Modem Setting

Trace send enabled – enable trace sending for the TraceView application via a modem. If this option is not checked, the application is connected but no system information is sent to the remote user. In this mode you can view the database only.

Peer authorisation required – enable a login dialogue request for modem connection. If this option is not checked, the connection is established without requiring the login and password settings. This option is used for connections via the NSAdmin configuration tool and the TraceView application.

7.5.8 Sets

SetThe is a routing object that is used for an easy object sequencing. For example, Setsequencing of routes with the aid of default destinations is not flexible enough, being obligatory for all incoming calls. Connecting into various parts of the string may be very tying. Sets enable you to create different sequences for different situations as necessary. In addition to routers, you can add AutoClip routers, ring groups, bundles, ring tables and other sets to the sets. Furthermore, you can add extensions, users, virtual ports, modems, DISA functions and services. Remember to include the DISA and services at the end of the structure to avoid any premature chain termination.

Having joined a set, a call is always automatically routed to the first object. To route the call back to the original set, set a row or default destination in the option. DefaultThis option is used as a signal for return to the set and proceeding to the next object (row) of the set. That is why routing to an extension mostly terminates the process. You cannot set the destination in the event of unsuccessful routing and the Defaultcall has no opportunity to return to the set.

Figure: View Set Configuration Menu


Set SettingTo configure a set use the menu. A list of available Routing – Routing objects – Setssets is displayed to the left. Add, delete and rename the sets using the context menu. The configuration of the selected set is displayed on the right-hand side of the menu. Unlike the other routing objects, the sets have no configuration parameters in the menu. The menu contains only an adding table for the objects to which incoming calls are to be routed. The table consists of four columns with the following meanings:

Destination type – select the type of object to be used for incoming call routing. Choose the routers, AutoClip routers, ring groups, bundles, ring tables, other sets, extensions, users, carriers, modems, DISAs and services. Remember that the process will be terminated when you select an object that has no opportunity to return to the set. You are recommended to add such objects to the end of the structure. Use the option to return to the higher level set Default(if you are using a set in a set).

Destination – select an object of the selected type.

Restart – this option relates to the called party number (CPN). If this number has been changed since it arrived in the PBX and there is a setting somewhere Truein the set, then the original, unchanged number is being searched for in the routers from this object on. Again, if the CPN is changed again in or behind the object and the parameter is set for the subsequent objects somewhere in Falsethe set, the call is routed according to this changed number until an object with the selection is found.True

Time condition – use the time conditions to change a set in time. You can define a different time condition for each row. The rows are then valid in the time of the preset time condition validity only.


What is Audio I/O?The ports are routing objects that cooperate with the audio ports of the Audio I/OAudio/IO/Relay board. Sounds enter the PBX or are played back through these inputs. The inputs can be used as a source of external progress tones and the outputs as a broadcast, for example.

Audio PortsThe Audio/IO/Relay board can be equipped with two or four stereo jack ports with the diameter of 3.5 mm, which represents four or eight ports (each stereo port is used as two mono ports). No port can be used as an input and output at the same time. Define the attenuation value for each port ranging from -70 to 70 dB.


The audio ports cannot be used as standard PBX virtual ports and should be assigned to a specific routing object of the Audio I/O type. Each audio port can be assigned to multiple Audio I/O routing objects.

Audio I/O SettingThe menu for the Audio I/O routing objects is divided into two parts. You can add, delete or rename the routing objects on the left and define a selected object on the right using the following parameters.

Name – display the name of a selected routing object only. It cannot be directly configured here.

Audio I/O – assign a selected Audio/IO/Relay board source to an object.

Cancelled by incoming call – is a radio function option (has not been implemented yet). If a call comes during radio playing, the radio function is terminated and the terminal starts ringing. If this option is not checked, the incoming call is rejected with the user busy cause (or queued).

Turn on tone to caller – enable playing of the below-defined tone to the calling user. The calling user should always hear the tone. It means that you cannot play a tone to the assigned source only but you can disable tone playing for both directions.

Turn on tone to Audio I/O – enable playing of the below-defined tone to a selected Audio/IO/Relay board source.

Tone – define the tone to be played to the calling party or/and to the selected Audio/IO/Relay source.

Turn off tone after a time limit – enable the tone time limit. If the time parameter is switched off, the whole tone is always played back. After the tone has been played, the calling user is connected to the assigned source for broadcasting.

Tone time [ms] – define the tone playing time. It may be shorter or longer. After this period, the calling user is connected to the assigned source for broadcasting.

Figure: View of Audio I/O Configuration Menu


Example 1 – BroadcastTo use the audio port for broadcasting set the selected port onto in the OutputBoards menu and then assign it to the selected Audio I/O routing object. The broadcast function is activated by an incoming call to this routing object. To play an announcement (e.g. We are beginning ... 5, 4, 3, 2, 1, on air...), select the message in the

parameter. When a call comes to the routing object, the selected message is Toneplayed in the selected direction (calling or both) first and then the calling user is connected to the assigned source for broadcasting.

Example 2 – External Music on Hold sourceTo use an Audio/IO/Relay board source as an external Music on Hold (or other progress tone) source, configure it as . You do not even need the Audio I/O Inputrouting object for this purpose. Add a new input to the tab in left-hand Audio inputsupper part of the menu and then assign Localisation and Global data – Progress tonesthe Audio/IO/Relay board port to it in the right-hand upper part of the menu. Subsequently, use the tab to select this Audio input as a Music on Hold Progress listinput. Now assign the progress tone to a selected user group on the Properties – Basictab in the parameter. The selected group of users will thus get music played Hold toneto the port from an mp3 player or another source instead of the Music on Hold.


What Is Binary I/O?The ports are routing objects cooperating with the binary ports on the Binary I/OAudio/IO/Relay board. Each port consists of a relay and a detector. Thus, the ports can be used both for relay switching and relay state detection. The port has only a weak current source and is not intended for switching door locks and similar equipment. If completed with an appropriate external source, however, the port can be used for this purpose too.

Binary PortsThe Audio/IO/Relay board can have four or eight binary ports. Each of them can be used in the Output (switch), Input (detector) or bi-directional mode (switch and detector). The function of each port also depends on the jumper hardware setting. For available modes refer to the Hardware Manual, Subs. . 7.5.9 Audio Inputs and OutputsThe setting contains a non-programmable parameter, which is set Pulse filtering [ms] to a fixed value of 100ms. This means that changes on the input are detected every 100ms even if they came more frequently. The parameter helps protect the PBX against overload.


The binary ports cannot be used as standard PBX virtual ports and should be assigned to a specific routing object of the Binary I/O type. Each binary port can be assigned to multiple Binary I/O routing objects.

Binary I/O Routing ObjectThe Binary I/O routing object can be set as a switch or a detector. The following parameters are common for both the modes.

Name – display the name of the selected object only. It cannot be directly configured here.

Binary I/O – assign an Audio/IO/Relay board source to a selected object.

Direction – define whether the selected routing object controls the binary port switch or detector.

Switch Setup

Switch status – this parameter displays the current status of the switch/relay (Active, Inactive, Unknown). With the option, the assigned binary port Unknownis probably configured as an input or the port or board is unavailable.

Do not pick up – this parameter helps execute the actions specified below without the call being answered (the call remains in the alerting mode). Thanks to this, for example, the relay can be activated and the call can be routed by the PBX at the same time.

Tone – set the announcement to be played to the calling user whenever a call comes to this routing object.

Action at pick up – define the action to be executed upon pick up of a call or upon SMS coming to this routing object.

None – the relay does not respond.

Connect – the relay is activated unless activated before.

Disconnect – the relay is deactivated unless deactivated before.

Connect pulse – the relay is activated for the time defined in the Pulse parameter and then re-deactivated. If activated earlier, it is only width [ms]

deactivated at the end of the pulse.

Disconnect pulse – the relay is deactivated for the time defined in the Pulse parameter and then re-activated. If deactivated earlier, it is only width [ms]

activated at the end of the pulse.

Action at hang up – define the action to be executed upon hang up of a call coming to this routing object.

None – the relay does not respond to the hang up.

Connect – the relay is activated unless activated before.

Disconnect – the relay is deactivated unless deactivated before.


Connect pulse – the relay is activated for the time defined in the Pulse parameter and then re-deactivated. If activated earlier, it is only width [ms]

deactivated at the end of the pulse.

Disconnect pulse – the relay is deactivated for the time defined in the Pulse parameter and then re-activated. If deactivated earlier, it is only width [ms]

activated at the end of the pulse.

Action after timeout/tone – define the action to be executed after the Timeoutexpiry or after .Playing whole tone

None – the routing object does not respond to the timeout expiry or the end of the played tone.

Hang up – after the timeout expiry or playing the whole tone, the call is hung up in the routing object with cause no. 16 – normal call clearing.

Call destination – after the time expiry or playing the whole tone, the call is routed as configured in the option.Destination

Play whole tone – enable playing of the whole tone independently of the preset time limit.

Destination – define the next routing destination after the time limit or playing the whole tone.

Connect by time conditions – enable the relay activation according to the selected time conditions. The relay is activated whenever one time condition at least is valid. If this option is not checked, the preset time conditions are not used.


Figure: View of Binary I/O – Switch Configuration Menu

ExampleSwitch activation/deactivation by incoming SMS

To activate the switch, route the incoming SMS using the text router to the particular binary object of the switch type where the parameter is set to Action at pick up

. The other actions are ignored. To deactivate the switch, route the SMS with a Connectdifferent text through the text router to a different binary object than that used for activation. This binary object, however, is assigned to one and the same binary source. But the parameter is set to this time.Action at pick up Disconnect

Detector Setup

Detector status – display the current status of the detector (Active, Inactive, Unknown). With the option, the assigned binary port is probably Unknownconfigured as an output or the port or board is unavailable

Tone connected – set the tone to be played to the calling user when the detector gets in the active state upon pick up. The playing mode depends on the

and parameters.Timeout Play whole tone


Tone disconnected – set the tone to be played to the calling user whenever the detector gets in the inactive state upon pick up. The playing mode depends on the and parameters.Timeout Play whole tone

Event tone enable – set the tone if is enabled. The tone is played Send eventsupon user calling to the detector if event sending is in the state. The Stopped playing mode depends on the and parameters.Timeout Play whole tone

Timeout – set the call duration. After the time limit, the call is hung up (unless the following option is checked).

Play whole tone – enable playing the whole tone independently of the preset time limit.

Send events – enable SMS sending for predefined events.

Send as user – define a user as an SMS sender. Be sure to select the SMS routing parameters for the SMS sender.

Destination for events – define the destination for sending the detector status messages. Select a user, extension or number. SMS messages to users and extensions are routed directly but those routed to an address use the From port of user routing as set in the Send as user parameter.

Figure: View of Binary I/O – Detector Configuration Menu


Messages for events

Sending events – display the current state of event sending. If such sending is enabled, the messages can be sent and the button is Enableinactive. If the sending is stopped, the messages are not sent and the

button is ready for use.Enable

Active detector state – enable a message about the active state of the detector. Within this section you can define the message text to be sent. Optionally, you can stop sending after this message by selecting the Stop

option.sending when message was sent

Inactive detector state – enable a message about the inactive state of the detector. Within this section you can also define the message text to be sent. Optionally, you can stop sending after this message by selecting the

option.Stop sending when message was sent

Detector unavailable – enable a message about an unavailable detector. Within this section you can also define the message text to be sent. Optionally, you can stop sending after this message by selecting the Stop

option.sending when message was sent

Detector ready – enable a message about a ready detector. Within this section you can also define the message text to be sent. Optionally, you can stop sending after this message by selecting the Stop sending when

option.message was sent

7.5.11 CallBack

What Is CallBack?CallBack is a function used for external PBX extensions. With the CallBack you can easily reduce costs of external extensions. The extension with the CallBack enabled only alerts the PBX or sends an SMS in the appropriate form and the PBX calls back to this extension. After answering the call, the external extension can dial through the PBX as in the case of a direct call. So you do not need an expensive fixed payment tariff for all external extensions but only for your PBX SIM cards. This function is subject to licence.

CallBack SettingFind the CallBack configuration menu in the menu. The Routing – Routing objectsmenu is divided into two parts. The left side is used for management and the right side for configuring the selected object. The context menu on the left consists of the following options:

Add – add a new object for CallBack.

Delete – delete a selected object.

Rename – rename a selected object.


Default – delete all current objects of this menu and create two default CallBack objects – one for calls and one for SMS messages.

Update – add the default CallBack objects preserving the current ones. If default objects have already been created, their parameters are set to default values.

On the right-hand side of the menu you can find the following parameters:

Ring CallBack

Name – display the name of the selected object.

CallBack delay [s] – define the delay between the CallBack recognition and execution.

Ring destination – this destination is used in the case of successful CallBack immediately after call answering by the external extension.

Ring detection time [s] – define the ringing time for incoming calls from a extension with the CallBack. If this timeout expires (the calling user does not hang up), the call is routed to Destination after timeout and behaves as a normal call and the CallBack function is not used.

Destination after timeout – this destination is used after the Ring detection timeout for another call routing. In the case of SMS CallBack, this destination is not used.

Figure: View of CallBack Configuration Menu for Calls


SMS CallBack

Name – display the name of the selected object.

CallBack delay [s] – set the delay between receving of the CallBack requesting SMS and request execution.

Delay in SMS content – omit the delay in the SMS to be sent.

Alerting destination – use the destination for routing the calls specified in the SMS.

SMS formatAn incoming SMS for the CallBack function has to be routed to the text router for CallBack object routing. The SMS text depends on the setting: if Delay in SMS content

is selected, the SMS to the CallBack function will be as follows:Yes

Called number, Delay, Calling number

If the is set to , the SMS message will be:Delay in SMS content No

Called number, Calling number

Called number – this parameter is mandatory. Calls are routed according to the d settings.Alerting estination

Delay – this parameter is optional and has the same function as , CallBack delayyet a higher priority.

Calling number – this parameter is optional and identifies the calling party if necessary. If absent, the SMS sender is used as the calling party.

Figure: View of CallBack Configuration Menu for SMS

Example 1 – Initiated by call


The external extension with an enabled and licensed CallBack function dials a PBX SIM card number. The call is routed to the CallBack object. When hearing the alert tone, the calling user can wait for the end of the . In that case, the Ring detection timeoutCallBack function is not activated and the call is automatically routed according to the

. When the calling user hangs up before the timeout expiry, Destination after timeoutthe and are activated. After the expiry, a CallBack CallBack delay CallBack delayCallBack to the external extension is established. The external extension user answers the call and can go on dialling through the PBX according to the .Alerting destination

Example 2 – Initiated by SMSThe external extension with an enabled and licensed CallBack function sends an SMS message to a PBX SIM card. The SMS is routed to the CallBack object. The SMS text may be , for example. The PBX waits for a timeout (30s) and after that 800123456,30establishes a CallBack to the external extension. The external extension user answers the call and the PBX sets up a call to the called party 800123456 using the Alerting

parameter.destination

7.5.12 Status Control ObjectsThe is a routing object used for keeping the defined state Status Control object(information) based on received information, which, for this purpose, means the called number or SMS. The state of the given Status Control object is determined by the called number or SMS. Create the Status Control objects in the Routing – Routing

menu. The Status Control parameters, which are objects – Status Control objects inseparable part of the Status Control objects, help you define the Status Control object states. Refer to the menu for the Global data – Status Control parameters Status Control parameter settings. This object is subject to licence!

You can be notified of a status change of the selected Status Control object via the Event reporter too if configured so (Error, Ok).


Figure: View of Status Control Routing Object Settings

Name – display the selected routing object name. This item cannot be directly configured.

Actual state – get information on the current state of the Status Control object.

Previous state – get information on the previous state of the Status Control object. This item is for information only and cannot be edited.

Default state – define the state to which the Status Control object will pass after the preset timeout. This option is available only if the t is Default state imeout non-zero.

Default state timeout – set the time period after which the default state is set automatically. '0' means that the automatic state change function is disabled.

Assistant – use this block to set the bundle with respect to the 2N® NetStar

.Assistant

Visible in Assistant – enable bundle displaying in the application. If disabled, the bundle is not available in the application and cannot be worked with.

Group – set a group/subgroup whose users can work with the bundle via the application. It holds true that if the given group/subgroup has more subgroups, the bundle is offered to the users of the set group/subgroup and not to the users of their subgroups.

Licences – display the Status Control object licence state. The Status Control objects can be used for call routing only if the licence is valid.


7.5.13 HTTP SMS CallBackThe routing object generates an HTTP request based on a HTTP SMS CallBackreceived SMS message. The phone number included as a parametric string in the %c SMS is sent within the HTTP request to the selected server for processing.

Note

NetStar2N® does not set up CallBacks. This function is primarily

intended for the Cisco SMS CallBack service of the Cisco Unified Contact Center Express.

Name – enter the object name and database ID.

HTTP server IP address – enter the IP address or DNS name of the server.

HTTP server port

Timeout for HTTP server answer – set the 2N® NetStar HTTP response waiting time.

HTTP URI (relative) – set the relevant source address, e.g. /httpcontact.asp?number=777123456. You can enter a parameteric string , which automatically %ccompletes the phone number included in the SMS.



What Is an Identification Table?The are used for changing the calling extension numbers. To Identification tablescreate and modify them use the menu. To view an Routing – Identification tablesidentification table, assign it to a virtual port or a virtual port type. The setup menu consists of two windows. A list of available identification tables is to the left. To configure a selected identification table, use the right-hand window. The context menu on the left side of the menu consists of the following options:

Add – add an identification table.

Delete – delete a selected identification table. If you delete an identification table, all the associate settings are removed from the database (e.g. assignment to a carrier).

Rename – rename an existing identification table.

Default – delete all current identification tables and create default identification tables. These tables have already been filled with corresponding objects (extensions, users, etc.).

Update – update a selected identification table according to the selected type. Select the type via a dialogue box. Earlier identification table records are not deleted.

Update all – add default identification tables and preserve all already existing ones.

Figure: View of Identification Table Configuration Menu


ExampleSuppose extension 1234 assigned to user Rubas Marek is calling through the virtual port with the assigned identification table from Figure 1. To change the extension identification proceed as follows:

First create number 261 301 000 of the National subtype.

Then, change the last three digits according to the original extension number – 234 as shown in row 1 to get number 261 301 234.

Finally, set the ISDN numbering plan, Screening and Presentation, which are transmitted via DSS1 signalling. Also, disable the Facility and Redirecting parameters independently of the previous routing.

Identification Table SettingIn the right-hand part of the menu, set the parameters of the identification table as selected on the left. The configuration window can be logically divided into four parts:

, , and Calling party determination New identification determination Advanced settings. The table rows are arranged according to priorities. To change a Default destination

row priority use the arrows on the right-hand side of the screen. To add rows with a certain priority use the and options. The Insert ahead selected Insert behind selected

option is used for adding a record behind the currently last one (i.e. the lowest Addpriority record).

Calling party determinationCalling party determination is performed at the beginning of each identification table row. Here define the object to which the below-selected identification rule will be assigned. For this purpose, use the following parameters:

Destination type – select a type of the calling party for the rule: , , Every Extension, , , or .Extension type User Group Virtual port Virtual port type

Destination – define a calling party of the selected type (e.g. a extension).

Scheme – specify if the calling party identification should be presented as a , or non-specified ( ).Number URI Every

Type – define the calling party number subtype before identification changing. Choose one of the , , , , and Unknown Internal Local National International Everyoptions. Use if you are not sure which number subtype is used. Use Every

only if the parameter is set to .Subtype Scheme Number

CPN prefix – ensure that one and the same extension can identify itself differently depending on the called number.


New identification determinationNew identification determination is executed in the second part of each row. For convenience, this part can has a yellow background. The identification rule sets a completely new calling party identification using five columns with the following meanings:

Scheme – define whether the calling party shall identify itself by a number or URI or use the previous identification after passing through this row.

Type – if the column is set to , choose , , , Scheme Number Unknown Internal Local or as the new CPN subtype.National International

Caution

The use of the set type depends on the settings on Keep number typethe tab for each port or port type.Basic

Number – set a number to be used for creating the new CPN identification within this row.

Replace from end – define the count of new CLI digits (as set in ) to be Numberreplaced by the original CLI digits. If the CLI is not transmitted, the resultant CLI is as set in the parameter.Number

Add to beginning – add selected digits to the beginning of the new CLI.

Time conditionYou can set a time condition in the last identification table column to define the validity time for each row. If the time condition is valid, the particular identification table row can be applied. If not, the row is ignored. This helps identify users and/or virtual ports differently for different parts of the day, week or month. You can assign the time conditions created in the subsection.Time Conditions

Advanced settingsYou can define advanced parameters for each identification table row – Numbering

, and – to be transmitted via DSS1 signalling. In addition, plan Screening Presentationyou can disable the advanced and functions, which are used in Facility Forwardingsome networks.

Numbering plan – set the used numbering plan for each table row.

Screening – set the screening information for each table row.

Presentation – set the CLI presentation restrictions for each table row.

Disable facility – disable for a selected identification table row.Facility

Disable forwarding – disable for a selected identification table row.Forwarding


Disable forwarding – disable for a selected identification table row.Forwarding

Default identificationThe lower menu is called and helps you set identification parameters for all the Defaultcalling parties that have not been found in the table. The functions of these parameters are the same as those of the yellow-highlighted parameters as described earlier.

Consider forward ed user – use this option to enable the use of the identification table for a changed CLI in the event of call forwarding. Refer to the example below for more details.

ExampleSuppose a call is coming to user B from user A. User B's calls are forwarded outside the PBX via the port to which the identification table with the Consider forwarded userparameter enabled is assigned. The identification table has records for both user A and user B, and, in this case, the record assigned to user B is used. If the above mentioned parameter were disabled, the record assigned to user A would be used.

7.7 AutoClip router

AutoClip RouterThe are used for automatic routing of incoming calls and SMS AutoClip routersmessages in case a match is found in the assigned AutoClip router. Records are added to the AutoClip routers while outgoing calls or SMS messages are passing through the carriers to which the AutoClip routers are assigned. All you need to add a record on an SMS is to send it. A record on an outgoing call can be added only if the call has been rejected or unanswered by the called party. For easier comprehension and use, examples are provided at the end of this chapter.

AutoClip Router UseTo set the AutoClip routers use the menu. The menu is Routing – AutoClip routersdivided into two parts. A list of available AutoClip routers is displayed to the left. Add, delete or rename the AutoClip routers using the context menu. Moreover, there is an

option, which enables you to add the default Autoclip router. Having selected Updatea router, you can see its record listing (the last 100 records) and also set some of its parameters in the right-hand side of the menu.

You can assign the AutoClip routers virtual ports or virtual port types on the tab. BasicAssign the AutoClip router to calls and SMS messages separately. However, one and the same AutoClip router may be assigned in either case. Each AutoClip router record (row) has a flag, identifying an outgoing call or an SMS message. Each record is stored


with a set of parameters. Some of the parameters depend on the call (CPN, CLI) and some on the AutoClip parameters assigned to the calling user or to the incoming virtual port. To set the above-mentioned AutoClip parameters use the Global data –

menu. Remember to assign the parameters to calls and SMS AutoClip parametersseparately (using the same parameter sets again). To assign the parameters to outgoing calls use the option of the tab on the user or group level. Properties RoutingTo assign the parameters to outgoing SMS messages use the Messages tab on the user or group level.

Note

If these AutoClip parameter sets are not assigned to a user (or group), the user records cannot be added to the AutoClip router!

AutoClip router settingYou can set four parameters for a selected AutoClip router:

Strategy – define the way of handling records from multiple users calling one number. This strategy refers both to record storing and subsequent record retrieving. Choose one of the following three strategies:

All – save all records to the database. If an incoming call matches more AutoClip router records, all the matching users are alerted at the same time.

Sequentially – alert all the matching users sequentially (starting from the latest record) if an incoming call matches more AutoClip router records. The next record is used for the next matching incoming call only if the previous record was marked as used and not deleted, the timeout validity was restarted or no action was made after the record use.

Last one – add the latest record to the matching user (deleting the earlier ones) if calls are made or SMS sent by multiple users and routed through the carrier to which the Last one AutoClip router is assigned. If an incoming call matches more AutoClip router records (e.g. after a router strategy change), the user matching the newest record is only alerted.

Check port – define whether or not to check the used ports. If this option is enabled, it is checked whether the incoming call came to the PBX through the same port as had been used for the outgoing call making the record. If not, the AutoClip router record will not be used.

Default destination – set the default destination to be used for incoming call routing in case no AutoClip router record match is found.

Destination for address – define the destination to be used for incoming call routing in case there is no user but only the CLI information in the Final

column.destination


AutoClip record tableThe AutoClip router table consists of eleven columns with the following meanings:

Validity – display the validity time for each record. Set the time limits in the AutoClip parameter set.

Last change with – define whether the record was created/changed with a call/SMS.

Scheme – select the CPN scheme for each record: or .Number URI

Number/URI – this column shows the called party number (CPN). This number is necessary for finding a match with the calling subscriber. Therefore, make sure that the CPN is saved in the appropriate format. Always consider specific network properties and incoming normalising if applicable.

Time [mins] – this parameter shows the validity time for each record.

Action after call use – define whether the record validity will be restarted or the record will be deleted after being used by a call.

Action after message use – define whether the record validity will be restarted or the record will be deleted after being used by an SMS .

Record is used – the parameter defines whether the record will be marked as used after passing alerting (upon alert message signalling), or after passing the active state (i.e. after being answered).

Virtual port – this column shows the port used for routing of the outgoing call that created this AutoClip router record. It is used if the option is Check portselected.

Final destination – this column shows the calling party that created this AutoClip router record. The name is displayed for a PBX user and the CLI is displayed for an external user. Any incoming call or SMS is then routed to such user.

Figure: View of AutoClip Router Configuration and Call RecordsExample 1

Suppose user is calling to the public network via a GSM port to which an AutoClip Arouter is assigned. The called user does not answer the call. A new call record is Badded to the above-mentioned AutoClip router containing the CPN, record validity


time, calling user, information on the carrier used for such call establishment and other parameters. Having found a missed call, user cannot identify the calling user Bbecause the CLI is represented by your PBX SIM card number. User tries to call back Bto that number and the call is coming to your PBX carrier that was used for the outgoing call earlier. The CLI of this incoming call matches the CLI stored in the AutoClip router. If this record is still valid, the incoming call is routed directly to user A.

Example 2

This example relates to calls that are not established by a PBX user but pass through the PBX from one port to another. In this case, the AutoClip parameters have to be assigned to the incoming port. The record added to the AutoClip router includes the CLI in the column instead of the user name. If an incoming call Final destinationmatches a table record, it is routed to the stored number as defined in the Destination

option.for address


8. UsersHere is what you can find in this chapter:

8.1 Users & Groups

8.2 User Rights

8.3 Extension Types

8.4 Extensions

8.5 Phone Directories






8.1 Users & Groups

User CreationUse the menu for user setting and group and extension Users – Users & Groupsmanagement. A list of available groups, subgroups, users and extensions is displayed to the left.

Figure: PBX User Structure from Groups to ExtensionsIn the context menu you can find the following options:

Add user – add a user to a selected basic group/subgroup.

Add extension – add a new extension to a selected user.

Copy extension – create an extension for the user with the same settings as the currently selected extension has. All the user has to do is enter a new name for the extension.


Create by wizard – initiate the automatic extension-creating wizard. With it you can import the extension list or create an extension according to preset numbers and ranges.

Add group – add a basic user group. You cannot add subgroups and users unless one basic group has been created at least (the options are unavailable).

Add subgroup – add a subgroup to a selected basic group/subgroup. The subgroups can be nested on several levels.

Move to root level – move the selected object to the highest level beyond all groups and subgroups created.

Move group content – move the content of a selected group to another group.

Set default parameters – set the default parameters for a new group (as in Set parameters to Default IN in the port type section of the Virtual Port Optionsmenu).

Delete – delete the existing basic groups, subgroups or users. It is unavailable if the basic group or subgroup contains any nested object (subgroup or user). To delete such items, delete or transfer all the objects nested therein.

Rename – rename a selected existing basic group, subgroup or user.

Move to – move users to another basic group/subgroup.

Find (F3) – initiate a searching dialogue within this menu using the preset rules. Items are searched for on all levels from groups to extensions.

Find next (F5) – use this option to enable repeated searching of the string that has been entered using the function.Find (F3)

Expand all – open the whole structure of groups and subgroups with users and stations easily.

Collapse all – close the whole structure of groups and subgroups with users and stations easily.

Moving records using the mouse ( ) has been implemented in this menu for drag & dropeasier moving of existing extensions, users, groups and subgroups.

While creating the basic groups or subgroups you are requested to set the group or subgroup name only. For user creation, however, a dialogue is displayed for you to define more parameters and even assign extensions to such user as shown in figure below.


Figure: View of User Creating Dialogue

The following part of this chapter describes the menu tabs: Users and Groups

BasicIn case a group of users is selected, two programmable parameters are displayed to the right with the following meanings:

Save messages – here enable message saving into the PBX memory. If message saving is disabled for a group, messages are not displayed on the group's Cornet ports.

Maximum number of messages – define the maximum count of messages to be saved in the PBX for a user. Whenever this limit is achieved, the messages are deleted as necessary (starting from the oldest ones).

If a user is selected, the tab includes the following additional parameters:

PIN – fill in the Personal Identification Number (PIN). This number should contain four digits and is used for access to protected PBX services (e.g. Private call). The default value is 1111.

Internal number – this number is primarily used for user identification within the PBX and represents the necessary condition for SMS messages routing.

E-mail address – set the user electronic mail address for external applications (PC operator and Application server) to get the user contacts from the exchange server, for example.


Alias – this parameter is used by the PC operator and Application server external applications. Alias in the PBX corresponds to the user name in the Active Directory. Alias and e-mail are used in the exchange server for checking user identifications and user profiles may subsequently be switched according to the calendar events. For detailed information refer to the Application server manual.

Status – select one of the pre-defined statuses to be used for user status identification by co-operating applications. The pre-defined statuses correspond to the standard statuses applied in instant messaging applications.

Presence string – fill in the text to be displayed to the user calling to one of your extensions.

Active profile – define the current active profile of the user. You can also select an item from the list of available profiles.

Automatic profile switching – enable automatic profile switching according to time conditions as defined on the tab.8.4 Extensions

If you select an extension, you will see more options. Refer to Subs. for 8.4 Extensionsall the options.

PropertiesThe tab consists of a lot of subtabs, which are described in a separate Propertieschapter for convenience. This tab is exceptional because almost all of its parameters follow the fall-down hierarchy. For the structure and all the parameters refer to S. 9.1

.Setting Properties

ProfilesUser facilitate user setting handling by changing multiple parameters in one profilesstep. Each user can use up to eight profiles (or nine if we include the no-profile setup), which feature an unlimited count of different parameters. To create a profile, use this configuration tool, a key phone or the Assistant user web application. In the context menu of this tab you can find the following options:

Add – add new user profiles. This option is disabled once the eighth profile has been created. A profile number is assigned to each profile automatically. This number is always greater by one than the current largest profile number assigned to this us er. Moreover, the dialogue helps create profiles according to the existing user profiles. The profiles are copied including all respective settings.

Delete – remove a selected profile from the database.

Rename – rename a selected profile.

The user profile configuration is divided into the three tabs:


Basic

Name – shows only the name of the selected user profile. It has an informative character only and cannot be changed here. To change it, use the Renameoption in the context menu as described above.

Number – enter a profile identifier to be used primarily for . If Profile activationyou do not fill in this field, you will not be able to use this service.

Bundle – assign a selected profile to one of the available PBX bundles. Upon activation of a profile to which a bundle has been assigned, the user is automatically added to this bundle. Upon deactivation, the user is automatically removed from this bundle.

ACD group – assign a user profile to one of the ACD groups available. When an ACD-assigned profile is activated, the user will log in automatically. When the profile is deactivated, the user will log out from the ACD group automatically again.

Caution

The user can only view the ACD groups to which it is assigned. Add the user to the ACD group via your PBX configuration tool to enable the user's automatic login through profile acivation.

Presence string – type the text to be displayed to the user calling to one of your extensions. This setting has a higher priority than the same setting on the no-profile user level. It means that if this profile is active, this text will be displayed independently of other settings.

Status – select one of the pre-defined statuses to be used for user status identification by co-operating applications if the given profile is active. The pre-defined statuses correspond to the standard statuses applied in instant messaging applications.

VoiceMailThis tab is similar to the VoiceMail tab on the user level. However, this tab does not support all parameters. It is only used for more precise settings of the user profile. The parameters of this tab have a higher priority. You can set the following:

Progress – set the progress tone to be played to the calling user in the case of call forwarding to VoiceMail.

CFNA (Forwarding at no answer) – set forwarding to VoiceMail in case the incoming call is not answered before the timeout end. Specify the timeout in the

subtab in the tab for the respective user. The default Forwarding Propertiesvalue is 30 seconds.

CFU (Forwarding unconditional) – set the unconditional forwarding to VoiceMail. It means that all incoming calls will be forwarded directly to the VoiceMail if this profile is active (unless there is a hierarchical exception).

CFEC (Forwarding on error cause or busy) – set forwarding to VoiceMail in the


CFEC (Forwarding on error cause or busy) – set forwarding to VoiceMail in the case of busy user or another error cause detection (e.g. call rejection).

PropertiesThe tab consists of a lot of subtabs, which are described in a separate Propertieschapter for convenience. This tab is exceptional because almost all of its parameters follow the fall-down hierarchy. For the structure and all the parameters refer to Chapter .Setting Properties

Profiles & Time ConditionsIn the tab assign time conditions to the user profiles Profiles & Time conditionscreated in the menu. The context menu of this tab Users – Users & Groups – Profiles has two options only:

Add – add a new row to the table. Doing this choose one of the given time conditions for this row. You can assign one time condition just once to one user. After all the available time conditions have been used, the option becomes Addunavailable until you create another time condition.

Remove – remove the table rows.

One profile may only be assigned to one time condition within the time condition validity period. However, different time conditions can be assigned to one user profile. To make the user profiles switch according to the preset time condition rules, check the option in the menu of the Automatic profile switching Users – Users & Groups Basictab.

Phone DirectoriesThe tab is located in the menu.Phone directories Users – Users & Groups

If you select one of the user groups to the left, you will see a list of phone directories assigned to this user group. In the context menu you can use the following options:

Add – add a phone book to a selected user group. Choose one of the items of the list of all available phone directories.

Delete – remove a selected phone book from a user group.

You can make use of the benefits of the hierarchical structure while assigning phone directories as described in the subsection. The phone directories Setting Propertiesassigned on the group level are also available to the users of these groups and subgroups.


If you select one of the users to the left, you will see the phone book assigned to the user to the right. The count of the phone book records is limited by the

on the subtab of the tab. The default Maximum user tel. nums. Basic Properties value is 1000 records. The context menu contains the following options:

Add – add a row to the user phone book. This option becomes unavailable when you reach the maximum count of the phone book records.

Delete – remove a selected row from a user phone book.

Delete all – remove all rows from a user phone book at once.

Export – export the current user phone book in the or format.xml csv

Import – import the user phone book in the or format.xml csv

The user phone book consists of twelve columns with the following meanings:

Name – shows the name of the user who appertains to this record. This name is shown on the calling/called user's display.

Nickname – shows the nickname of each record. It is primarily used for easier searching of the phone book.

Scheme – set the user identification scheme. Choose either or .Number URI

Prefix – set the access prefix defined in the Global data – Global parametersmenu. This prefix automatically precedes the user number in dialling from a phone directory .

Number – fill in the user or according to the column.Number URI Scheme

Ring pattern – choose a specific ring tone for each user phone book record. If the PBX accepts an incoming call with a CPN matching this record, your extension will use this ring tone.

The remaining six columns are used for forwarding incoming calls to a specific destination. The call forwarding settings in the phone directories have the highest priority of all within the PBX.

VoiceMailThe tab is used for configuring the user VoiceMail and is available in the VoiceMail

menu. The tab occurs in two forms depending on the level to Users – Users & Groupsbe configured.


VoiceMail for GroupsIf you select one of the user groups on this tab, you can change only some parameters. The settings on this level have a lower priority than those on the subgroup or user level.

Progress – choose a VoiceMail progress tone from a list.

Message – set the message text to inform the user of a new VoiceMail message. In addition to a static text, you can use dynamic strings with the following meanings:

%u – called user name;

%n – calling user name;

%c – calling user number;

%d – VoiceMail creation date and time.

Save to user – enable/disable saving messages at the user regardless of the user settings, or respecting the user settings . The selection is intended for displaying

messages on Cornets and in .2N® NetStar Assistant

SIP extensions – enable/disable resending messages to user SIP stations regardless of the user settings, or respecting the user settings (According to stations).

Email extensions – enable/disable resending messages to user email stations regardless of the user settings, or respecting the user settings (According to stations).

Mobility Extensions – enable/disable resending messages to user external stations regardless of the user settings, or respecting the user settings (According to stations).

VoiceMail for UsersIf you select one of the users, you can set more parameters than on the group level. The meanings of the parameters are as follows:

Enabled – enable VoiceMail function for a selected user. It is subject to licence and after saving the setup you have to check whether the option is Licensedticked off. If not, then you do not have enough licences for this function.

Forwarding

CFNA – Call Forwarding at No Answer – set forwarding to VoiceMail in case the incoming call is not answered within the preset period the time. To specify the time limit, use the Forwarding subtab in the Properties tab for the respective user. The default value is 30 seconds.

CFU – Call Forwarding Unconditional – set unconditional forwarding to VoiceMail. It means that all incoming calls will be forwarded directly to the VoiceMail if this profile is active (unless there is a hierarchical exception).

CFEC – Call Forwarding on Error Cause – set forwarding to VoiceMail in the case


CFEC – Call Forwarding on Error Cause – set forwarding to VoiceMail in the case of busy user or another error cause detection (e.g. call rejection).

Welcome note

Welcome note – choose a VoiceMail progress tone from a list.

Set welcome note – enable/disable recording of a VoiceMail progress tone via the VoiceMail Record welcome note (*35) service.

Messages

Maximum record length [s] – set the maximum voice message recording time. After this time limit, the incoming call will be cleared automatically.

Do not store – enable/disable saving of VoiceMail to the PBX. Voice messages are only resent to the corresponding e-mail (according to the setting).

Maximum record term [s] – set the maximum voice message storing time in the PBX. After this time limit, the voice message will be removed the moment another message is saved.

Delete oldest at no space – remove the oldest voice messages in order to get more space for new voice messages.

Maximum record count – set the maximum count of voice messages to be stored in the PBX.

Notification

Message – set the message text to inform the user of a new VoiceMail message. In addition to a static text, you can use dynamic strings with the following meanings:

%u – called user name;

%n – calling user name;

%c – calling user number;

%d – VoiceMail creation date and time.

Save to User – enable/disable saving messages at the user regardless of the user settings, or respecting the user setting s. The selection is intended for displaying

messages on Cornets and in .2N® NetStar Assistant

SIP extensions – enable/disable resending messages to user SIP stations regardless of the user settings, or respecting the user settings (According to stations).

Email extensions – enable/disable resending messages to user email stations regardless of the user settings, or respecting the user settings (According to stations).

Mobility Extensions – enable/disable resending messages to user external stations regardless of the user settings, or respecting the user settings (According to stations).


Send unless voice record had meen left – enable message sending even if the calling subscriber hangs up before recording starts. Remember to enable Send

to make this function work.to user mail

Box parameters

CallBack destination – set the destination to which the VoiceMail CallBack is routed in case the record creating user is not known but the PBX has the CLIP.

Default CallBack extension

FilesThe tab is located on the group and user levels and used for viewing files with Filescalls recorded via these objects. Refer to the Files section in Subs. 3.10 Virtual Port

for details on the overview table columns and/or context menu options.Options

AssistantThe tab can only be used on the user level. The default Assistant settings are Assistantavailable in the menu. To change a Global data – Assistant – Administration settinguser setting, create an individual user setting using the following parameters:

Application main page – define the Assistant home page for a selected user.

Default language – define the Assistant default language for a selected user.

Image directory – define the set of images to be used by the Assistant for a selected user.

CSS style filename – define the Assistant's appearance for a selected user.

Free Minutes/SMSThe tab helps you set free minutes and SMS for a selected user. The Free minutes/SMSset free minutes and SMS are only subtracted from the user account on the ports via which the calls go out of the PBX and which are assigned call billing via port ( Basictab for the given port). All the ports are such ports by default.Default OUT

Select tariff rateClick the button to display a dialogue and select one of the Set free minutes/SMStariff rates as defined in the menu. In addition, you can Accounting and tariff ratesassign here a setting to the selected user tariff rate as defined earlier for any other user or virtual port. To change the tariff rate if necessary, use the Used tariff rateoption. If you do so, you will lose all data saved on free minutes with the given tariff rate via this user. To cancel the user tariff rate, push the Cancel free minutes/SMSbutton.


Free minutes/SMS settingsOnce a tariff rate is selected, the tariff rate credit rows are displayed in this section. Click on a row to display a setting dialogue for the count of free minutes, SMS messages and other credit parameters for the given user. See the figure below for the dialogue.

The table includes columns with the following meanings:

Credit name – the credit name as defined during tariff rate creation.

Free minutes for month – display the count of free minutes per month for the given user. This count is credited to the given user at the beginning of the accounting period. If the free minute count changes within a month, the port credit is not increased until the beginning of the next accounting period unless provided otherwise in the setting dialogue.

Free minutes for this month – display the current count of free minutes to be used in this month. The value includes free minutes transferred from the previous accouting period if any.

Spent minutes – display the current count of minutes spent in the accounting period.


Free SMS for month – display the count of free SMS messages per month for the given user. This count is credited to the given user at the beginning of the accounting period. If the free SMS count changes within a month, the port credit is not increased until the beginning of the next accounting period unless provided otherwise in the setting dialogue.

Free SMS for this month – display the current count of SMS messages to be used in this month. The value includes free SMS transferred from the previous accouting period if any.

Spent SMS – display the current count of SMS sent in the accounting period.

Day of account – set the day in the month on which a new accounting period shall start. On this date, the free minute and SMS counts are increased according to the selected transfer mode. The mimimum values are set in the Free minutes for month a Free SMS for month columns. Setting means (Manually) and 0 Neversetting means .32 Every day

Mode – select the method of transfer of old free minutes into the next accounting period.

Do not transfer – no free minutes and/or SMS are transferred.

First use new – old free minutes and SMS are transferred but new ones are used first. Unused units older than one month are not transferred.

First use transferred – old free minutes and SMS are transferred and new ones are not used until these old units have been exhausted. Unused units older than one month are not transferred.

8.2 User Rights

LoginsA list of all users and logins is displayed on the left-hand side of the Users – Users

menu. The list is divided into sections according to user groups and subgroups. rightsThe user name is on the left and the respective login name, if any, on the right. You can use the following context menu options here:

Create login – create a login for a selected user. This option is active only if the user has not been assigned any login. You can choose one of the types specified below.

Change login – change the login type. The option cannot be used for Admin login. Each login is also assigned a type that defines the respective right assignment level. Choose one of the following options:

Vice Admin – is a login with all rights except for and Delegate logins.Manage own group

Super – is a login with the same rights as the Admin login.

Vice Super – is a login with the same rights as the Vice Admin login.


Manager – is a login designed for the manager of all user logins in a group. Hence, the and rights are assigned to Delegate logins Manage own groupthe manager.

Vice Manager – is a login to be used by the user.Manage own group

User – create a general user login.

Change password – change the password of a selected login.

Remove login – delete a selected login. The Admin login cannot be deleted.

Generate missing logins – automatically create logins for all the users who have not been assigned one. Such logins are of the user type, have no password and their names are respective user names (without spaces and diacritic marks, with small letters).

Figure: View of Logins According to Groups

BasicAfter selecting a user, a list of all the users of the respective group including logins and rights is displayed on the right-hand side of the tab. This view is useful for Basicsetting similar rights in the user group. The table of rights is divided into sections with the following meanings.

Basic

Disable – disable a login for a period of time without deleting it.

Must change password – set automatic advice of a password change upon access to the Assistant application.

Tab directly

Read – enable reading of the database via the configuration tool.

Write – enable writing into the database via the configuration tool.

Database

Write – save completed changes into the database.

Trace

See – display the tab in the tool.Trace

Enable – enable trace downloading from the PBX.


Statistics

See – view statistic data through the tab in the tool. Statistics

AssistantAfter selecting a user, a list of all the users in the respective group including logins and rights is displayed on the right-hand side of the tab. This view is useful for Assistantsetting similar rights in the user group. The table of rights is divided into sections with the following meanings:

User management – view and change the settings of other users.

Telephone directory management – view and change the directories of other users.

Call history management – view the call history of other users.

Telephone management – view and change the telephone settings of other users.

Extension management – view and change the extension settings of other users.

Global configuration management – view and change the global configuration settings.

Operator management – view and manage the operator settings.

Alarm management – view and change the alarm settings.

SMS management – view and manage the SMS messages of other users.

Conference room management – view and manage the conference rooms.

Hotel view – view and manage alarm clocks and emergency alarms in a hotel structure.

Recorded calls – view and manage recorded calls of the user.

ACD groups / Bundles – view and manage the ACD groups and bundles.

8.3 Extension Types

Extension Type CreationThis tab gets displayed whenever you click on the menu. Use Users – Extension Typesthe extension types for easier setting of groups of extension. A list of available extension types is displayed to the left and you can set a selected extension type to the right. On the left, you can use the context menu with the following options:

Add – add an extension type.

Delete – delete a selected extension type.

Rename – rename a selected extension type.

Copy extension type – create an extension type with the same settings as the currently selected extension type has.

Extension Type Properties


Extension Type PropertiesThe tab consists of a number of subtabs, which are described in a separate Propertieschapter for convenience. This tab is exceptional because almost all of its parameters follow the fall-down hierarchy. For the structure and all the parameters refer to S. 9.1

.Setting Properties

8.4 Extensions

Extension CreationClick the menu to display this tab. A list of available extensions is Users – Extensionsto the left and settings for the selected extension to the right. The context menu to the left includes the following options:

Add – add an extension. After clicking this option you will see a dialogue box as shown in Figure 1. First define the extension name. If you choose an already existing name, the extension will not be created and you will be warned. Assign each extension to a specific user in this dialogue box too. Furthermore, fill in the

and, if you are creating an external extension, fill in the extension number too (for call routing within other networks) and, if applicable, Routing number

check the option to resend all incoming SMS messages to the Resend SMSexternal extension.

Figure: Extension Creating Dialogue

Delete – delete an extension.

Rename – rename an extension. If you fill in an already existing name, you will be warned.

Copy extension – create an extension for the selected user with the same settings as the currently selected extension has. All the user has to do is enter a new extension name.

Find – search extensive corporate databases for an extension by its name or


Find – search extensive corporate databases for an extension by its name or number. The name or number to be searched need not be complete (may be a part of the string only).

Find next – enable repeated searching of the string that has been entered in the function. This option is unavailable until you fill in the string to be searched Find

by the function.Find

Basic SettingsIf you select an extension on the right-hand side of the screen, three tabs will get displayed to the right: , and . The tab contains the Basic Properties Profiles Basicfollowing parameters:

Object – display the object type.

Name – display the name of the selected station.

Station type – define the station type. The following options are available:

Normal – a normal internal station.

SIP – a SIP station. It should be assigned to a terminal on the SIP Proxy.

E-mail – an e-mail station. Not intended for calling.

External – a Mobility Extension station.

Scheme – define the station identification scheme. Choose either a telephone number or URI.

Prefix – choose one of the prefixes defined in the menu. This Global parametersprefix partly substitutes the number subtype and facilitates CallBacks.

Number/URI – define the station identification. Enter a number, e-mail address, or URI. The function of the parameter depends on the Station type setting. When an external station is identified, the originally dialled number is changed and the call is routed with this number via the defined destination.

User – display the name of the current user. Use this option to assign an extension to another user too.

Type – assign an extension to a specific extension type. It can facilitate setting of the common parameters for a group of extensions (e.g. outgoing routing via a GSM bundle for all external extensions).

Ring group – select the ring group in which you may take over calls from the members of the group without being a member of the group (default dial *#).

Active – activate/deactivate a selected extension. A deactivated extension becomes unreachable for other extensions (incoming calls are rejected) but is able to establish outgoing calls.

Do not ring at call to user – route a call to this extension in case it is routed to the user. If it is checked, only the calls routed directly to this extension alert the extension.

Resend SMS – enable/disable SMS resending. If this option is checked, all the SMS messages delivered to the user are resent to this external extension.


Enable CallBack object – enable the CallBack function for a selected extension. The function is subject to licence and so make sure that the option has Licensedbeen selected after data saving. If not, check your licence in the Global data –

menu. Licences

Others

Virtual port – this parameter shows the port to which the extension is currently assigned. The parameter has an informative character only and cannot be changed in this menu.

Protocol – this parameter defines the communication protocol to be used by the virtual port to which the extension is currently assigned. The parameter has an informative character only and cannot be changed in this menu.

Terminal – this option provides a correct identification of the calling user. It is used only for the extensions that are assigned to the ISDN, SIP or Cornet ports. In other cases, you can connect one terminal only to each physical port and so the terminal identification matches the extension number.

You can connect two terminals to the Cornet port – and but Master Slavethe PBX can only connect one digital telephone to the physical port and so you are recommended to keep the setting.Master

You can connect a bus with up to eight terminals to the ISDN BRI port. Each terminal has its own identification (Multi Subscriber Number, MSN). Assign the MSN numbers to the terminals created on the tab for the StackBRI port. Use the option to assign a selected extension to one of Terminalthe available terminals.

The SIP terminals identify themselves with their SIP URI. Define the terminals on the SIP proxy level. Use the Terminal option to assign a selected extension to one of the available terminals.

Active – this option means that the station on the given port is the main station. Its outgoing calls are identified as this station. Incoming calls are routed to secondary stations too.

Goto virtual port – click the button to move to the current virtual port settings.


Required licencesThis section displays the licence requirements and statuses for the CallBack, Mobility Extension and Call recording services. The fact that a licence is required yet absent or insufficient in the PBX is signalled by a red text. The fact that a licence is required and present and valid in the PBX is signalled by a blue text.

Figure: View of Extension Options

Extension PropertiesThe Properties tab consists of a lot of subtabs, which are described in a separate chapter for convenience. This tab is exceptional because almost all of its parameters follow the hierarchical structure. For the structure and all the parameters refer to S. 9.1

.Setting Properties


ProfilesIn this tab define the properties of an extension within a selected user profile. The extension profile is the highest priority setting. You cannot create new profiles but can edit the existing ones. A list of the profiles created on the user level is displayed to the left. When you select one of these profiles, you will see two new tabs – and Basic

. Find the following parameters in the tab of the extension profile:Properties Basic

Active – activate an extension within a selected user profile. If it is not checked off, all calls coming to this extension are rejected. The extension can establish outgoing calls.

Do not ring at call to user – enable call routing to an extension within call routing to an extension user when the user profile is active. If it is checked off, only the calls routed directly to this extension alert the extension.

The Properties tab consists of a lot of subtabs, which are described in a separate chapter for convenience. This tab is exceptional because almost all of its parameters follow the fall-down hierarchy. For the structure and all the parameters refer to S. 9.1

. Setting Properties Remember that the extension profile level setting has the highest priority!

8.5 Phone DirectoriesHere is what you can find in this section:






8.5.1 User Phone DirectoriesHaving been created, each user is automatically assigned a private phone directory (phone book). A list of user phone directories is displayed to the left of of the Phone

menu. The phone directory has a limited capacity directories – User phone directoriesof records. The default value is 1000 records per user. This limit can be changed using the Maximum user tel. nums parameter on the tab in the user settings. To edit Basicthe records use the menu on the tab. Users – Users & Groups Phone directory

In the context menu on the right-hand side of the menu you can use the following options:

Add – add a row to a selected phone directory.

Delete – remove a selected row from a selected phone directory.

Delete all – remove all rows from a selected phone directory.


Delete all – remove all rows from a selected phone directory.

Find (F3) – search a selected phone directory for a record. Enter complete initial words respecting the lower and upper cases.

Find next (F5) – enable repeated searching of the string that has been entered in the function.Find (F3)

Export – export the current phone directory into an or file.xml csv

Import – import a phone directory saved in the or format.xml csv

The phone directory table in this menu consists of records divided into six columns with the following meanings:

Nickname – set the name to be used for easier phone directory searching.

Name – set the name of the extension for which the record has been created. This name will be displayed on your phone.

Scheme – define whether the entered string represents the or .Number URI

Prefix – set the access prefix defined in the Global data – Global parametersmenu. This prefix automatically precedes the user number in dialling from a phone directory.

Number/URI – define the phone number (or URI) to be entered in the format corresponding to the selected subtype.

Ring pattern – assign different ring patterns to each record of the phone directory. The ring tone will be used if the calling party number of the incoming call matches the phone directory record. If this ring pattern is not set, the default pattern for the final destination will be used.

The last six columns of this tab are used for call forwarding settings for each contact. One pair of columns is intended for each forwarding type. In the first column of each pair set the destination type for forwarding and in the other set a destination of the selected type. This forwarding mode has a higher priority than the mode selected on the Forwarding and Forwarding-exceptions tabs!

8.5.2 Group Phone DirectoriesFor each group of users, a group phone directory is created automatically and filled with the user telephone numbers. You cannot add or remove records manually in this directory. You can just edit appropriate parameters in the , , Scheme Subtype Ring

and call forwarding columns. For the group phone directory refer to the pattern Phone menu. directories – Group phone directories

In the context menu to the right you can use the following options:



Delete all – remove all rows of a selected phone directory.

Find (F3) – search a selected phone directory for a record. Enter complete initial words with respect to Lower/Upper Case.



Export – export the current phone directory into an or file.xml csv

The phone directory table records are divided into six columns with the following meanings:


Name – set the name of the extension for which the record has been created. The name will be displayed on your phone.



Number/URI – define the phone number (or URI) to be entered in a format corresponding to the selected subtype.


The last six columns of this tab are used for call forwarding settings for each contact. One pair of columns is intended for each forwarding type. In the first column of each pair set the destination type for forwarding and in the other set a destination of the selected type. This forwarding mode has a higher priority than the mode selected in the Forwarding and Forwarding-exceptions tabs!

8.5.3 Group Phone Directories (Generated)For each group of users, a dedicated phone directory is generated and filled with the users or extensions as defined in the parameter Generate phone directories from usersin the menu. Every change in the name, number, Global Data – Global parametersscheme or subtype is automatically made in the generated phone directory too. For group phone directories refer to the Users – Phone directories – Group phone

menu. directories (Generated)

In the context menu to the right you can use the following options:

Find (F3) – search a selected phone directory for a record. Enter complete initial words respecting the lower and upper cases.


Export – export the current phone directory into the or file.xml csv


The phone directory table recorda are divided into twelve columns with the following meanings:






8.5.4 Common Phone DirectoriesTo create common phone directories use the Phone directories – Common phone

menu. You can create an 'unlimited' number of phone directories and directoriesassign them to selected groups of users.

The context menu to the right offers the following options:



Delete all – remove all rows of a selected phone directory.

Find (F3) – search a selected phone directory for a record. Enter complete initial words respecting the Lower/Upper Case.


Export – export current phone directory into the or file.xml csv

Import – import phone directory from the or file.xml csv

The phone directory table records are divided into six columns with the following meanings:







1.

2.

3.


The last six columns of this tab are used for call forwarding setting for each contact. One pair of columns is intended for each forwarding type. In the first column of each pair set the destination type for forwarding and in the other set a destination of the selected type. This forwarding mode has a higher priority than the mode selected in the Forwarding and Forwarding-exceptions tabs!

8.5.5 SIP Phone DirectoriesFind the SIP phone directories in the Users – Phone directories – SIP phone directoriesmenu. You can define one general phone directory source for the whole PBX and distribute it to the SIP extensions.

Phone directory source

Here define the phone directory source. On the basis of the source, a phone directory is generated for the SIP extensions and stored in the TFTP storage for the SIP terminals. The following options are available:

Disabled – no directory is generated.

Group – the directory is generated from the directories assigned to the selected group.

User – the directory is generated from the directories of the selected user.

Extension – the directory is generated from the directories of the selected extension.

Each own – the directory is generated from the directories of the user whose SIP extension sent the directory downloading request.

Having received the downloading request, gs_phonebook.xml 2N® NetStar

generates the file in the telephone format from the selected source GrandStreamand sends it.

Having received the downloading request, tftpPhoneBook.xml 2N® NetStar

generates the file in the telephone format from the selected 2N® StarPoint IP T2x

source and sends it.

Having received a downloading request for another file, searches the 2N® NetStar

TFTP storage and sends the file if available.


9. Setting PropertiesHere is what you can find in this chapter:



Fall-Down HierarchyAll the parameters are used according to a fall-down hierarchy of the PBX. PropertiesIt means that setting a parameter on one level you cannot be sure that it will be used. Each level of this fall-down hierarchy has a preset priority. The following figure defines all the fall-down hierarchy levels. The higher the level, the higher the priority.

Figure: View of PBX Fall-Down Hierarchy. Higher Levels Have Higher Priorities

It implies from the figure above that the parameters set on the extension profile level have the highest priority and the parameters set on the virtual port type level have the lowest priority. If a parameter is set to the value on a level, a different setting Defaulton a lower level is searched for this parameter. If a parameter is not set on any level (Default is set on all levels), the PBX uses the value preset by the source code.

Properties


PropertiesThe tab is situated in the menus of all routing objects as mentioned above Properties(Figure 1). By default, the properties are not set on all levels as they are unnecessary for normal PBX operation. To set a parameter for an object, simply push the Create

button. To cancel a parameter, click the button. properties Reset default propertiesThe tab consists of fourteen subtabs, which are logically divided according Propertiesto functions. Some are only used on certain levels because they have no sense on others. The text below explains all the parameters available in the subtabs.

BasicThe parameters of this subtab are mostly divided into sections according to their functions:

No answer timeout [s] – set the maximum time of alerting the called phone. After the timeout, the call establishing process is terminated with the 'User not responding' cause and the calling user hears the congestion tone. The default value is 180s. The maximum value is 1 hour.

Holdoff parameters

Call hold – enable holding of a call. The default value of this parameter is YES(Hold enabled).

Hold tone – set the tone to be played to the user during call holding by the other party. The default tone is the progress tone.Music on Hold

Maximum hold level – define the maximum count of held calls per extension. Reclick the call holding button to get connected to the first held subscriber, push it once again to get connected to the second held subscriber, and so on. If a held subscriber hangs up, the released position can be used for another hold.

Transfer parameters

ICT enabled – (Implicit Call Transfer) – enable call transfer via a connected terminal (phone). You can use one of the following options:

Blind transfer (with hang-up) – the call is not answered by the other party before transfer. E.g. user calls to user . User answers the call and A B Busers and are talking with each other. User holds the call. User A B B Ahears the Music on Hold and user establishes a new connection to user . B CUser does not answer the call and user terminates the call. User is C B Atransferred to user and hears the alert tone instead of the Music on Hold. CWhen user answers the call, users and can talk to each other.C A C

Transfer without hang-up – the call is answered by the other party before transfer. E.g. user calls to user . User answers the call and users and A B B A Bare talking to each other. User holds the call. User hears the Music on Hold B Aand user establishes a new connection to user . User answers the call and B C Cuser terminates the call. User is connected to user and they can talk to B A Ceach other.


Transfer return timeout [ms] – define the timeout for return from the blind transfer. After this timeout (if not answered), the call returns to the extension that transferred it. The default value of this parameter is 20s.

Queue parameters

The queue parameters are only available on the group and user levels. The Station is the only parameter on the station level.polling timeout

Queue – enable call queuing. It means that if an incoming call is routed to a busy extension with a queue, the call is not terminated, the calling user hears the alert tone and can wait for connection. After the current call is terminated, the phone of the called user is alerted again with your call from the queue. If the queue is disabled, the incoming call on a busy extension is terminated with the 'User busy' cause. The default value of this parameter is (queue disabled).NO

User busy when station busy – select whether or not an incoming call shall make the user busy. The option is enabled by default, which means that the user cannot use two own stations at the same time or call from one own station to another.

Queue depth – set the maximum count of calls to be queued. All excessive calls will be rejected as if the queue had been disabled (User busy). If you set this parameter to zero, the count of queued calls will be unlimited. The default value of this parameter is zero (unlimited).

Queue timeout [ms] – define the delay between the end of the previous call and the beginning of the next queued call alerting. The default value 0 s (alerting starts without delay).

Maximum time in queue [s] – define the period of time for which the calling subscriber may stand in the queue. When this time elapses, the incoming call is terminated.

Repeated attempt timeout for stations [s] – set the time interval for the PBX to re-try to route a call to the selected station. The selection is designed primarily for external stations, which may be occupied by a call that is not made via the PBX. The busy status is identified the moment the PBX tries to make a call and the network sends the busy tone (cause 17 – User busy is only respected). Call routing is terminated temporarily, but, with an active queue, it is necessary to know when the given station is free and ready to answer. In these cases, the other user stations are not called repeatedly. The parameter is also useful for internal ports with multiple logged-in stations. While one station is speaking on one port, the other cannot be used and is attempted repeatedly until it gets free or the calling subscriber hangs up. In this case, the other user stations are alerted too since the busy cause from the virtual port does not mean a busy user.

Next call – compile a caution on an incoming queued call. The default value of this parameter is (caution disabled).NO

Next call tone – set the progress tone to be played to the user during an active call if there is a call in the queue for the user. The progress tones with the Offoption can only be used for this function. The tone is a good Queue Alertexample. If you do not have this progress tone in your list, click on Add default

.progresses


Identification parameters

All identification parameters are available on all hierarchical levels of the PBX except for the parameter, which is active on the group and user levels Outgoing hold CLIPonly.

Incoming hold CLIP – forward the called party number to the called user in the case of call transfer made by the extension where this parameter is being enabled. It means that, if is selected, the transferred call will be identified by YESthe CLI of the transferred user ( ) instead of that of the user who transferred it (A

). The default value of this parameter is .B NO

Outgoing hold CLIP – display the original calling party number in the case of call transfer. It means that, if is selected, you will see the calling party number of YESthe transferred user ( ) instead of that of the user who transferred the call ( ). A BThe default value of this parameter is .NO

Use Replace in identification table – replace a part of the preset identification (as specified in the identification table) with your own calling extension identification. The default value of this parameter is (replacement enabled).YES

CLIR – restrict the calling line identification. Use a service or a pre-programmed

phone button to change the station settings. The parameter is set 2N® StarPoint

to NO by default.

Language – select the language to be used by the terminals. 2N® StarPoint

Choose one of the languages listed. The default value corresponds to the preset of the PBX.Localisation

Max phone directory item count – set the maximum count of records in a user private phone directory. This parameter is available on the user level only! The default value of this parameter is 10 records.

Routing

To port – route an outgoing call through the selected port. Outgoing means the outward direction from the PBX.

Fro m port – the section includes two types of call routing.

Normal – define routing of the calls that arrive in the PBX via the port.

For services and calls on hold – set the routing destination for calls on hold or services without a destination of its own. This helps you easily create a complete routing system for PBX external stations.

AutoClip parameters for calls – choose an AutoClip parameter set for storing of AutoClip router records. This option can be used on the user profile, user and group levels only.

No port – set routing for extensions not assigned to a virtual port. It is primarily used for the external, email and virtual port extensions that are used for special routing cases. This setting is available on the virtual port and virtual port type levels only.

For more information on call and SMS routing refer to Subs. .7.1 Routers


Message routingThe tab is available on all hierarchical levels. Its structure is similar to Message routingthat of the Call routing tab but includes several additional parameters.

To port – set routing rules for the messages that go out of the PBX through the port.

From port – set routing rules for the messages that come into the PBX through the port.

AutoClip parameters for messages – assign a set of AutoClip parameters as defined in the menu. All records on outgoing Global data – AutoClip parameters SMS messages are stored in the AutoClip router including these parameters.

No port – set routing rules for the extensions that are not assigned to any port. Such extensions include, in particular, PBX external or virtual port extensions used for special routing cases.

For call routing via the PBX refer to Subs. .7.1 Routers

Parameters of unsuccessful sending

Repeat at fail – enable repeating of a failed SMS sending attempt. An attempt may fail due to a GSM network rejection or bad signal quality. This option does not refer to SMS delivery failures due to switch-off or temporary unavailability of a remote GSM extension.

Number of repeated attempts – set the maximum count of sending attempts at an SMS sending failure due to network rejection or temporary signal unavailability. The default value is 4000.

Time for message repeat [s] – set the interval between successive SMS sending attempts. The default value is 180 s.

METhe subtab is used for setting parameters for external extensions. They mostly MEinclude parameters for call holding using an external extension. Mobile phones are not equipped with a standard PBX call holding key and that is why this function has to be replaced with a sequence of DTMF symbols. This tab contains the following parameters:

Transfer – enable call holding from an external extension. This function is subject to licence and hierarchy. If you define this parameter on the user level, you need as many licences as many extensions the user has (the external extension is not included because it needs licence for use).

Pattern time interval [ms] – set the time interval between individual characters of the FLASH and DISCONNECT patterns. This time interval is important for a correct recognition of the pattern. If the delay between the pattern characters is longer than the preset time value, the pattern will not be recognised.

FLASH pattern – set the sequence of the FLASH pattern characters used for call holding. Re-enter the FLASH pattern to reconnect a held call or switch between two calls (one active and one on hold).

DISCONNECT pattern – set the sequence of the DISCONNECT pattern


DISCONNECT pattern – set the sequence of the DISCONNECT pattern characters. When you have one active call and one call on hold, use this pattern to terminate the active call and reconnect the call on hold.

Do not end outgoing call – select that an outgoing call from the PBX to an external station shall not be terminated after the calling user hangs up. Instead, the user gets the busy tone from the PBX and can, using the DISCONNECT PATTERN, return to the dialtone and dial the required station through the PBX. The selection is disabled by default.

Do not end incoming call – select that an incoming call from an external station to the PBX shall not be terminated after the calling user hangs up or the DISA fails. Instead, the user gets the busy tone from the PBX and can, using the DISCONNECT PATTERN, return to the dialtone and dial the required station through the PBX. The selection is disabled by default.

ForwardingThe whole tab is available on the group and user levels only.

This subtab is used for call forwarding setups. The unconditional call Forwardingforwarding (CFU) has the highest priority. The other two forwarding types have the same priority and each is used in a different situation. The call forwarding settings on this tab can be changed for a selected group of users in the Forwarding exceptionstab, which has a higher priority. Furthermore, it holds true that if extension forwards Aits calls to extension , then extension can call to extension without being B B Aforwarded. This function is called . The following parameters are Boss-secretaryavailable in the subtab:Forwarding

CFNA – Call Forwarding at No Answer – set call forwarding in case the called user fails to answer within a timeout.

CFNA timeout [ms] – set the timeout for CFNA forwarding. After the timeout expiry, the call is forwarded to the preset destination.

CFU – Call Forwarding Unconditional – set call forwarding of all incoming calls (highest priority). Each incoming call is forwarded to the preset destination regardless of other settings of this subtab.

CFEC – Call Forwarding on Error Cause or Busy – set call forwarding in the case of routing on a busy extension or routing ending up with an error cause.

Cause object – select one of the cause objects as pre-defined in the Global data menu. Basically, a cause object is a set of error causes, – Causes – Cause objects

which are subsequently respected in the CFEC forwarding mode. You can disable the cause objects or use the default settings of any of the lower levels (Default).

Forwarding – exceptionsThe whole tab is available on the group and user levels only.

The subtab is used for specifying exceptions from the Forwarding – exceptionsforwarding rules set in the subtab. The exceptions are also applied when Forwardingno forwarding rule has been set in the subtab. This tab is also called a Forwarding


Black/White list. Use the phone directory to fill the list with addresses.A field is available here for each type of forwarding (CFU, CFNA and CFEC), where you can add a limited number of exceptions. Each row represents one exception and is divided into two parts. In the first part, the calling party is defined for which this exception will be valid, and in the other part, a new call routing rule is determined. Setting the calling party, select a extension, user, ring group, virtual port, virtual port type and extension type. Setting the rule, choose any of the destinations available in the subtab, or select one of the following three options:Forwarding

Disabled – disable call forwarding as defined in the subtab for a Forwardingselected calling user (users) and allow this user (users) to call the selected destination.

Enabled – route a forwarded call to the destination preset in the CFW field. If this field is not filled in, the call is routed as with (Forwarding enabled)

the option.Disabled

Rejected – terminate a forwarded call with the CALL REJECT cause. The calling user hears the congestion tone.

TonesUse the tab to define the basic tones of the PBX to be played to the calling Tonesuser. The menu is divided into three parts. The first part, , helps you set various Dialdial tones, the second part, , helps you set various alert tones and the third part, Alert

, helps you set various congestion tones. To add a row defining which tone Congestionwould be used for which situation use the context menu. A list of situations (states) related to specific types of tones is displayed in the column. A list of available Typeprogress tones is displayed in the column. It holds true for the dial tones that a Tonehigher row has a higher priority. It is because there may be more valid conditions than one in the dial tones. To change the priority of the rows, use the two arrow buttons on the right. To add a row with a certain priority easily use the and Insert ahead selected

options. The option is used for adding a record behind the Insert behind selected Addcurrently last one (i.e. the lowest priority record).

Ring patternsIn the tab assign the ring tones according to the calling party. Add more Ring patternsparameter rows to the table using the context menu. The meanings of the columns are as follows:

Destination type – choose one of the listed objects. When this object is recognised, the extension rings according to the column. Choose one of Patternthe CLIP, extension, extension type, user, group, virtual port and virtual port type options.

Destination – set an object of the type selected in the column. If is Type CLIPselected, fill in (Calling Party Number).CLI


Scheme – this column is active only if the option has been selected in the CLIP column. It defines whether the incoming CLI (Calling Party Number) Type

scheme is or . If this column fails to match the incoming scheme, this Number URIrow will not be used.

Pattern – select a the ring pattern from the list of available PBX patterns.

Again, it holds true that a row situated higher in the configuration has a higher priority. To change the priority of the rows, use the two arrow buttons on the right. To add a row with a certain priority easily use the and Insert ahead selected Insert behind

options. The option is used for adding a record behind the currently last selected Addone (i.e. the lowest priority record).

Softphone

The subtab is used for setting parameters of the key phones. Softphone 2N® StarPoint

These settings are not created automatically (except for the Default IN virtual ports), but on all levels of the fall-down hierarchy using the Create Softphone extensionbutton. To delete a setting, use the button. The Remove Softphone extension

tab has two subtabs: and . The subtab is Softphone Keypad Parameters Parameters

used exclusively for setting , or key phone 2N® OptiSet 2N® StarPoint 2N® OpenStage

parameters. The tab helps you set the type of the terminal connected. The key Keypadphones are detected automatically on the extension level only. The other subtab

settings relate to , or key phones again.2N® OptiSet 2N® StarPoint 2N® OpenStage

Terminal – choose an item from a list of available terminals:

ANALOGUE – indicates any analogue terminal.

ECONOMY – indicates a key phone – type.2N® StarPoint Economy

ADVANCED – indicates a key phone – type.2N® StarPoint Advanced

ENTRY – indicates a key phone – type.2N® StarPoint Entry

BASIC – indicates a key phone – type.2N® StarPoint Basic

STANDARD – indicates a key phone – type.2N® StarPoint Standard

ISDN – indicates any ISDN terminal.

GSM – indicates any GSM terminal.

VoIP – indicates any VoIP terminal.

Optiset Advance – indicates a key phone – type.2N® Optiset Advanced


Optiset Standard – indicates a key phone – type.2N® Optiset Standard

Optiset Entry – indicates a key phone – type.2N® Optiset Entry

OpenStage 10 – indicates a type 10 key phone.2N® OpenStage





2N StarPoint IP T20 – indicates a type T20 IP phone.2N® StarPoint IP






Well VP-2009 – indicates a Well VP-2009 IP phone.

Well VP530 – indicates a Well VP530 IP phone.

Well YV2 – indicates a Well YV2 IP phone.

Well YV3 – indicates a Well YV3 IP phone.

Yealink T19P – indicates a Yealink T19P IP phone .

Yealink T21P – indicates a Yealink T21P E2 IP phone.

Yealink T23G – indicates a Yealink T23G IP phone.

Yealink T27P – indicates a Yealink T27P IP phone.

Yealink T29G– indicates a Yealink T29G IP phone.

Yealink T46G – indicates a Yealink T46G IP phone .

Yealink T48G – indicates a Yealink T48G IP phone.

Yealink W52P – indicates a Yealink W52P IP phone .


Helios IP 1 (3, 6) – indicates a intercom with one, three or six 2N® Helios IP

buttons and with the keyboard or display.

Extenders

Having chosen one of the key phones, you can connect 2N® StarPoint

extenders with further programmable keys. You can connect up to four extenders with sixteen keys (extender type S16) or up to two extenders with ninety keys (extender type S90).

One eighteen-button extender (type S18) or ninety-button extender (type

S90) can be connected to the phone. One eighteen-2N® OpenStage 40

button extender (type S18) can be connected to the or2N® OpenStage 30

phones. OpenStage 15 2N®

Caution

Use an external phone and extender power supply for the ninety-button extender.

Up to two 38-button extenders (IP key module) can be connected to the

and phones.2N® StarPoint IP type T26 T28

40-button extender EXP40 can be connected to the andYealink T46G Yealink T48G.

8/16-button extenders or an Infopanel can be connected to the selected 2N

intercom.® Helios IP

Restart IP terminal – restart the selected IP terminal. The function is available for

the terminals only. 2N® StarPoint IP T2x Make sure that the terminal type is

.correctly completed in the Virtual ports – SIP – Stack – Terminals menu

With the , , , and terminals, you can set the Entry Economy Basic Standard Advanced following parameters:

Key setting

To program the phone keys display a dialogue box by clicking on the selected key. Select the key function and legend in the window and choose any of the following functions:

Auto answer – set the timeout after which the incoming call is answered automatically.


CLIR – restrict the calling user identification. The shining button LED indicates that the CLIR function is active.

DEFAULT – clear all the key functions on the given fall-down level.

DO NOT DISTURB – activate the DO NOT DISTURB mode, in which the extension is inaccessible for incoming calls and the calling user gets the busy tone. Outgoing calls from the extension are not limited in this mode.

ESC – push the key to reject incoming calls, return to a superior level or Escapeclear a character in an item.

FLASH – push the key to hold calls. If a call is on hold, you can dial another Flashuser or service number. Re-push the key to switch between two calls (one active and the other on hold).

STATE – click the button to set speed dialling for the selected number and Statemonitor the state of the selected virtual port, user or extension at the same. The user state displays all user extensions. The state is indicated by a LED at the button:

INTERCOM – push the key to interconnect two key Intercom 2N® StarPoint

phones. All you have to do is enter the CPN in the dialogue box. After you push the key, the connection is established automatically. The called user needs a HandsFree key phone for this function.

Quit – the LED is off if all the user extensions are at relax (the selected extension or virtual port is at relax).

Hook off – the LED is on if one of the user extensions at least is off-hook (the selected extension or virtual port is off-hook).

Is alerted – the LED is flashing if one of the user extensions at least is ringing (the selected extension or virtual port is ringing).

PHONE DIRECTORY – display the phone directory. This key has the same function as the right arrow.

MUTE – push the key to temporarily deactivate your key phone Mutemicrophone. Push the key again to re-activate it.

REDIAL – push the key to dial the last-dialled number.Redial

HANDSFREE – push the button to switch a call from the HandPhone HandsFreeto the HandsFree mode and back for the StarPoint key phones only. The HandsFree mode uses a microphone and loudspeaker placed on the terminal body.

MISSED CALLS – enter the menu.Missed calls

DIALLED CALLS – enter the menu.Dialled calls

ANSWERED CALLS – enter the menu.Received calls

NO FUNCTION – this option has no function and ignores any fall down from lower-priority levels.

NEW MESSAGES – enter the menu.Received messages

PROFILES – enter the menu for profile activation/deactivation.Profiles

ACTIVATE PROFILE – activate or deactivate a selected profile directly. The option is not available until the user profile has been created.


CALL SLOT – use the button as a slot for another call on the selected extension. The count of the call slots is limited by the parameter on the Maximum hold level

– tab. Push this button during a call to hold the call, the held Properties Basicsubscriber hears the dialling tone and can route the call to another extension. If the Queue parameter is enabled on the – tab, the speech slots Properties Basicare occupied with incoming calls. You can thus switch between the calls: the inactive call is on hold and the caller hears the call holding tone.

CALL RECORDING – click the button to start call recording and reclick the button to stop. Refer to the subtab for call recording settings.Recording

Parameter setting

The subtab offers the following parameters:Parameters

Key volume – set the loudness of the key pushed in the handset or HandsFree. The parameter may range from 0 to 15.

Ring volume – set the loudness of the ring tone. The parameter may range from 0 to 8.

HandsFree volume – set the HandsFree loudness. The parameter may range from 0 to 15.

Headset volume – set the loudness of the headset. The parameter may range from 0 to 15.

Display contrast – set the display contrast. The parameter may range from 0 to 7.

Time format – set the time format. Choose either a twenty-four-hour or twelve-hour format.

Call list type – set the displaying of call records in one of the following formats: .Name and time, Number and time, Name list or Number list

Call list type – set the call list displaying in one of the following formats: Name .and time, Number and time, Name list or Number list

Phone list type – set the phone directory displaying in one of the following formats: .Name list or Name and number

Message list type – set the displaying of received messages in one of the following formats: .Name and time, Number and time, Name list or Number list

Default tune – set the ring tune for the StarPoint key phone. The key phone uses a predefined ring pattern, but this pattern is played with the tune defined in this parameter. The default tune is Cornet Elephant.

Hang up timeout – set the hang-up timeout after which the opposite party terminates the call. Having received disconnect, the Cornet port hangs up after the timeout regardless of whether the call is HandsFree or uses a handset. When used with applications and HandsFree, this timeout should be considerably reduced to approximately 5s.

Message ring tone – define the ring tone for the caution on a received message.

Intercom ring tone – define the ring tone for the caution on an incoming intercom call.

Information type at relax – set the format of information to be displayed on the second line of the key phone at rest. Choose any of the following options:

Name – shows the extension name.


Name – shows the extension name.

Number – shows the extension number.

Profile – shows the active profile name.

Name and profile – shows the extension and active profile names.

Number and profile – shows the extension number and active profile name.

Name and number – shows the extension name and number.

Information type at incoming call – set the format of information to be displayed on the first line of the key phone at the time of incoming call ringing. Choose one of the following options:

CLIP – shows the calling party number (CLI) only.

CLIP and CPN – shows the calling party number (CLI) and originally called party number (original CPN).

CLIP and CPN list – shows the calling party number (CLI) and originally called party number (original CPN). In both cases, the numbers are compared with the phone directories. If a match is found, the name is added to the number.

Waiting for next key – set the cursor rate for proceeding from one position to

another while typing a text on a key phone. Choose one of the 2N StarPoint ®seven levels, starting from 'extremely fast' to 'extremely slow'.

Transfer incoming call with speed dial – enable switching to speed dialling during ringing. If this parameter is disabled, the incoming call is rejected upon the speed dial button press.

Phonebook edit enabled – enable/disable phone directory editing via a key phone.

AoCThe whole tab is available on the group and user levels only.

The subtab helps you define the warranted count of call records to be displayed AoCon the key phone. Set the count of , and calls separately. The Missed Received Dialleddefault count of records is twenty. This limit applies to PBXs with a high number of users (carried calls) only. If a PBX has a capacity to store more records, all of them are displayed at any time.

Caution

The web application can display up to 20 records 2N® NetStar Assistant

of each type, i.e. the total of 60 records.

SMS at no answer


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Use the subtab to set parameters for the SMS at no answer SMS at no answerfunction. An is a message that is sent automatically to the called SMS at no answeruser if the following conditions are satisfied:

The called party rejects or does not answer the incoming call.

The SMS at no answer is activated on one of the levels of the fall-down hierarchy for the calling or called user.

The SMS at no answer is sent through the destination defined here. Usually is this destination defined as bundle of virtual GSM ports. Type the SMS text to be sent to the called users in the SMS subtab. You can also add information on the calling number (string %c) and the calling extension name (%n) to the SMS body.

Note

If you use a text string in the SMS text, select a phone directory for %nthe port via which the SMS is to be sent in order to complete the name. Refer to the in the Send information on name 3.10 Virtual Port Optionssubsection.

SMS setting:

Send SMS at no answer – send this SMS as an to the called SMS at no answerextension. This setting has a higher priority than that for an external extension.

Send SMS at no answer of external extension – send this SMS as an SMS at no to the called external extension. This setting has a lower priority than the answer

general one mentioned above.

Moreover, you can define the sending timeout. If you set this timeout to 20 s, the outgoing call has to alert the called extension for more than 20 s so that the SMS can be sent after the call end. The timeout is the fourth condition for SMS at no answer

.sending

Examples of SMS at no answer sending:

User calls to an external GSM phone via a GSM port of the PBX. The SMS at no Aanswer has been enabled and the timeout is 20s. The GSM phone begins to ring, but the called user does not answer the call. After the timeout (20s), user Ahangs up (or the call is rejected by the other party). Then the SMS set in the upper configuration row is sent to the selected destination. If the Origin type is selected for the destination, the SMS will be sent via the same GSM port that was used for the call .

User calls to an external GSM extension of user . The SMS at no answer is A Bdisabled for User while enabled with no timeout for user . After no answer or A Bcall rejection by the external GSM extension, an SMS at no answer is sent


2.

containing the text as defined in the bottom row of the configuration of user . BWhen the SMS at no answer is enabled for user too, an SMS at no answer is Asent containing the text from the upper row of the configuration of user .A

Services

The subtab helps you create individual service settings, thus replacing the Servicesglobal ones. You can modify such parameters as progress tones, timers and routers, or activate the PIN request in the service settings. To disable individual settings click the

button.Remove individual setting

User reservation

The service helps you reserve the user that is not accessible at the User reservationmoment (is busy or does not answer). When the function is enabled both for the called and calling users, the PBX starts monitoring the called user's activity. As soon as the user becomes available, the User reservation service makes a CallBack to the user-reserving subscriber and automatically connects the two users.

Note

The service is only available for internal PBX calls.

Note

The service is enabled by default.

The tab is divided into three sections: User reservation Reservation made by other enables/disables a user to be reserved by the other users and users Reservation made

and help the calling user set the reservation by this user Settings for serviceparameters.

Reservation made by other users

Others can make reservation on this user – enable/disable the called user to be reserved.

Maximum time of reservation – set the maximum reservation time. Reservation will be cancelled after this timeout.

Reservation made by this user – set the reservation parameters for the calling user or enable reservation in the following situations:

Enabled when busy – enable/disable reservation if the called user is busy.

Enabled when no answer – enable/disable reservation if the called user does not answer within a timeout.

Enabled when queue – enable/disable reservation if the user's queuing time expires.


Settings for service – enable the caller to change the service parameters contrary to the global settings.

Reserve when busy – set the tone to be played to the calling user if the called user is busy and enabled reservation.

Reserve when no answer – set the tone to be played to the calling user if the called user does not answer and is enabled reservation.

Reserve when queue – set the tone to be played to the calling user if the queuing time expires and the called user is enabled reservation.

Reservation confirmation – set the tone to be played to the calling user when the called user reservation has been successful (upon entering the reservation code).

Dialling error – set the tone to be played to the calling user after the third attempt to enter an invalid reservation code.

Incoming call tone – set the tone to be played to the calling user after receiving the User reservation call. After that, the reserved user is dialled automatically.

Reservation code – set a numeric code to be entered by the calling user for user reservation confirmation.

Create settings / Delete individual settings – create/delete individual service settings for the given level.

Recording

The whole tab is available on the user, group, virtual port and virtual port type levels only.

You can set the call recording parameters in this tab. Find the tab in the Properties on the user, group, virtual port and virtual port type levels.

Recording – specify the recording mode for the level.

Default – settings from other hierarchical levels are used.

Disabled – disable call recording regardless of lower level settings.

Upon request – enable call recording activation/deactivation during the call. The parameters Pattern length, ON pattern and OFF pattern are used.

Turn on at alerting – call recording is activated by ringing detection and deactivated by the call end.

Turn on at connect – call recording is activated by the call connection and deactivated by the call end.

Recording direction – define which calls are to be recorded from the viewpoint of the PBX (not the user or group).

All – all the object calls are to be recorded.

Incoming only – all incoming calls to the PBX are recorded only. These calls are outgoing calls when viewed by the user/group (the user/group calls the PBX).


Outgoing only – all outgoing calls from the PBX are recorded only. These calls are incoming calls when viewed by the user/group (the user/group phone ringing ).

Marked only – the outgoing calls from the PBX are only recorded that have the option enabled in the C omplex router.Recording mark

Marked only and incoming – the outgoing calls from the PBX that have the option enabled in the C omplex router plus all incoming Recording mark

calls to the PBX are recorded.

Pattern length [ms] – set the time for which the recording ON/OFF pattern characters are to be awaited. If a character comes after the timeout, it is identified as invalid (the pattern is not recognised).

ON pattern – define the recording ON pattern during the call. The parameter is applied by PBX extensions only if the recording mode is active. Upon requestThe default pattern is 1*.

OFF pattern – define the recording OFF pattern during the call. The parameter is applied by PBX extensions only if the recording mode is active. Upon requestThe default pattern is 3#.

Hang up at unsuccessful recording – use the parameter to enable call termination in case call recording fails (due to an unavailable storage, invalid licence, etc.).

The following parameters relate to the saved files with call records. If they are inactive, create new properties of the object used. Thus, you add a row to the Call recording logical storage.

Item lifespan [s] – set the storing time for a file with a call record. When the time expires, the items are deleted sequentially. The item lifespan is automatically set to 10 years if this option is not selected.

Maximum count of items – set the count of records in a storage for an object. After the limit is reached, no more calls are recorded unless deleting of oldest files is enabled.

Available space for all items – define some space in a storage to be reserved for records of the given object. When the space is full, no more calls are recorded unless deleting of oldest files is enabled.

Delete oldest after reaching limit – enable deleting of the oldest record files if necessary.

Caution

2N TELEKOMUNIKACE a.s. shall not be held liable for any recording errors due to unavailable network disks and/or exceeding of the maximum storage capacity.

Customer


The subtab provides parameters for functions that have been implemented Customerfor a specific customer and so their meanings will be explained marginally only. This subtab is divided into three sections. Define the supported method of the CPN sending for call billing purposes in the first section. Set the called party identification to be transmitted via a DSS1 message in the remaining parts.

Forwarding number – is used in DSS1 messages for communication with Nokiaexchanges for billing purposes. Set the (Number or URI), Scheme Subtype(Unknown, Internal, Local, National, International) and (specific Number/URInumber or address).

Facility – is used in DSS1 messages for communication with exchanges Ericssonfor billing purposes. Set the (Number or URI), (Unknown, Scheme SubtypeInternal, Local, National, International) and (specific number or Number/URIaddress).

Enabled – enable/disable sending of the Facility or Redirecting number in signalling for the given level.

Enabled when forwarding – enable/disable sending of the Facility or Redirecting number in signalling for the given level in the case of call forwarding . If this option is enabled for a user with call forwarding enable to a destination off 2N®

, enter the user number into the Facility or Redirecting number NetStarparameter.

The number for call billing is sent using this information element. Make sure that the parameter is set for the virtual port too. Choose , or .Yes, No Default

In addition to DSS1, the parameter is also used in SIP signalling, where it sets the header.Diversion


10. Billing and TariffsHere is what you can find in this chapter:

10.1 Billing and Tariffs


10.1 Billing and TariffsThe menu describes tariffs offered by network providers. The tariffs Billing and Tariffsare then used for deducting free minutes and SMS messages for virtual ports. In future, the menu should facilitate accounting and least cost routing.

ProviderAdd a provider in the left menu column. The item is just a group including all call billing rules.

Context menu options:

Add – add a provider.

Rename – rename the selected provider.

Delete – remove the selected provider.

Default – reset the pre-defined provider.

Credit ListYou can enter any number of credits for each provider and describe each credit with a different set of properties.


Add – add a credit.

Rename – rename the selected credit.

Delete – remove the selected credit.


Destinations/Time ConditionsYou can add a destination and time condition to each credit in this section. Destination means the target network to be dialled.


Add – add a destination.

Rename – rename the selected destination.

Delete – remove the selected destination.

Tariff setting – set or change the time condition for the selected destination.

Tariff DescriptionContext menu options:

Add – add a row to the destination describing table.

Delete – remove the selected description table row.

Column description:

Note – for information only.

Minimum charged time [s] – set the minimum call cost. If a call is answered, these seconds are charged to the calling subscriber regardless of the duration of the call. Typically, this value is set to 60 s.

Charge to [s] – set the call billing interval after the Minimum charged time elapses.

Valid to [s] – set the end of the interval to which a row applies. For example, if you set 360 s, the row will be used for the first 6 minutes only and then the next row will be applied. If you set , the row is valid without limitation.0

Tariffication pulse time [s] – set the tariff pulse sending frequency for the DDS1-supporting devices. Tariff pulses are not sent by default (value 0).

The arrows next to the window are used for modifying the row sequence.


Prefix ListHere you can manage the prefixes that relate to the selected destination and credit. If one of these prefixes is dialled via a specific virtual port and the time condition is met, the rules for this credit are used for billing.


Add – add a row to the prefix table. Matching prefixes are highlighted.

Delete – remove the selected prefix table row.

Column description:

Prefix – display the prefixes that are assigned to the selected credit destination. If * is included, the rule applies to all prefixes.


11. Configuration ExamplesHere is what you can find in this chapter:

11.1 Other Useful Information



COM Port and Communication Program SettingThe basic equipment of the Miscrosoft Windows OS, HyperTerminal, is used for connection. The whole setting of this application is shown in the figure below.

Console setting

Figure: View of HyperTerminal Application Settings


Console structure

Figure: View of Console Structure for Easier Orientation



Mobility Extension

The is an extension feature of the PBX, which enables Mobility Extension 2N® Netstar

external extensions to make use of the features that are not normally available as well as practically all PBX services. The Mobility Extension is associated with the existence of external extensions. Before starting creating external extensions and configuring their routing rules, make sure that you have a valid licence and if so, for how many extensions the licence is valid. Verify this in the licence tables in the lower right-hand part of the menu. In the row called you Global data – Licences Mobility Extension usercan see the count of licences owned (third column) as well as the count of licences required by the PBX (last column). The last column shows not only the external extensions created but also the Transfer parameter on the Properties tab on the extension and user levels. If you set this parameter to Yes, only one licence will be required for the extension, and the count of licences required for the user will be the same as the count of the user extensions (external extensions are counted only once).

External Extension CreationIn principle, an external extension can be created in three ways. First, you can create an external GSM/ PSTN extension while adding a user as shown in . A Figure 1conventional GSM extension (cellular phone) is configured in the GSM part and a normal fixed line of a public network subscriber is configured in the PSTN part. In the case of GSM external extensions, the GSM text is automatically added to the end of the name, and in the case of PSTN external extensions, the PSTN text is added. To forward SMS messages delivered to the user's cellular phone of an external extension, check the Resend SMS option. For the SMS routing configuration see below.


Figure 1: Creating External Extension by Adding a User

The second way to create an external extension is to create an external extension and then assign it to a specific user (an external extension may not exist without its user). This can be done in the menu. In the context menu, add Users – Extensions – Externalan extension, enter the name and number, assign the extension to a user and, if necessary, enable the Resend SMS option. Finally, click OK for confirmation. See Figure 2 below.


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Figure 2: Additional Adding of External Extension to UserThe third and last way is to add an external extension over the given user. In the Users

menu select the user, display the context menu and click – Users & Groups Add . Complete the extension name and number and enable the Resend SMS extension

option if necessary. See Figure 2 above again.

Having added an external extension and saving the configuration, check the external extension for a valid licence on the tab in the section on the Basic Required licences extension level. Refer to the menu for the total count of the Global data – Licences Mobility Extension user licences.

Routing Incoming Calls with Mobility ExtensionWhat is necessary:

an external extension

From port routing

DISA direct inward dialling

permitted transmission for putting a call on hold

Routing of incoming calls from external extensions is associated with the recognition

of these calls immediately after arrival at one of the ports. This 2N® NetStar

recognition is based on CLIP (caller identification) compliance with its subtype. Further routing is then governed by the settings of the recognised external extension (

tab in the extension properties, see ). The item helps route Routing Figure 3 From portthe incoming calls. It consists of and . The first item sets the Normal Services and helddestination for the initial routing of an incoming call. The other is used for holding of a call or dialling a service with no destination defined. For easier understanding, Annex 1shows a flow chart of processing of an incoming call from an external extension.


Figure 3: View of External Extension Routing Tab

As shown in Fig. 3, incoming calls from an external extension are routed to the DISA dial-in. See for specific DISA settings. With this configuration, the incoming Figure 4call is routed to the Default router. Then a 10-second dialling timeout follows, which is detected by the DTMF detector (to include it, check the DTMF option in the lower part). If the 10-second timeout expires without dialling detection, the call is routed to the Operator extension.


Figure 4: View of DISA Configuration for Mobility ExtensionThe Default router usually gives the user a much broader scope of operation than the

above-mentioned Internal router because it is one level higher in the hierarchical structure. It allows the user to call internal extensions, use the services and also call public networks. Restriction of calling international numbers can be achieved, for

example, by including an authorisation router.

To distinguish the rights of different external extensions, just create multiple DISA dial-ins and routers and assign different routers to groups with identical rights in the DISA to specify the calling user rights.

After the call has been made, it is possible to put it on hold any time. In order to access this function it is necessary to permit it at one of the hierarchical levels. Such permission is made in the ME tab in the Properties ( ). is used for Figure 5 Flash patternputting a call on hold and switching between active calls. helps Disconnect patternterminate one of two active calls and return to the other one. The Pattern time intervalspecifies the possible delay between the entering of the characters of one pattern (time between pressing 7 and pressing *). If the delay is longer than the set value, the entered pattern will be evaluated as invalid.


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Figure 5: Common ME Tab Configuration with Enabled Transmission of Dialling of External Extension

What Is Necessary for Routing Outgoing Calls with Mobility Extension

an external extension;

No port routing;

From port routing;

a bundle of ports;

permitted transmission for holding a call.

Routing of outgoing calls to an external extension mainly depends on the configuration of this extension and on the way it is called. In principle, an external extension can be called in two ways. The first is to call a number, which is then routed to the respective external extension. The second way is to call a user of this extension. In such case, it is necessary to uncheck the option on Do not ring when calling a userthe tab at the extension as shown in .Basic Figure 6


Figure 6: Part of Configuration of External Extension with Number Used for Routing

The port via which the call is to be made determines the setup of the Without portrouting on the tab in the extension properties (or, as the case may be, the Routingtype of the extension when using mass configuration by the fall-down structure).

shows a suitable routing solution for an outgoing call of an external Figure 3extension. The inclusion of a bundle of GSM ports in the Without port routing reduces the probability of call rejection at the respective external extension when one GSM port is busy at the moment.

One of the possible settings of the bundle is shown in . The selected cyclic Figure 7allocation strategy means that at the first attempt to call one of the external extensions with this bundle set in the routing, the call is routed to the Without portfirst row of the bundle, i.e. to port GSM 1. At the second call attempt it is routed to port GSM 2, subsequently to port GSM 3, and then to port GSM 1 again. In case the specific port is busy at the moment of routing, the next row is automatically used. To ensure this, check the option. In case all the ports are busy, the Next row if caller busycall can be routed to the Default destination, which is a back-up solution.

By credit is another suitable bundle strategy for this case. This strategy is intended for credit-monitored bundles with virtual ports. An incoming call is routed to the virtual port of the bundle with the currently highest count of free minutes. If there are more rows with the same count of free minutes, the sequence of rows in the bundle is respected. If a row object is busy or unavailable, the call is routed to the next row or terminated.


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Figure 7: Typical Configuration of GSM Port Bundle for Outgoing Routing to External Extension

What Is necessary for SMS at No Answer


SMS at no answer setting in Properties.

These SMS messages are used for information on missed calls. To send them, configure the item on the tab on one of the hierarchical SMS at no answer Propertieslevels correctly. A typical configuration is shown in . Here you can see that the Figure 8configuration is divided into two parts. The first part represents configuration for the SMS sent by the PBX to the counterparty when the call initiated by an internal or external extension of the PBX with these settings is not answered or is rejected. The second configuration part is used for notification on a missed/rejected call of the external extension from the PBX, which was initiated over the GSM network.

You can enable/disable the SMS and set the after which the Minimum alerting timeSMS shall be sent if the call is not answered by the counterparty. Both the SMS configuration parts can be active at the same time as shown in Fig. 8. The SMS at no answer in the incoming direction (first part of configuration) has the priority in sending. The SMS is sent via the selected destination, typically a GSM port or a GSM port bundle. If you set as the destination type, the SMS will be sent directly via Origin the port (GSM) via which the unsuccessful call attempt was made.


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The possibility to use the strings %n and %c for caller identification is very useful: %c provides the caller's number (CLIP) and %n includes the name specified in the respective phone book.

Figure 8: Typical Settings for Sending SMS at No Answer

What Is Necessary for Routing Outgoing SMS to External Extension


No Port message routing

Routing of outgoing SMS in is governed by the message routing settings 2N® Netstar

on the tab on one of the hierarchical levels (typically Group, User or PropertiesExtension). This tab, together with the typical settings, is shown in . The part Figure 9of configuration marked as is used for SMS routing or forwarding to an No portexternal extension. Here a specific port is set via which the SMS is sent to the routing number of the external extension, but it is also possible to set a bundle of ports.


Figure 9: View of Typical Settings for Message Routing to External Extension

To make SMS sending to an external extension work properly, check the Resend SMS option in the configuration. The flow chart for SMS sending to an external extension is included in . The procedure of forwarding of SMS received by the external Annex 3extension user is shown in .Annex 4


Appendix

Annex 1: Flow Chart Showing Processes for Incoming Call from External Extension

Annex 2: Flow Chart Showing Processes for Outgoing Call to External Extension


Annex 3: Flow Chart Showing Processes for Sending SMS to External Extension


Annex4: Flow Chart Showing Processes for Forwarding SMS to External Extension



Setting IP Address and Time

IP address

Connect to with HyperTerminal tool2N® NetStar

Rate: 115200

Flow control: None

You can also use Putty

Rate: 115200

Serial line: set the COM interface number you are using for connection

between your PC and 2N® NetStar

Once you are connected, press <ENTER> to get the login screen. The default login information is and . If there is a # character and no login request on the Admin 2nscreen, type and push Enter for confirmation. The login request should get NsCondisplayed.


Having inserted correct login information, you will get the initial configuration screen.

By pressing the proper digits you will get to the configuration menus. Press 1 and 1 for IP configuration.


The IP configuration screen gets displayed. Press options 2 to 4 for IP setting. To escape from the menu or cancel the current operation, use the <ESC> key. Having completed all settings, push <ESC> twice to get to the default menu.

TimeWhen you are in the initial screen for time configuration, press 1 and 3. You will get into the menu. For correct configuration, push 3 to set the time zone and Time settingthen enter the number according to your location. Then you can modify time and date using options 1 and 2.

You can also use the NTP server available in the LAN. Press option 4 to set the IP address or domain name of the NTP server (one DNS at least has to be set in the IP settings).


Connection of Configuration Tool to 2N ® NetStar

Start the configuration tool. If there is no connection to , 2N® NetStar 2N® NetStar

create a new one.

Create a new group for the customer by choosing the Folder icon and the On the option and name the new group. In our case it is called .same level Test

Once a new group is created, right click on it and go to the Create PBX – To group


and create a new PBX called .Test

Click to open the IP setting screen. Set any name you want. In our case we use OK to mark that the local IP will be used. Fill in the IP address into the IP address Local IP

field that you set for in the first step.2N® NetStar


Having completed all the steps above, create the connection to . Just 2N® NetStar

double click on the option with the On-line text at the end of the line. Before connection you will be asked to enter your user name and password.

Configuration WizardThe aim of the configuration wizard is to provide you with an easy basic installation. The ISDN BRI parameters are specified during configuration (click not to use NextISDN BRI).


Then the hardware is activated. When the activation is completed, you will get the screen shown below. Please note that hardware activation can take more than 5 minutes depending on the hardware configuration used.

When the hardware detection is finished, click to continue.Next


After the hardware is activated, the wizard will guide you through the basic gateway configuration settings like localisation, where you have to choose the country where

will be installed,2N® NetStar

time zone settings

and purpose of the . Here choose the GSM GW option.2N® NetStar


When asked for SMTP settings, choose .Next

The last screen will ask you for settings. Here choose the preferred LCR Routerstructure. In our case it will be Default routers. Then click .Next


When you get to the final overview, click to get to the configuration interface.Finish

To apply the configuration created by the wizard scrip, save the changes to 2N®

using the saving icon.NetStar


Interface Configuration

PRI ISDN

The most important aspect of PRI interface configuring is the configuration of the PRI

line between and the PBX systems. The first information we need is the 2N® NetStar

PRI interface configuration in the PBX. In case you are not sure about the PBX PRI port configuration, contact a person responsible for the PBX maintenance without delay. In our case, the PBX was connected to the PSTN and so the PRI port is

configured as TE. For correct interconnection, has to be configured as 2N® NetStar

NT. To do so you have to:

Set PRI card jumpers – switch into the service mode. When the light 2N® NetStar

goes off on the PRI card, remove the card from the rack and check the jumper configuration. For a correct jumper placing use the sticker on the PRI port.

Set correct communication protocol in the configuration tool. Go to 2N® NetStar

the menu in and choose the port to be configured. Set the Boards Hardware and tabs for this port. When you are in the Stack menu, set Virtual port Stack

the interface mode to .NT


To apply your new configuration, save the changes to using the saving 2N® NetStar

icon (or Ctrl+S). To configure the interface into the TE mode, take the same steps and set the jumpers and interface mode to TE.


BRI ISDN

The most impotant aspect of BRI interface configuring is the configuration of the BRI

line between and the PBX systems. The first information we need is the 2N® NetStar

BRI interface configuration in the PBX. In case you are not sure about the PBX BRI port configuration, contact a person responsible for the PBX maintenance without delay. In our case, the PBX was connected to the PSTN and so the BRI port is

configured as TE and Point-to-Point. For correct interconnection, has to 2N® NetStar

be configured as NT and also PTP. To do so you have to:

Set BRI card jumpers – switch into the service mode. When the 2N® NetStar

light goes off on the BRI card, remove the card from the rack and check the jumper configuration. For a correct jumper placing use the sticker on the BRI port.

Set correct communication protocol in the configuration tool. Go to 2N® NetStar

the menu in and choose the port to be configured. Set the Boards Hardware and tabs for this port. When you are in the Stack menu, set Virtual port Stack

the interface mode to NT and mode to PTP.


To apply your new configuration, save the changes to using the saving 2N® NetStar

icon (or Ctrl+S). To configure the interface into the TE mode, take the same steps and set the jumpers and interface mode to TE.


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2.

3.

LCR CreationThe final configuration step is to create the LCR rules and configure the interfaces to work properly according to these rules.

Our task is to enable all outgoing calls to be passed to GSM and all incoming calls to be played the DISA welcome note or passed to the PBX IVR.

Outbound callsWe need to take three steps for outbound calls:

Create a GSM bundle responsible for a correct and well-balanced use of all GSM modules.

Create a router responsible for routing calls to the GSM bundle.

Assign this router to a virtual port connected to the PBX.

Create GSM bundle

Go to , click on the right mouse button and Routing – Routing objects – Bundleschoose to create the default set of bundles. One of them is called GSM Defaultand filled with all GSM ports.


Make sure that the count of the GSM ports in the GSM bundle matches the count

of ports available in .2N® NetStar

Configure the bundle – set the allocation strategy to .Cyclic

Create router

Go to , click on the right mouse button and choose .Routing – Routers Add


Fill in the router name and keep the type selection.Called number

Add 2 rows as shown in the figure below (click on the right mouse button and choose ).Add

Assign router to PRI/BRI port

Go to the and choose a port connected to the PBX.Hardware – Boards

On the bottom side of the configuration tool choose the tab, then Virtual port and finally .Properties Routing

On the tab set , Type to Router and Id to your router.Routing From port


1.

2.

Save your new configuration to using the icon.2N® NetStar

Now NetStar is properly configured to pass calls from the PBX to GSM or PSTN 2N®

through a bundle of GSM ports.

Inbound callsWe need to take two steps for inbound calls:

Create an incoming router responsible for routing calls into the connected PBX

and assign the router to the virtual port through which is connected 2N® NetStar

to the PSTN (GSM ports).

Set the DISA function for processing incoming calls.

Create router

Go to , click on the right mouse button and choose .Routing – Routers Add


Fill in the router name and keep the type selection.Called number

Suppose that the PBX PRI port cannot be re-programmed. In this case you have to send a call request in the same format as the PSTN. Suppose the company number is 020123xxx. When DISA passes the digits to your router, you have to take into account that the user can dial the number as a full PSTN number (first line) or as a short extension number (second line). In the latter case, you have to add 020123 to make the PBX receive the number as from the PSTN. In both the cases, the call will pass to the PRI interface that is connected to the PBX.

Configure DISA for incoming calls

Go to , click on the right mouse button and Routing – Routing objects – DISAchoose .Add


Name your new DISA and set it as shown in the figure below.

Assign DISA for all GSM channels

Go to .Virtual ports – Default OUT – Properties – Routing

On the tab set , Type to DISA and Id to your new DISA.Routing From port

Save your new configuration to using the icon.2N® NetStar


Now NetStar is properly configured to answer incoming calls from GSM, play 2N®

DISA to them and pass dialled numbers to the PBX.


2N® NetStar Billing Data

A billing line is an ASCII character string of variable length, with each line ending in CR LF (or optionally, LF in LINUX). A billing line is divided into sections by colons ‘:’ and individual parts of each section are divided by ‘-’. Missing data may be found by finding the respective position of ‘-’ or ‘:’, which remains even if the data are missing.

The billing line is divided into the following sections: record identification : date and time : caller A : caller B : status : billing data

Billing line example

34-28:1.3.2010-9.31.43:I-N6-201-Alice-201-200:O-N5-201-Bob-200-200:N:2-2-0-N-0-2

The importance of individual sections is described in the following chapters :

Record Identification

Date and time

Caller

Status

Billing data


Record IdentificationThe record identification contains two numbers: I-C

I – represents an unambiguous number of line record (32-bit number), used to store the REFERENCE item from the billing record. The entry is used for the clear identification of a billing line.

C – call reference; used for matching multiple lines related to a single call; an unambiguous number generated upon call initiation. For example, when calling a group, an number of lines for each extension within the group will be generated, nwith the number being identical for all of these lines.


Date and timeThis section has following format: day.month.year – hour.min.sec

day.month.year – represents the date entered at the end of the billing line, i.e. the charging date.

hour.min.sec – represents the time of call termination; the time of call initiation may be obtained by subtracting the ringing time and the call time from this number (and/or date).


CallerThere are two callers in every call, A and B. The call may either be placed through a port or a DISA-type service. The first letter in the section determines the type of caller. There are several caller types, always distinguishing between an incoming and outgoing call. It might seem that caller A is always incoming and caller B always incoming, but this is not always so, especially due to call switching. Let us imagine the following situation: an operator calls an outside number, speaks for a while, then dials another outside number and puts them through together. Both callers still have to be charged as outgoing calls. The called to be charged for the call must be determined by the billing software.

O – caller is an outgoing call on a port.

I – caller is an incoming call on a port.

S – caller is a service.

D – caller is a DISA.

C – caller is a .2N Contact Centre Solution®U – caller unknown; this letter should never appear.

Call on port

In case of a call on port, the section format is following: O[oIi] – A[N]port – realclip – user – CLIP – CPN

port – ID of port through which the call was made; the letter before the port indicate whether the port has been accounted or not. Therefore, A means Accounted and N m e a n s N o t A c c o u n t e d .

– r e a l c a l l e r i d e n t i f i c a t i o n .r e a l c l i p – n a m e o f u s e r p l a c i n g t h e c a l l .u s e r

– caller identification; in case of an incoming call, a number which arrives in the CLIPport; in case of an outgoing call, a number sent to the port as an ID. This parameter contains the user CLI from NetStar version 3.0.0 (if call is routed from or to one of the u s e r ' s e x t e n s i o n ) .

– caller identification; in case of an incoming call, a number actually dialed by the CPNcaller; in case of an outgoing call, a number sent to the port as the target destination.

Service

In case of a call to service, the section format is following: S – service identifier – service parameter


Each service is identified by a 4-letter identifier and has its own parameter. A service parameter is a number generated for the service. The meaning of each number is

explained in the 2N NetStar service manual. The following 4-letter service identifiers ®have been defined:

FWUN – forward user – unconditional

FCUN – cancel user forwarding – unconditional

FWBU – forward user – busy

FCBU – cancel user forwarding – busy

FWNA – forward user – no answer

FCNA – cancel user forwarding – no answer

FCAL – cancel all user forwarding

ASST – take over from extension

ASUS – take over from user

ASGR – take over from group

ASMG – take over from my group

ALRM – alarm

ALCN – cancel alarm

CTID – DTMF to CTI

UREN - ring at call to user

URDI – do not ring at call to user

STLI – extension login

STLO – extension logout

STLG – extension login as guest

UCHP – change PIN

PRMS – private call from my extension

PRCL – private call

CTOR – call to order

PARK – park

UPRK – unpark

CFCD – create conference on demand

BSLI – bundle extension login

BSLO – bundle extension logout


BULI – bundle user login

BULO – bundle user login

ACLI – ACD group login

ACLO – ACD group logout

AGLI – all ACD groups login

AGLO – all ACD groups logout

VMSU – forward user to voicemail – unconditional

VMRU – cancel user forwarding to voicemail – unconditional

VMSB – forward user to voicemail – busy

VMRB – cancel user forwarding to voicemail – busy

VMSN – forward user to voicemail – no answer

VMRN – cancel user forwarding to voicemail – no answer

VMRA – cancel all user forwarding to voicemail

VMGR – record voicemail greeting

VMGD – delete voicemail greeting

VMGP – play voicemail greeting

CTOC – connect to call

ACPF – activate profile

DCPF - deaktivate profile

SUPR – set presence

STIM – set time

SDAT – set date

PRGR – record progress

PRGP – play progress

PRGD – restore default progress

CFAC – add to conference

CFRO – call to conference room

CLRE – enable CLIR

CLRD – disable CLIR

VMEN – voicemail entry

VMEA – authorized voicemail entry

SWRS – set switch router


SWRG – get switch router

EALR – emergency alarm

EALC – cancel emergency alarm

REUS – user reservation

CBIN – internal callback

LEXP – license expired

EMMO – emergency mode

RBST – ring back to extension

RBNU – ring back to user

DISA

In case of a call to DISA, the section format is following: D – DISA id – number dialed in DISA

Contact Centre Solution2N®In case of a call to CCS, the section format is following: C – CCS id – dialed number


StatusThe following call statuses are relevant:

Normal

N – The call was dispatched in the usual manner: i.e., caller A picked up the phone and dialed caller B’s number. In such case, caller A has an outgoing call and caller B has an incoming call.

Forwarded

F – C[I]r – In case of a forwarded call it is necessary to know the reference line from which it has been forwarded. Reference r is used for this purpose. In case there is a letter before the reference, it refers to the call; the letter refers to the billing line.I C

Switched

T – C[I]r – C[I]r – In case of a switched call, two references are required, with the same meaning as in the case of a forwarded call.

Message

M – indicates a message (SMS or email).


Billing dataThe last section of billing line contains following: ringing time – call time – tariff i m p u l s e s – p r i v a t e – o r d e r – c a u s e

– duration of ringing until call was accepted.ringing time

call time – call duration in seconds; 0 indicates that the call has not taken place, i.e. has been recorded as unaccepted.

tariff – number of tariff or pseudo-tariff impulses during a call (or during ring impulsestone as well in case of simulated ringing through DISA).

private – calls identified as private are marked P; all other calls are marked N.

order – an entry used to break down each line into individual orders, projects, etc. Before, during or after the call, the service will allow to identify the order to which the call should be charged. For example, Cornet phones will allow to select an order from an Order List in the phone menu.

cause – cause of call termination; a number from 0 to 255 (1 byte) which identifies the cause of call termination in accordance with standards ITU Q850 and ETSI 300.102.


2N® NetStar CTI documentation

This manual is available in standalone document: .2N® NetStar CTI Documentation

https://manuals.2n.cz/display/TNSCTI


2N® NetStar Assistant Manual

Content:

1. About 2N® NetStar Assistant Application

2. Launching Application

3. Quick Navigation

4. Telephone Parameter Settings

5. Tray Launcher

6. Advanced Settings


1. About 2N® NetStar Assistant ApplicationThis application helps users work with the PBX more efficiently giving them comfortable access to information on their telephone line traffic. It combines CTI telephony with telephone line operation allowing the user to monitor the call history, send and receive text messages, comfortably define profiles and time conditions, program telephone buttons or ringing modes, manage the user telephone directory and make use of many other convenient options.

The application is run in an Internet browser. The recommended monitor resolution is 1024 x 768 pixels. This manual will give you a complete guide to the application and quick navigation to selected items. Refer to the application HELP item for more details.






Safety


Warning


Caution


Tip


Note




2. Launching ApplicationHere is what you can find in this chapter:

2.1 First Launch

2.2 Main Toolbar Menus

2.3 2N® NetStar Assistant Server Installation

2.1 First LaunchEnter your PBX address as communicated by your IT administrator into a web browser. For example: , or, if alias is set for the defined IP http://192.168.100.100address by the network administrator, the access address may be http://netstar.

. Now fill in your username (as defined by the administrator), leave the company.compassword empty and press OK for confirmation.

You will be asked to change the password after the first login. However, you need not change the password after the first login, you can change it any time later. You will automatically proceed to the password editing page upon login until you change the password. In this menu, you can also select the language mutation for interface comunication. You can change either item any time later. Push the Modify button for confirmation.

http://192.168.100.100

http://netstar.company.com

http://netstar.company.com


2.2 Main Toolbar Menus

The main toolbar menus helps you browse through windows. 2N® NetStar Assistant

The figure above shows the standard main toolbar view with default user right settings, which may change according to the current user right settings. All the main toolbar menus are described below.

Listings – this menu provides call history, SMS history, telephone directory items

and a list of PBX service access codes. If defined in the user rights, 2N® NetStar

management of conference rooms and recorded calls can be displayed too. If the logged-in user role is Manager, telephone directories, recorded call and call history of the whole company are displayed.

User settings – this menu helps you manage your PBX account, personal profiles, PBX user right settings and ringing tone settings. If set so in the user rights, the PBX user hierarchy and sessions of currently logged-in users can be displayed in this menu.


Telephones – the menu provides all user terminal settings, including speed

dialling buttons, volume control and information to be displayed on , 2N® OptiSet

and digital telephones.2N® StarPoint 2N® OpenStage

Operator – this menu is accessible with authorisation only. You can define operator stations and station groups to be monitored, and transfer and forward calls between such stations using CTI in the Assistant. In addition, you can control bundles here.

Hotel – this menu is accessible with authorisation only. It is used for easy room station operation, emergency alerting and alarm clock control when the PBX is configured in the Hotel mode.

Configuration – this menu is accessible with authorisation only. It helps you

monitor licences, PBX HW and SW settings and 2N® NetStar Assistant

administration settings comfortably.

Favourites – here you can set direct access to all pages.2N® NetStar Assistant

Help – opens a Help window for the application.2N® NetStar Assistant

Matt Gilroy – displays the name of the currently logged-in user.

Profile: none – this menu helps you directly set a profile for the currently logged-in user, and, after clicking, you proceed to a profile adding and managing menu. You can also create user time conditions and assign them to profiles here.

Logout – use this option to log out when you have finished your work.

2.3 2N® NetStar Assistant Server Installation

The following window displays pre-defined basic port settings as used by the 2N®

and the currently used PC domain name. If the PC you are Netstar Assistant Servercurrently using for installation contains another application that uses the same ports (Microsoft IIS, e.g.), modify the ports to avoid collision. Remember to enter the server address with the correct port in that case. If there is no port collision, leave the pre-defined values unchanged and click on Next to continue.

To be used efficiently, the has to be installed on a pre-2N® NetStar Assistant Server

prepared server or server PC. Refer to the Administrator installation 2N® NetStar

directory for installation. The benefits of the use of the 2N® NetStar Assistant Server

include a higher number of concurrent connections to 2N® NetStar Assistant Server 2N

and secured https:// connection.® NetStar Assistant


2N ® NetStar Assistant Server InstallationHaving started the application, select a language for installator communication. This selection does not affect the resultant language of the web application, which is set in the connected PBX. Check the version in the next window for 2N® Netstar Assistantcompliance with your PBX firmware or ask your PBX administrator if necessary. Otherwise, your application will not work. The installation wizard will ask for agreement with the licence terms and conditions. Please read the terms and conditions and select Agree... and then click on Next.

Enter your name and company in the Personal data window and push Next to continue.

Now enter your PBX IP address as communicated by your administrator.2N® Netstar


The installation wizard will ask you for installation location for the 2N® NetStar

.Assistant Server

Caution

The installation requires 13.4 MB of free disk space at least.

It will ask you for location of the application shortcut icon in the Start menu and on

the desktop as required. Now the wizard is ready to install the 2N® NetStar Assistant

and displays a list of set parameters.Server


Check the set parameters carefully as you will not be able to change them after installation without reinstallation. In case you find a mistake, push Back to return to the window concerned. If all parameter settings are correct, push Install to install the

.2N® NetStar Assistant Server

The webserver service with the appropriate IP address is 2N® NetStar Assistant

launched automatically upon installation. If the automatic launch failed for some

reason, start the service manually. Now the is ready for 2N® NetStar Assistant Server

use.


2N® NetStar Assistant Server Use

Open the web browser to be used. The optimum option is Microsoft Internet

Explorer for the . Enter the IP address or the 2N® NetStar Assistant Server

active PC domain name into the URL row. For example:

Confirm the address. The security certificate approval page gets displayed. Add a security exception for this server. Now the login window gets displayed. Log in as described in Subs. 1.1.First Start.

Single PC Launch of Multiple 2N® NetStar Assistant Servers

This setting is not recommended but may be used if the following conditions are met:

The NetStar Assistant Servers may not be installed in identical 2N®directories!

The NetStar Assistant Servers may not use identical TCP ports!2N®

Warning!

If you install another with a 2N® NetStar Assistant Server

different IP address into a directory with your current 2N®

, the Apache web server function may NetStar Assistant Server

be blocked and none of the servers can probably work or be


be blocked and none of the servers can probably work or be uninstalled. In that case, contact your PBX administrator. Refer to for frequently asked questions or contact the 2N faq.2n.czTelekomunikace a.s. Technical Support department.

http://faq.2n.cz


3. Quick NavigationHere is what you can find in this chapter:

3.1 Call History

3.2 Personal Profile

3.3 Telephone Directory

3.4 Favourites

3.5 Access Password

3.6 Language Selection

3.7 SMS Messages

3.8 Application Help

3.9 Services

3.10 Call Forwarding

3.11 Ringing Tones

3.1 Call HistoryThe Call history section includes a menu that helps you select displayed items. It is not

a filter but a function that deletes selected items from the 2N® NetStar Assistant

application and telephone menu listings leaving them in the AOC 2N® StarPoint

record for later billing purposes. This function can only be used for the data that are available to the application at the moment. This means that if the Do not show outgoing stations function is enabled, for example, outgoing calls are not visible in the listing. However, any outgoing call made from the station will be displayed in the listing again.

Click on the Listings menu, or on Homepage and then on Call history to view your call history. The list of calls shows missed, incoming and outgoing calls including call duration and data on the calling/called users. You can send an SMS or call to a selected contact, or hide a selected list row.


You can filter calls according to variable parameters to find calls according to a time period or number more easily and quickly than by sequentially searching the list. Filtering helps classify missed, outgoing and answered calls, calls according to the opponent's name and called/calling number. The most frequently used time periods are pre-defined such as the last two days, this month, or as you select by specifying the period beginning and end. Click on the Use filter to activate the filter. Click on Abort filter to cancel an inconvenient filter and set a new one. You can hide the function again after filtering by pushing the Hide filters button.


Caution

Once selected and thus non-displayed calls cannot be redisplayed.

3.2 Personal ProfileClick on the User settings menu and then on Personal profiles, or on the Profile toolbar item to work with user profiles. You will be forwarded to the Personal profiles page. There is a table with your profiles in the upper part of the screen, where you can activate, edit or delete a profile. Click on Add personal profile to create a new profile. The maximum count of profiles per user is eight.

Press Edit to modify an existing profile or set a new one. You get into the Modify personal profile window, where you can change the profile name and/or number. Doing so, make sure that the profile numbers are not identical. You will be notified of an error if any. In addition, you can make assignment to bundles that have been made accessible to you by the Administrator or bundle Manager. Once defined, the user call

message will be displayed to the calling users on their key system 2N® StarPoint

phones. Once defined, the state will be displayed in the connected 2N® NetStar

and applications. It can also be used for state Operator 2N® NetStar Communicator

mapping according to MS Outlook calendars in the above mentioned applications.


Once defined, the VoiceMail welcome note will be played to the people using your VoiceMail system. Use the Call recording item to specify the type of recording for a selected profile. Click on Modify to confirm and save the changes. If you are not sure, click on Default to restore the factory settings.

You can also enable and disable profiles using time conditions. One profile can be assigned to one time condition. One and the same profile, however, can be assigned to different time conditions. Select the Switch profiles automatically option to switch user profiles according to time conditions. Click on Assign profile time conditions and then select the time condition and the profile for assignment. There is a list of all PBX time conditions in the bottom part. Push Add to save the setting.


Caution

If the Switch profiles automatically option is enabled, the profiles switch according to the rules defined in the Profile time conditions window.

However, by changing a profile manually or using the 2N® NetStar

, a service, a key system phone button or the Assistent 2N® StarPoint

menu you deactivate the automatic profile switching and have to re-activate the function if necessary.

If no suitable time condition is available to you, ask your administrator to 2N® NetStar

create one or create a time condition of your own. To do so, click on the Time conditions button to display the list of all PBX time conditions. Then click on Create new time condition, enter your condition name and push Add for confirmation. However, the validity of the time condition you have created is Never. Therefore, click on Edit to open the Modify time condition window, where you can change the name and, particularly, the validity term for your time condition. Every time condition may have multiple validity terms but you need just one for your needs. Push Edit to open the Modify time interval window.


1.

2.

3.

4.

5.

Set the time condition validity intervals according to your needs in this menu. Select any item from the available options. The time condition validity shown in the figure below includes every workday plus Sundays, every month and whole day from July 16, 2010 till January 1, 2011.

The following rules have to be obeyed while setting validity:

If used, the 'from – to' data must represent valid days and time values. If both values ('from' and 'to') are defined, then the 'from' value must not be higher than the 'to' value.

If a bank holiday is selected, no other selections may be made (except for interval negation).

If a day is selected, the 'from' and 'to' values must range within 1–31 and the 'from' value must not be higher than the 'to' value.

If a month is selected, the 'from' and 'to' values must range within 1–12 and the 'from' value must not be higher than the 'to' value.


5.

6.

If a time value is selected, the 'from' and 'to' values must range within 0–23 for hours and within 0–59 for minutes and the 'from' value must not be higher than the 'to' value for the whole time value (hour + minute).

If negation is selected, validity of the whole time conditon is negated.

3.3 Telephone DirectoryClick on the Listings menu and then on the Telephone directory item to view, modify and use the telephone directory. Your user telephone directory (phone book) including the telephone directories assigned to your group gets displayed. These telephone directories are assigned by the PBX administrator. You can edit your telephone directories entries and, if you have higher rights, the group telephone directory entries. Move to the Own numbers window to view your telephone directory exclusively.

A filter is available for easier telephone directory search. Filter the items according to user names or station numbers. An initial contact letter filter accelerates searching.

Click on Add telephone number to add a number and enter the contact name and number in a new window.

Caution


If the required prefix is not defined, enter numbers including the PBX DID. Example: when you are calling a mobile number via '0', enter

or set . Make sure that the DID has been set 0774236891 Prefix 0properly. Otherwise, you will not be able to call the contact using CTI in

.2N® NetStar Assistant

Select the Ringing tone parameter to assign different ringing tones to the contact selecting an item from the list of pre-defined PBX tones. If you do not select any item, the standard ringing tone will be used. Select the Allowed numbers (White list) or Barred numbers (Black list) for the contacts in the forwarding section or set such call forwarding option that will ensure that your call will be answered in the event of your line error ir unavailability. Refer to the figure above for such setting.

3.4 FavouritesFrom now on the added page including your footnote will be displayed whenever you move your mouse cursor onto the menu.Favourites


The application offers you the option to include frequently used 2N® Netstar Assistant

pages among your favourites. Being on the selected page, move your mouse on the menu to open a pull-down menu and select . The window Favourites Add to favourites

as shown below will open with the pre-defined page URL within 2N® NetStar Assistant

and an editable description (please do not change URL). Push Add for confirmation.

Take the same steps to remove or rename an item, but select this Organise favouritestime.


3.5 Access PasswordClick on the menu and then on the item to change your User settings Settings

password within . Enter the existing password and then the new 2N® Netstar Assistant

password twice. Push Add in the right-hand bottom corner for confirmation.


3.6 Language Selection

Click on the menu and then on the item to modify the User settings Settings 2N®

language. The language options are in the upper part of the window. NetStar AssistantYou can also set the default page that will open upon login.


3.7 SMS MessagesClick on the menu and then on to send/receive a text message Listings SMS messagesfrom/to any of your stations. The following list of received SMS gets displayed. Click on Send SMS in the right-hand bottom corner to send a new message. To reply to a text message, click on the SMS row, mark the message as 'Read' with the pen icon or remove it with the rubber icon.

If you select send or reply, the following window will be displayed. For reply to a received SMS, the SMS recipient's number is pre-defined. A text message may contain up to 480 characters and diacritics may be used.

The telephone number is either pre-defined, or can be entered manually or using the telephone directory, which is available when you click on the link next to the telephone number field. The telephone directory window is displayed lower and can be searched through if it contains a high number of entries using the filters located in the lower part of the window. Filter the items according to the contact name and number. If the search is unsuccessful, try searching without diacritic symbols.


SMS sending may take some time depending on the PBX load. If a message cannot be sent from the application, contact your PBX administrator.


3.8 Application Help

Click on the toolbar menu to open a window with details on all Help 2N® NetStar

functions. Thus, you can easily use the application and consult Help if you Assistantare beginners.

Click on the Close button or close the active browser window to quit Help.


3.9 ServicesClick on the menu and then on to display the long list of PBX service Listings Servicesaccess codes. Or, use the below shown list of services, which, however, need not be up-to-date as the service table is downloaded from the PBX and contains default access codes. Some codes may be changed by the Administrator and not match the table items. Therefore, you are recommended to consult the up-to-date list of services

in . NetStar2N® Assistant

Services

Prefix Name

*21 User call forwarding

*22 User call forwarding if busy

*23 User call forwarding if no response

#20 Cancel all user call forwardings

#21 Cancel user call forwarding

#22 Cancel user call forwarding if busy

#23 Cancel user call forwarding if no response

** Take over station

## Take over user

#* Take over group

*# Take over my group

*50 Alarm-clock

#50 Cancel alarm-clock

*8# DTMF to CTI


Prefix Name

*55 Ring at call to user

#55 Do not ring at call to user

*66 Station login

#66 Station logout

*67 Station login as guest

*70 Pin change

*71 Private call from my station

*72 Private call

*73 Call to order

*42 Hold

#42 Return from hold

*1# Conference

*64 Login station to bundle

#64 Logout station from bundle

#30 Reset all forwardings into voicemail

*31 Set unconditional forwardings into voicemail

#31 Reset unconditional forwardings into voicemail

*32 Set forwardings if busy into voicemail

#32 Reset forwardings if busy into voicemail


Prefix Name

*33 Set forwardings if no response into voicemail

#33 Reset forwardings if no response into voicemail

*65 Login user to bundle

#65 Logout user from bundle

*35 Record voicemail greeting message

#35 Delete voicemail greeting message

*36 Play voicemail greeting message

*9# Connect to call

*60 Activate profile

#60 Deactivate profile

*61 Set presence text

*11 User call forwarding

*12 User call forwarding if busy

*13 User call forwarding if no response

#10 Cancel all user call forwardings

#11 Cancel user call forwarding

#12 Cancel user call forwarding if busy

#13 Cancel user call forwarding if no response

*51 Set time


Prefix Name

*52 Set date

*56 Record user progress tone

#56 Play user progress tone

*57 Delete user progress tone

*0# Add to conference

*2# Call to conference room

*43 Enable CLIR

#43 Disable CLIR

3.10 Call ForwardingClick on the menu and then on Forward calls in the lower part of the User Settings

page to forward calls from a station. Thus you display a Modify user parametersconfiguration window for forwarding calls to a target destination or VoiceMail and set the no answer timeout. Push Modify to save the parameters.


Always to – Forward always – all incoming calls are forwarded regardless of other call forwarding settings in this subtag. Top priority.

On busy to – Forward on error or busy – incoming calls are forwarded if the user is busy or routing fails.

At no answer to – Forward at no answer – incoming calls are forwarded if the user does not answer.

No answer timeout [ms] – the timeout after which the call is forwarded to a selected destination.

Caution

Call forwarding to VoiceMail has a lower priority than the standard forwarding type. If you set VoiceMail forwarding and then the standard forwarding mode, VoiceMail forwarding is not cancelled but not applied. When the standard forwarding mode is deactivated, VoiceMail forwarding is recovered.

Caution

Moreover, it applies that if station A calls are forwarded to station B, station B can call station A without being forwarded. This feature is called .Boss-secretary


3.11 Ringing TonesClick on the menu and then on to modify a ringing tone User settings Ringing tonesfor any of your telephones or assign a special ringing tone to a calling user. You will be notified upon your first access that there is no item in the database and you will see the Add ringing tone in the right-hand bottom part of the screen. Click on it to display the following window for assigning ringing tones to events (defined calling number, station, user, etc.). The tone defined here is applied to all stations of the user!

Click on the menu and select the required ringing tune in the tag Telephones Ringingfor a station. Refer to Subs. for more details.4.2 2N® OpenStage Button Configuration


4. Telephone Parameter SettingsHere is what you can find in this chapter:

4.1 2N® StarPoint Button Configuration

2N® StarPoint Extending Button Module Configuration

4.2 2N® OpenStage Button Configuration

2N® OpenStage Extending Button Module Configuration

4.3 Digital Key System Terminal Parameter Settings

4.4 Other Terminal Parameter Settings

4.1 2N® StarPoint Button Configuration

Click on the menu and select the required station to change the Telephones 2N®

key system phone programmable buttons. The following menu gets StarPointdisplayed.

Click on the button to be configured to open a function assigning window.


Push Modify to save the setting. Push Default to undo the changes unless saved.

The menu provides a convenient option to copy buttons and/or parameters to/from

other phones of the user or other PBX users. Click on any option to open 2N® NetStar

the following window (or a similar one depending on your selection).


Caution

The user must be assigned the required level of authorisation in order to

copy buttons and/or parameters to/from the PBX.2N® NetStar

Having programmed all required telephone buttons, select the Prepare to print option to generate a PDF file that allows you to print out or save the button tagname label. A printed label can be cut out and inserted under the telephone slide.

2N® StarPoint Extending Button Module ConfigurationClick on the menu and select the required station to modify the function Telephones

of the extending module buttons. Click on the Button module tag to 2N® StarPoint

display the following menu.


Note

Refer to the instructions included in Subs. 4.1 Nastavení tlačítek telefonu

for further settings.2N StarPoint®


4.2 2N® OpenStage Button Configuration

Click on the menu and select the required station to modify the Telephones 2N®

button functions. The following menu gets displayed.OpenStage

Note

Refer to the instructions in Subs. 4.1 2N® StarPoint Button Configurationfor further settings.


2N® OpenStage Extending Button Module Configuration

Click on the menu and select the required station to modify the Telephones 2N®

extending module button functions. Click on the Button module tag to OpenStageopen the following menu.

Note

Refer to the instructions in Subs. .4.1 2N® StarPoint Button Configuration


4.3 Digital Key System Terminal Parameter SettingsClick on the menu and select the required station to set parameters for Telephonesthe digital key system terminals. The following five-item menu gets displayed.

Basic

Station active – use the option to mark a station active or inactive. An inactive station is inaccessible for other users (all incoming calls are terminated), but can set up calls.

Mobility Extension – use the option to enable station ringing if the call is routed to the station user. If this option is not checked off, the station is not alerted when the call is routed to its user but is alerted when the call is routed to the station itself.

Enable CLIR – use to parameter to restrict the calling line identification. You can also use a service or a programmable button function of the key system phone to enable CLIR.

Alarm clock – use the option to set the alarm service for a user station. Click on Set alarm clock to display a window to set the date, time and user station to be

called at the set time by the PBX. If the first incoming call is not 2N® NetStar

answered, two more calls will be made. Push Cancel alarm clock to deactivate the alarm.


Note

A user with the required level of authorisation may set the alarm clock

for other users too.2N® NetStar

Ringing

Default ringing tune – set the standard ringing tune.

Message tune – set the ringing tone for a received message.

Intercom tune – set the ringing tone for an incoming Intercom call.

Volume


Key click volume – set the handset or HandsFree volume for a key click.

Handset volume – set the handset volume.

HandsFree volume – set the HandsFree/Speakerphone volume.

Ringing volume – set the ringing volume.

Display

LCD contrast – set the phone display contrast.

Time format – set the time format to be displayed: 24 or 12 hours.

Key timeout – set the delay between subsequent characters while editing.

View


Call history view type – set the viewing format for call records.

Telephone directory view type – set the viewing format for telephone directories.

Message view type – set the viewing format for information on received messages.

View type at relax – set the viewing format for information on the display bottom row when the telephone is at relax.

Incoming call view type – set the viewing format for information on the display upper row during ringing.


Note

Click on Store telephone parameters to save the key system phone settings.

4.4 Other Terminal Parameter Settings

Basic

There are limited options for other telephones that can be connected to the 2N®

PBX. Click on the menu and select the required station. The NetStar Telephonesfollowing menu gets displayed with one or two tags (depending on the telephone type).

This tag is displayed for all other telephone types that can be connected (IP, analogue, ISDN and GSM).

Station active – use the option to mark a station as active or inactive. An inactive station is inaccessible for other users (all incoming calls are terminated), but can set up calls.

Mobility Extension – use the option to enable station ringing if the call is routed to the station user. If this option is not checked off, the station is not alerted when the call is routed to its user but is alerted when the call is routed to the station itself.

Enable CLIR – use to parameter to restrict the calling line identification. You can also use a service or a programmable button function of the key system phone to enable CLIR.


Alarm clock – use the option to set the alarm service for a user station. Click on Set alarm clock to display a window to set the date, time and user station to be

called at the set time by the PBX. If the first incoming call is not 2N® NetStar

answered, two more calls will be made. Push Cancel alarm clock to deactivate the alarm.

Note

A user with the required level of authorisation may set the alarm clock

for other users too.2N® NetStar

Ringing

This tag is displayed for analogue telephones only.

Default ringing tune – set the standard telephone ringing tune.


Default ringing tune – set the standard telephone ringing tune.

The other parameters displayed here are not applicable in analogue telephones.

Note

Click on Store telephone parameters to save the telephone settings.


5. Tray LauncherHere is what you can find in this chapter:

5.1 Tray Launcher Installation

5.2 Tray Launcher Use

5.3 Tray Launcher Function

5.1 Tray Launcher InstallationAn option called is available in all basic menus below the Homepage Tray launcher

option. This item is used for downloading and installation of the 2N® Netstar Assistant

launching program.

Click on the Tray launcher to display the program downloading page including a Help for basic settings. Having read the instructions, push Download in the right-hand bottom corner to display the following window (or a similar one depending on your web browser).

We recommend you to save the installation file on your PC disk before installation. Open the saved file to start the installation wizard.


The system security window can be displayed depending on your PC settings. Do not hesitate to push the Run button to launch the installation wizard. You will be asked to

select the installation wizard language, which does not affect the selection of the 2N®

language.Netstar Assistant

Now you are advised to check the version in the following 2N® Netstar Assistant

window for compliance with your PBX firmware as communicated by your PBX administrator. Otherwise, your application will not work. The installation wizard will ask for agreement with the licence terms and conditions. Please read the terms and conditions and select Agree... and then click on Next

The wizard will ask you to complete your personal data in the following window. They are mostly pre-entered and so click on Next to continue.

The most important step is to set the IP address of the PBX on which the 2N® Netstar

application is running and other account parameters such as your login Assistant

name and password as usually entered for the web interface. 2N® NetStar Assistant

See the figure below.

Enter the number of your station to control the station via the tray launcher and display calling line identifications. The tray launcher will not display information on the line state without this number. Fill in the domain on the following page if you log in using the Active Directory. If not, leave the field empty.


In the next step, you will be invited to enter the IP address or DNS name of the 2N®

. Please enter the address (e.g. 192.168.1.200, or NetStar Assistant Server assistant.firma.

, etc.) in case you are using the and push Next to cz 2N® Netstar Assistant Server

continue. The PBX administrator will advise you whether it is necessary to fill in the data.

Now you will be asked whether the KeepAlive packets should be sent. If you have instructions from the PBX administrator, modify this entry. If not, leave the field unchanged and push Next to continue. You will be asked about the installation disk

location for the .2N® NetStar Assistant Server

Caution

The installation requires 2.8 MB of free disk space at least.

The wizard will ask you for location of the application shortcut icon in the Start menu

and on the desktop as required. Now the wizard is ready to install the 2N® NetStar

and displays a list of set parameters.Assistant Server

http://assistant.firma.cz

http://assistant.firma.cz


Check the set parameters carefully! In case you find a mistake, push Back to return to the window concerned. If all parameter settings are correct, push Install to install the

.2N® NetStar Assistant Server

Note

Should there be a need to change the Tray launcher later, you can modify all the parameters using the Settings menu in the application.

An icon will be displayed on the main panel upon installation, which will enable the 2N® application to be launched without entering the IP address into your NetStar Assistant

web browser. Click on the icon with the left mouse button to launch the application.


If a message appears on your PC after power up about insecure software, click on a request in the left bottom corner to avoid future displaying of that message.

5.2 Tray Launcher UseClick on the Tray launcher located on the main panel next to the clock with your left

mouse button to run the application.2N® NetStar Assistant

Click on the Tray launcher with the right mouse button to display the context menu for making calls, settings and other application options.

If the Tray launcher is not displayed on your main panel, it may be hidden (if set so in your operating system), or the application is not running. Launch the application using the Start menu or from the desktop through the application representative depending on the installation setting.


5.3 Tray Launcher Function

Calling Line Identification PresentationDuring an incoming call, the calling user number and, if registered in the PBX, name are displayed. Set the number of the controlled station to make the function work (refer to Subs. .).5.1 Tray Launcher Installation

Missed Call Message

The Tray launcher notifies you of a missed call with a message. Launch 2N® NetStar

or just make a call to process the missed call easily.Assistant


Calling through Tray LauncherIf you decide to make a call directly, the following menu with the pre-entered missed station number is available to you. Or, you can call another number from your telephone directory or just type the number into the field. If you click on the Tray launcher with the right mouse button and select the Call option, the same window will get displayed.

Note

When you click on Call, your telephone apparatus is shortly alerted (the

telephones are automatically switched ino the HandsFree 2N® StarPoint

mode) and connection to the required station is set up. If you use an analogue or IP telephone, answer the phone to initiate call setup.


Send SMSClick on the Tray launcher in the main panel with the right mouse button to send an SMS message through the Tray launcher. Select the Send SMS option in the menu displayed. Now the SMS sending window opens and you can select the SMS recipient. Type the text including diacritic symbols if necessary and push Send. SMS sending may take some time depending on the PBX load.

Change Personal ProfileClick on the Tray launcher with the right mouse button and select the User profiles option. The window displays the up-to-date list of your profiles. You cannot create profiles but you can switch them.

Caution

If enabled, automatic profile switching will be cancelled and have to be

re-activated using .2N® NetStar Assistant



6. Advanced SettingsHere is what you can find in this chapter:

6.1 What to Find in Advanced Settings

6.2 2N® NetStar Assistant Administration

6.3 Admin Login

6.4 Administration by 2N® NetStar Assistant

6.5 User Settings

6.6 User Management

6.7 Configuration Management

6.8 Hotel Management

6.9 Operator Management

6.10 Operator Management - Bundles

6.11 Conference Room Management

6.12 Recorded Calls

6.13 Tray Launcher Settings

6.1 What to Find in Advanced SettingsThis section of the manual provides information on PBX administration and advanced

licenced features of . The first subsection describes 2N® NetStar Assistant 2N® NetStar

administration. The following subsections explain how to use Assistant 2N® NetStar

for your PBX administration and describe licenced services and their use. AssistantFinally, you will know the advanced settings of the Tray launcher.

6.2 2N® NetStar Assistant Administration

All that is mentioned in this subsection relates to your programming tool. 2N® NetStar

Use the menu to the Network / Service setting / System services enable/disable

internal web server for .2N® NetStar Assistant


Refer to the menu for basic settings Global data / Assistant / Administration settings

of .2N® NetStar Assistant

Confirm deleting – use this option to enable deleting call history records. If the option is ticked off, you are invited to confirm deleting.

Default language – set the default application language. Czech, English and Finnish are available now.

Image directory – enable the use of one of the pre-defined image sets.

CSS style filename – set the CSS style to be used in the application.

Maximum user session time [min] – set the maximum time for a user session after which an idle user will be logged out.

Caution

In case you use the , make sure that the server 2N® NetStar Assistant Server

installation matches the CSS style. Otherwise, display errors might occur in the application web interface. Be sure to keep the basic rules.

Use the standard for Standard and Funny 2N® NetStar Assistant Server

displays.

Use a tailor-made installation for the customised appearance.


Use a tailor-made installation for the customised appearance.

You will specify the global behaviour of with these settings. To 2N® NetStar Assistant

set the same parameters for some users only use the same menu but from the user parameters: . Create individual Users/ Users and groups / Group / User / Assistantsettings in this menu and then specify the Assistant parameters related to the user only.

Find a list of all currently connected application users in the Global data / Assistant / menu. The meanings of the columns are as follows:User sessions

Login name – the user to which the given session corresponds.

ID session – identification of the user session within the database.

Last access time – time of the last user activity within the session.

User rights is the last important menu. User logins have to be created to enable 2N®

to be used. A role has to be assigned to each login. The following NetStar Assistantroles with different levels of authorisation are available:

Vice Admin – is a login with all rights except for and Delegate logins Manage .own group

SUPER – is identical with Admin as to rights.

Vice SUPER – is a login with the same rights as are assigned to Vice Admin.

Manager – is intended for the user that needs the right to manage logins of all the other users in the group and so has the right to delegate and manage its own group.

Deputy Manager – this login allows the user to manage its own group.

User – represents a general user login.


When a user is selected, the list of all users assigned to a group including their 2N® rights is displayed to the right of the tag. Such summary is NetStar Assistant Assistant

useful for setting similar rights in user groups. The tables of rights has the following sections:

User management – helps view and change settings of other users.

Telephone directory management – helps view and change telephone directories of other users.

Call history management – helps view call histories of other users.

Telephone management – helps view and change telephone settings of other users.

Station management – helps view and change station settings of other users.

Global configuration management – helps view and change global configuration settings.

Operator management – helps view and change operator settings.

Alarm clock management – helps view and change alarm clock settings.

SMS management – helps view and change SMS mesages of other users.

Conference room management – helps view and change conference rooms.

Hotel view – helps view and manage alarm clocks and emergency alerts in hotel structures.

Recorded calls – helps view and manage user calls recorded.


6.3 Admin Login

Admin is a special login to . No rights can be assigned to Admin 2N® NetStar Assistant

in the programming tool. This login has the top priority automatically and cannot be removed. The Admin login data are the same as those for login of the programming tool to the PBX. It is launched in English and the following menu is available.

User settings – Subsection 6.05.

Telephones – Section 4.

Configuration – Subs. 6.07.

Help – Subs. 3.08.

Admin – displays the currently logged-in user.

Logout – quits work with .2N® NetStar Assistant

6.4 Administration by 2N® NetStar AssistantRefer to the following subsections for the user settings and/or relevant modifying menus. The login of user with the and is shown 'novy' Manager role full Assistant rightsfor illustration.


You can use for an easy and comfortable configuration and 2N® Netstar Assistant

administration of most user functions. Of course, you need the PBX administrator for configuration, but most user settings can be delegated to the users that are assigned the and the necessary rights.SUPER/Manager/Vice Admin roles

6.5 User SettingsThe User settings page may look different if different rights are assigned. For the maximum count of items see the figure below. Let us explain the meaning of each item. Select the menu and click on .User settings / User list / Select user Edit


Group – specify the user assignment to a PBX group.

Login name – display the current user login.

User – change the user name for key system phone identification.

Alias – this parameter is used by external applications connected via the Application server. Alias in the PBX corresponds to the user name in the Active Directory. Alias and email enable the user identity to be checked in the exchange server for subsequent profile switching according to calendar events, for example. Refer to the Application server manual for more details.

State – set one of the pre-defined states to be used by cooperating applications for user state identification. The pre-defined states correspond to standard states in instant messaging applications.

Language – define the language for the selected user.2N® NetStar Assistant

Default page – set the default page to be displayed upon 2N® NetStar Assistant

launch.

Image directory – define a set of images to be used in .2N® NetStar Assistant

CSS style filename – define the appearance.2N® NetStar Assistant

User calling message – enter the text to be displayed to the calling users on their key system phones if they are calling a station of the user that is assigned this text.

E-mail – set the user electronic mail address for user voice message forwarding. If this option is blank, the VoiceMail forwarding and voice message forwarding cannot be used.


VoiceMail welcome note – enter the welcome note to be played to the calling user to announce VoiceMail forwarding within the user profile.

Call recording – define whether the user calls should be recorded (always, upon request) or not.

Call forwarding – refer to Subs. 3.10 for details.

Pasword change – refer to Subs. 3.05 for details.

Note

Furthermore, user rights can be assigned. See the rights and their meanings in the list below.

User management – helps view and change settings of other users.

Telephone directory management – helps view and change telephone directories of other users.

Call history management – helps view call histories of other users.

Telephone management – helps view and change telephone settings of other users.

Station management – helps view and change station settings of other users.

Global configuration management – helps view and change global configuration settings.

Operator management – helps view and change operator settings.

Alarm clock management – helps view and change alarm clock settings.

SMS management – helps view and change SMS mesages of other users.

Conference room management – helps view and change conference rooms.

Hotel view – helps view and manage alarm clocks and emergency alerts in hotel structures.

Recorded calls – helps view and manage user calls recorded.


6.6 User ManagementIf you are assigned the User management rights, a different structure will be displayed to you in the User settings menu. In addition to your own account, you can configure accounts of other PBX users using a few windows as described below.

User TreeClick on the User settings to display the User hierarchy and User tree, where you can see each user, its group and stations. Click on an item to display the item settings. You can change group names and click on stations to display their settings, refer to S. 4.

.Telephone Parameter Settings

Click on a user to open a special administration menu as shown in the figure below. It is called the User panel. There you can easily set basic parameters for an account, such as user name, language, password and profile, and enable automatic profile switching.

Note

If a user forgets the password, you can change the user password without knowing the old one. All you have to do is enter a new password, confirm and save the change. From now on the user may use this new password for login.


Click on Default to restore the default values or on Confirm changes to save the changes. Push the Previous and Next buttons to proceed to other PBX users according to the database sequence. The links in the right-hand screen section help you perform advanced account management. Click on the links to open the basic menu with settings as shown in the telephone directory menu below.

User list – you will be forwarded to the User list window, see below.

Call history – opens the call history window for the selected user.

Telephone directories – helps manage the telephone directory for the selected user.

Ringing tones – helps define ringing tones for the selected user.

User rights – displays the list of rights assigned to the selected user. If you are authorised to assign rights, you can do so here.

Telephones – displays a list of user telephones.

Stations – here you can easily make user station settings such as active/inactive state, ME enable/disable and Edit for complete station settings.

Recorded calls – displays a list of recorded calls for the selected user.


User GroupsThe User groups window displays all groups created in your PBX. You can edit the name and telephone directory for each group, add other PBX telephone directories and create a new general telephone directory.

User ListThe User list displays an alphabetical list of all PBX users. A link refers to the most frequently used parameters such as user panel, ringing tones, telephone directories, stations and user settings as described in Subs. .6.5 User Settings


6.7 Configuration ManagementIn the Admin login or if you are authorised to manage the global configuration, find the on the main toolbar and links to the following pages in it.Configuration tag

PBX – enter this section to display the PBX HW and SW information. The SW build includes, among others, your PBX firmware version. The HW build displays the

components of your system.2N® NetStar

Licence – display a list of available licences including their validity terms and histories. Display the licence downloading history to know when a licence was added.


Settings – display the administration settings. The parameters 2N® NetStar Assistant

defined here are saved in the PBX Global data section. Refer to Subs. 6.2 2N® NetStar of this manual.Assistant Administration


Time conditions – display the list of all your time conditions. You can edit 2N® NetStar

and delete all time conditions. The time conditions added by users are labelled with the respective user names in the parentheses behind the condition name.

6.8 Hotel Management

The Hotel view is a special licenced view of the PBX used when your 2N® NetStar

structure is a Hotel. This selection is made in the initialisation wizard upon the first PBX installation, or after the factory reset. The menu gives a clear overview of hotel floors and apartments/rooms. The tag provides two types of PBX view: Simple viewand .Hotel view

The displays all floors (groups) and apartments (stations) including their Simple viewprofiles, which can be modified. This view is useful, for example, for the reception clerk to see which room is currently vacant and set the reservation/cleaning request promptly.


The is the other option. This view displays a similar structure enhanced Hotel viewwith additional settings. You can unpack floors (groups) as you currently need them without searching the page. Moreover, the Emergency alarm control element is permanently available to all groups.

Refer to the figure below for the Hotel view.


Emergency AlarmPush the Set emergency alarm button and confirm the button settings in the next window to enable emergency alarm. The emergency alarm will start alerting in 20

seconds. is responsible for emergency alerting of all stations in 2N® NetStar Assistant

the group for which the alarm was enabled. The 20s timeout enables the emergency alarm to be cancelled in the case of error as it cannot be cancelled once alerting

starts. The moment all stations are answered and the voice message is heard, 2N®

considers the emergency alarm process terminated. If a station fails NetStar Assistantto answer or rejects the emergency alarm call, it will be redialled after a preset time interval, which is identical with the alarm clock interval, i.e. three times within three minutes. If the calls are still unsuccessful, the station will not be alerted any more.


Caution

Emergency alarm calls are not made to inactive stations!

Caution

If the PBX is switched off after alarm activation and before 2N® NetStar

alerting, e.g. due to power failure, remembers the 2N® NetStar Assistant

alarm activation and re-activates alarm in 40s after the PBX emergency power on. If the PBX is switched off during alerting, no alarm re-activation is made.


Station ManagementYou can use several controls to assign station parameters to apartments. See the figure below. Let us provide some details.

Active – define whether or not a station should be active. An inactive station is inaccessible to other users (all incoming calls are terminated), but can set up calls.

Language – select the language version for the key system 2N® StarPoint

phones.

Set alarm clock – push the button to activate the alarm clock service for a station. The alarm clock setting window gets displayed.

You can configure depending on the type of the station connected. Other settingsRefer to Section .4. Telephone Parameter Settings


6.9 Operator ManagementThe tag is subject to licence and you can display it if authorised so. The tag Operatorhas three sections: and are relevant for operator Station states Operator stationsmanagement and will be referred to in the next subsection.Bundles

Operator StationsDefine which of the stations should be operator ones. This means that they will be

monitored and their incoming calls will be processed by . There 2N® NetStar Assistant

may be up to 2 operator stations with an active queue. Ask your PBX administrator for the active queue setting if unavailable.

Station StatesEntering the menu for the first time you will probably find no stations Station stateshere. Therefore, click on Add monitored stations, enter the station name and push Add. You can display, name or delete any new station.

Now display the group of stations to be monitored. No item will be seen. Click on the button to open a window. Here select the group to which the Add monitored station

station to be monitored should be assigned, select a station and edit a note if necessary. Then push . The station will be displayed in the window of the Addmonitored station and can be deleted or modified.


The window also includes the option. Click on it to update station states Auto refreshautomatically. The minimum auto refreshing interval is 10s. Re-activate Auto refresh whenever you enter the menu.

Operator ManagementIf there is an incoming call to the operator station, a call queue will be displayed to the left. If you are equipped with a key system telephone and HandsFree, answer the calls using the green button next to the call queue. Click on the red handset button to reject or hang up a call. To transfer a call, tick off the call and select any of the stations monitored. You will be asked for confirmation in a new window and so confirm the selection to transfer the call.

A well-arranged list of stations is displayed to the right. States are colour-coded and include a current state note to let you know which stations are busy.


If the call queue is enabled in your PBX, the queuing calls are displayed at the operator station. Each of the queue calls has an arrow for you to prefer the call if necessary. See the figure above.

6.10 Operator Management - BundlesA bundle is a routing object, which enables the incoming call to be routed to one (all) of the objects included in the list of bundle objects. The object is selected according to the strategy selected. If the selected routing object is busy, the call is not necessarily terminated. Depending on the bundle setting, the call can be routed to the next routing object in the sequence after the object busy state identification or after a timeout. The PBX administrator defines the bundle parameters and let some bundles

be displayed in . A properly authorised user can then add users, 2N® NetStar Assistant

set the and manage numbers of the stations to be billed.No answer timeout

ACD group is the bundle name. There is an icon next to it for the No answer timeoutand accounted station settings.

Below are the users that are logged in to the bundle. Next to them you can use the call history icons and, using arrows, set user priority within a bundle and use the Editand buttons.Delete

Caution

The button is displayed only if the user has been added through Delete

. A user assigned by the Admin cannot be 2N® NetStar Assistant

deleted.


User Login to Bundle

Bundle – use the parameter to select the bundle for user adding.

User – set the user to which an incoming call or SMS message should be routed.

Order – set the bundle location for user adding.

Caution

If you leave the default value in the last-named parameter, the user will be added to the end of the list.

No Answer Timeout Setting

No answer timeout [s] – specify a timeout for transition to another bundle row at no answer.


Accounting Group Setting

Accounting group – enable adding of the selected group number to the accounting record for the selected object for cost distribution purposes.

User Logout Disable

Disable logout – if this option is ticked off, the user/station cannot log out using the button from bundle service. If the user tries to log out, an error Logoutmessage is displayed.

6.11 Conference Room ManagementUse this menu to configure conference rooms and define authorised users. The conference room is a conference with pre-defined participants so that the users do not have to be dialled using the service in advance. Additional participants may join the conference during the call of course. This function is subject to licence, so make sure that the available count of conference room licences is sufficient for successful operation (one conference room requires one licence).


The menu shows all conference rooms that were made visible to your group in 2N®

. If there are unused conference room licences, the NetStar Assistant Add conference button is available.room

The name and access code are displayed for each conference room. Use the *2# service to enter the conference room. When you get access to the access code #

conference room, alerting of the defined destinations will be initiated as pre-programmed and you will be played the voice message. You Welcome to conferencecan also edit (see below) the conference room and delete/add a conference participant.

The pre-defined conference participants are displayed below the name including their station numbers and states, where 'inactive' means idle and 'active' means taking part in the conference. You can edit and delete any destination.

Conference Room Adding and ManagementClick on the button to open the following window. Refer to the Add conference roomdescription below for details.


Name – enter the name of a new conference.

Access code – set a unique access code for each conference room for identification within the service.

Time condition – define a time condition to time restrict the use of a conference room. If no time condition is selected (---), the conference room is available continuously.

Access for selected only – use the parameter to restrict the conference room for selected users only. Access of users other than selected is rejected.

Conference Participant Adding and EditingClick on the icon to add a conference participant. You will proceed to the + Add

'Conference name' window. Refer to the description below for conference participantdetails.

Destination type – select a user, station or address.

Destination – select a particular station or user.

Dials others – set whether or not the user has the right to call a conference for the conference room.

Is dialled – set whether or not the user should be dialled during conference room calling.


6.12 Recorded Calls

User LoginThe Recorded calls menu provides a comfortable management of recorded calls. If you have the right to see the recorded calls, you can play and save them. If you have the right to edit, you can delete the recorded calls too. A deleted record is deleted from the PBX memory and/or connected storage too. You can also filter the calls in the Recorded calls window according to the opponent's name. Click on to Abort filterquit filtering.

Click on the icon to play and/or save a call record. A new window gets PLAYdisplayed with the required options.


Manager LoginIf you have the , you have access to the menu Manager login Company recorded callstoo. The list includes all users in the group and, by clicking on a user name, you will display all recorded calls of the user. Work with the menu as with the menu above.

Admin LoginIf you have the , you can access recorded calls too either in the Admin login User panelor in the window. The option is available to User hierarchy User groups Recorded callsyou in the window. Click on the figure to proceed to the list of all users in the RECgroup and then click on a user name to display the recorded calls of the user.


6.13 Tray Launcher SettingsThis subsection provides a summary of the Tray launcher advanced settings. Click on the with the right mouse button to display the menu and select the Tray launchersetting option to display the window. The window includes all Tray launcher settingthe parameters that you defined during installation. Refer to the text below for details and, for parameters relevant for the Tray launcher, examples.

Assistant IP address – enter the IP address or domain name of the PC on which

the is running as described in 2N® NetStar Assistant Server 5.1 Tray Launcher

.Installation

Login name – set a name for login to the web interface.2N® NetStar Assistant

Password – set a password for login to the web interface.2N® NetStar Assistant

Domain (optional) – enter this parameter only under the direction of your network administrator (for login only).

Monitored station number – enter the number assigned to your station.

Save and reconnect – press the button to save the changes into registers and restart the application.


Caution

Terminate and restart the application. Re-log in to upon 2N® NetStar

power up. This may take some time depending on the system load, typically 30s. Do not use the Tray launcher until the new setting is accepted and a message occcurs announcing that the station is logged in and monitored.

Cancel – leave the original setting and terminate configuration.

Advanced – open advanced settings for the Tray launcher. Refer to the text below for more details.

Help – open the Help window.

Advanced Settings

Automatically launch on Windows startup – tick off this option to activate Tray launcher automatically upon your PC power up.

Show missed call popup window – use this option to display a pop-up window in the middle of the screen to announce a missed call to the user.

Active call announcement type – choose one of the two available options according to your needs.

Balloon – display announcement in the form of a traditional system information balloon.


Window – display announcement in a window generated by the Tray launcher.

None – display no line status information.

Announce missed calls – specify the type of missed call announcement.

Always – the Tray launcher displays information regardless of the currently active call.

If no call is active only – the Tray launcher does not display information until the active call ends.

Open Application upon Incoming CLIPThis menu section helps you set the application function that is able to recognise incoming calls and identify the calling number, and open the appropriate application according to the available settings. There are 5 tags in the application, which means that up to 5 numbers can be specified and processed. The example below shows a simple and frequent use with a door access communicator.

Tip

Suppose we have a door access communicator with an IP camera, e.g. 2N

, with number 6254. The receptionist uses the Tray lancher to ® Helios IP

see the CLIP and all missed calls. Set the application as follows: enter the door communicator number 6254 into the 'upon CLIP' field and let run a program able to launch streamed video. To do so, specify the path to the program .exe file into Launch and enter the door communicator address and the video stream protocol as the parameter – rtsp://192.

. From now on, when this door access communicator makes a 168.10.63call, a video player will be switched on for the receptionist to see the person in front of the door.

rtsp://192.168.10.63

rtsp://192.168.10.63


AdvancedClick on the button to open an additional advanced setting menu of the Advancedapplication. Refer to the text below for details.

Assistant HTTP port – specify the port on which your is 2N® NetStar Assistant

working if you do not use default settings.

Get method – select the preferred method: either or .HTTP HTTPS

Run command – specify in which browser should be 2N® NetStar Assistant

launched by the left mouse-click on the Tray launcher.

PBX IP address – enter your IP address as advised by your network 2N® NetStar

administrator.

PBX bridge port – this parameter cannot be set. It specifies the library communication port for PBX communication. This port must be allowed if you use the NAT between the PC and the PBX.

Add DID – before numbers with min length – these parameters are tied together. They specify the DID to precede numbers with minimum length replacing the prefix used in the PBX.

Caution

Suppose your PBX DID to the GSM network is '0'. Set the Tray launcher to pre-dial '0' for you. Enter '0' into the field and '9' into the Add DIDnumbers with min length. Now you can call GSM numbers from the Tray launcher as follows: you dial 777515668 but 0777515668 will be actually dialled.

KeepAlive packets – specify whether or not the KeepAlive packets should be


KeepAlive packets – specify whether or not the KeepAlive packets should be sent.

Timeout (ms) – set the time interval between the KeepAlive packets.


2N® Contact Centre Solution Manual

The is an ideal Contact Centre solution for small and 2N® Contact Centre Solution

medium enterprises with two or more operators, comprising a software application

and hardware on a platform. The Contact Centre can be used not only by 2N® NetStar

companies with customer lines, but also by any service and contact departments, help desks and dispatching centres, i.e. wherever distribution and monitoring of incoming calls are needed.

You can log in to the via a web interface without 2N® Contact Centre Solution

performing multiple installations. Detailed statistic data are available providing you with a very clear insight in the Contact Centre’s efficiency from the viewpoint of operator statuses, campaigns, individual operators or calls. Access to such data can also be permitted for certain user roles only – Superadmin, Admin, Manager and Operator.

Content:

1. 2N® Contact Centre Solution Installation Wizard




Installation

Install the directly on a hardware PC or virtualisation 2N® Contact Centre Solution

platform (VMware, Oracle VM VirtualBox, etc.).

Caution

Make sure that the following minimum PC/virtualisation platform

configuration requirements are met while installing the 2N® Contact

to avoid system malfunction!Centre Solution

64bit processor

4 GB RAM

100 GB free hard disk space

Note

Make sure before installing the onto a 2N® Contact Centre Solution

virtualisation platform that VT-x or AMD-V or any other similar virtualisation function is enabled according to the hardware PC processor type.

Example of Virtualisation Platform Installation

Connect the received ISO file with the installation to the 2N® Contact Centre Solution

CD/DVD drive in the virtualisation platform that meets the minimum configuration requirements mentioned above. Set booting from the CD/DVD drive and start the virtual PC. When the ISO file has been read, the following screen will be displayed:


Select the first option on this screen and the first option on the next screen (Use all space) and click Next.


When the installation is completed, the virtual PC gets restarted. If the booting sequence was changed in the virtual PC configuration, we recommend you to return Booting from hard disk to the first place.

After booting from hard disk is completed, you will see the login window:

Log in to complete the installation process: login and password .admin 2n

Upon login, the post-installation script will start automatically to complete the 2N®

installation.Contact Centre Solution

Caution

The post-installation script takes a few minutes. Be very careful while entering data and do not interrupt or skip over the script to avoid the 2N

malfunction.® Contact Centre Solution

Language is the first script selection.

Having selected the language version, choose the type of user connection to the 2N®

: IP address or Domain name. If you select the latter option, Contact Centre Solutionyou will be asked to enter the domain name.

Caution

Make a corresponding record in your DNS server if the users are connected via their domain names.


If you select the IP address option, choose the interface to be used for connection (the IP address will then be entered into the Internet browser) and set the following parameters:

NS IP addresses for communication.2N® Contact Centre Solution

Port for communication. The default port is .2N® NetStar 6543

User name for authentication.2N® NetStar

User password for authentication.2N® NetStar

Note

We recommend you to create a special user in used only 2N® NetStar

for authentication with the .2N® Contact Centre Solution

The service will be enabled when all the required data have been set.


Press any key to complete installation. Now use any Internet browser for connection

and start using the . The configuration menu below will 2N® Contact Centre Solution

be displayed whenever Admin logs in.

Caution

Please set the correct time (via the NTP server - option 5, e.g.) before

using the as correct time setting is crucial 2N® Contact Centre Solution

for some functions.



First Login

Having completed installation, enter the server IP 2N® Contact Centre Solution

address or domain name into your Internet browser.

Note

It is recommended to use Internet Explorer 10 and higher, Mozilla Firefox 31 and higher or Google Chrome 31 and higher for your web login.

For the first login, use the server Superadmin account - 2N® Contact Centre Solution

name: and password: . Click the logged-in user name in the right-hand superadmin 2nupper screen corner to change the Superadmin password after login if necessary.

The start screen is displayed after a successful login.


Upload the licence before creating companies or 2N® Contact Centre Solution

projects.

The following data are displayed after licence upload:

SN - serial number2N® NetStar

Date - last licence verification date

Managers - count of Manager users

Detailed statistics - count of project statistics

Unlimited - licence time validity

Companies - count of available companies

Agents - count of Agents

Company CreationClick to display the following company parameters.Add


GeneralCompany - set the name of the company that operates the Contact Centre.

Opening time - set the Contact Centre working time beginning. Combine this parameter and the Working time (minutes) parameter to create a virtual Working time value. All incoming calls before or after the working time are automatically routed to

the default destination defined in . The working time also affects daily 2N® NetStar

statistics displayed in the Agent/Project or Status screens. The calls made before or after the working time are not included in the daily statistics.

Working time (minutes) - set the working time in minutes. Combine this parameter and the Opening time parameter to create a virtual Working time value. All incoming calls before or after the working time are automatically routed to the default

destination defined in . The working time also affects daily statistics 2N® NetStar

displayed in the Agent/Project or Status screens. The calls made before or after the working time are not included in the daily statistics.

Position range from - set the lowest possible position to which a user can log in. Combine this parameter and the Position range to parameter to define a range of positions, which is used for graphical depiction of the current user location only.

Position range to - set the highest possible position to which a user can log in. Combine this parameter and the Position range from parameter to define a range of positions, which is used for graphical depiction of the current user location only.

Statistics - select this option to make detailed statistics accessible and allocate one Detailed Statistics licence to a selected company. Detailed statistics are accessible to Managers and Admin users only.

Active - check off this option to make a company or project active. If a company

/project is active, project synchronisation will be made from the 2N® Contact Centre

to the Contact Centre routing object in . Set the routing rules to Solution 2N® NetStar

call the project.

Enable project selection - select this option to enable Agent/Manager to choose projects to log in. Unselect this option to allow Agent/Manager to view the projects assigned to it by Superadmin/Admin without changing them arbitrarily.


RouterAllocation strategy - select the allocation strategy to define the incoming project call distribution between Agents.

Linear - an incoming project call is routed to Agents in the sequence in which the users are assigned to the project.

Cyclic - an incoming call is routed to the Agent following the Agent that was the last to process a call.

All - an incoming call is routed to all the logged-in Agents at the same time.

Time to pick up call [s] - set the maximum ringing time for the selected Agent. When this timeout elapses, the call is returned to the project queue, or routed to another Agent.

Wait for active operator - if routed to a project to which no Agent is assigned, the incoming call will be processed as follows: The call is queued as defined in the Maximum queue waiting time if this option is selected and then routed to the Default

destination defined in . If this option is unselected and no Agent is logged 2N® NetStar

in to answer the call, the call is automatically routed to the Default destination

specified in .2N® NetStar

Route to next Agent - the call is routed to the next Agent in the sequence as defined in the selected Allocation strategy after the timeout defined in the Time to pick up parameter.

Interval between call tries [s] - define the interval in seconds during which an Agent is inactive and no call will be routed to this Agent until this timeout elapses.

Maximum queue waiting time [s] - the incoming call is queued until this timeout

elapses and then routed to the Default destination defined in .2N® NetStar

Inform about queue position - activate a service to inform the calling clients on their current queue positions. See below for the service parameters.

CallBackCallBack enabled - the CallBack function asks for activation. When the calling user requests CallBack activation (by playing a tone), enters the correct CallBack code (refer to DTMF CallBack code) and hangs up, the CallBack service will be enabled for this user. The calling user will be dialled by the according to the parameters defined below: First interval after CallBack after, Interval between CallBack and Number of CallBacks.


DTMF CallBack code - set the DTMF code to be entered for confirmation of the CallBack activation request. When the calling user enters the correct code and hangs up, the CallBack service will be enabled for this user.

Duration of DTMF detection - set a time in seconds after which the DTMF detector will be connected into the call to detect and recognise the DTMF characters sent. When this time period elapses, the DTMF detector will be deallocated and no DTMF characters will be recognised any more.

First interval of CallBack after - set the delay after a successful CallBack activation request after which the calling user shall be dialled automatically. Set the interval in minutes. The minimum value is one minute.

Interval between CallBacks - if the calling user did not answer the preceding CallBack, the next CallBack attempt will be made after this interval elapses. Set the interval in minutes.

Number of CallBacks - set the count of unsuccessful CallBack attempts. When this limit is exceeded, the CallBack will be deactivated and the calling user will have to call the Contact Centre.

TonesNormal - progress tone - the calling user hears this tone when the call is routed to an Agent and is ringing.

Infinite loop - select this parameter to define that tone playing is controlled by 2N®

and the application is not responsible for repeated tone playing.NetStar

Tone duration [s] - set the ring tone duration in seconds. Define this value precisely, otherwise the calling user hears silence or unfinished ring tone. After this period of time, the tone is replayed from the beginning. If you select , this parameter Infinite loopwill not be applied.

Queued - progress tone - the calling user hears this tone while queuing. It is mostly a supplementary tone without any information value, such as music on hold, for example.

Infinite loop - select this parameter to define that tone playing is controlled by 2N®

and the application is not responsible for repeated tone playing.NetStar

Tone duration [s] - set the ring tone duration in seconds. Define this value precisely, otherwise the calling user hears silence or unfinished ring tone. After this period of time, the tone is replayed from the beginning. If you select , this parameter Infinite loopwill not be applied.

CallBack tone - the calling user hears this tone when asked for the CallBack activation code.


Begin after [s] - set a time in seconds during which the calling user is queued before being asked about CallBack activation for the first time.

Tone duration [s] - set a value in seconds after which the calling user will be asked about CallBack activation again unless done so in the preceding request.

Information about queue position - progress tone - the calling user hears this tone while being informed of its current queue position.

Begin after [s] - define how long the calling user has to be queued before being informed of its current queue position for the first time. Set the parameter in seconds.

Interval [s] - set a time value after which the calling user shall be informed of its current queue position again.

Tone duration [s] - set duration of the tone informing the calling user of its current queue position. Define this value precisely, otherwise the calling user hears silence or unfinished ring tone.

Company OptionsThe following options are available:

Edit - edit a company.

Delete - delete a company.

Projects - display an overview of projects assigned to a company.

With Edit, the following tabs are available:

Company - modify the company data.

Users - add and edit users and their roles.


User States - set the user states. You can add 2N® Contact Centre Solution

states of your own to the pre-defined state categories if necessary and set the

required display colour. Having logged in, the user 2N® Contact Centre Solution

is in the Ready state and can receive calls. You can switch all the defined states via a web interface except for the Busy state, into which the user gets during ringing or active call only. Agents may modify their states only, while Managers and Admins are authorised to modify states of other users too.


Caution

Calls are only routed to the users that are in the Ready state.

VIP - create a list of VIP numbers to be preferred in the queue while calling the

. Adding the VIP numbers you can also set a priority 2N® Contact Centre Solution

(numeric value 1 and higher) to give selected VIP numbers even higher priorities. The priority 1 VIP numbers have the highest priority. You can create several separate lists within a company to be assigned to individual or all projects in the project settings.

Caution

VIP list can be imported using file, which must have the following .csvformat:

name,number,priorityJonathan Smith,+420251271276,1George Burton,+420251271288,1

SLA colours - set colours for each SLA level in percent for quick SLA evaluation in the project list on the Projects tab.

Remember to add users to a company by clicking the Edit button on the tab if Usersyou are logged in as Superadmin.

Now assign the users to a company and define their roles. If no users are displayed, refresh the list.


User roles:

Admin - is authorised to create and modify projects and create statistics (a company may be created by Superadmin only).

Manager - may log in even without a station and may create statistics.

Agent - processes incoming and missed calls.

Technician - is a consultant without the right to join a project. Used for statistic purposes only.

Project CreationCopy the project parameters from a superior company to create a project. The project parameters can be changed any time and project settings have a higher priority if the project and company parameters are different. Set the following project parameters:

Abbreviation - a five-letter/digit project abbreviation, used for incoming call display and user assignments to projects.

Project - any project name, which characterises the project/campaign/incoming queue purpose.

SLA - set a time value. Any incoming call that is answered after this time or missed does not satisfy the SLA condition. The SLA value is then lowered (in per cent) by the unsatisfactory call.

Project OptionsThe following options are available:

Edit - edit a project.

Delete - delete a project.

Click Edit and select the tab to add users to projects. First add the project users Usersto a superior company to display them in the project user list. Agents and Managers are only displayed. If a role is changed for a user, the change will be made within the company too.


On the tab, enable a project VIP number list defined within a company or define a VIPVIP number list of your own for your project.

Completion

View all the objects in the Routing - Routing Objects - Contact Centres in the 2N®

configuration tool. Assign the Default destination (for calls off the Contact NetStarCentre working time) and Connectivity loss destination to each project.

Having set the destinations, set the routing rules for incoming calls to be routed to the

selected projects and processed properly. Now you can start using the 2N® Contact

.Centre Solution

Caution

Manage (rename, delete) the Contact Centres only via the 2N® Contact

server web interface. Do not remove or rename the Centre Solution

Contact Centres (projects) using the configuration tool.2N® NetStar



LoginLog in with the user name and password assigned to you. Select the position, station (if you are assigned multiple stations) and project (if the project settings allow so).

Now click Start or save the selected station and project and log in to the 2N® Contact

. If you have stored the selections, select to skip Centre Solution Use stored sessionover the station and project selections in the next login.

Now select the place or position (for graphical depiction of the current user location only). If you have the Manager role, you can also log in without selecting a position/station. In that case, you will not be active for receiving calls and will not be displayed among the logged-in Agents. You can only view statistics, monitor the current states of logged-in users and so on.


If you have more stations than one, you can select the one to receive the 2N® Contact

calls.Centre Solution

Now select a project to log in if this option is available.

AgentThe upper tool bar provides the following options:

State

Projects

Agents

Calls

Filter by project

User settings

The start screen displays the tab including the current project statistics. Select a Stateproject in the menu to display the required statistic data.Filter by project


State

General

Operators OverviewOnline - count of all logged-in operators

Busy - count of speaking/ringing operators

Active - count of Active operators

Away - count of Away operators

Offline - count of Offline operators assigned to project(s) by filter

Calls TodayCalls - total count of calls (both received and missed)

Active - total count of active calls (answered by operators, ringing or queued)

Missed calls - count of missed calls

Average wait time - client's average waiting time (before call answer)

Average call time - average operator call time

ProjectsSLA - SLA value (average above all active projects or for a selected project) per working time or current day

Active - count of active projects


System

System Status

Status now - current status2N® Contact Centre Solution

Up time - time after the last run2N® Contact Centre Solution

NetStar

Status now - current connection status with 2N® NetStar

Status for - time after the last successful connection

System Loguser log in/out log


Projects

The overview displays the current state(s) of all projects or a selected project: number, whole name, abbreviation and some statistic data (SLA, total calls, missed calls, average wait time) within the project working time.

Agents

The tab includes a list of logged-in Agents. Select one of the three Agent display modes to the right (micro, compact, standard). Click on the arrow to the right to find Agent statistic details in the standard mode:

Number of Agent position

Agent name

Agent extension number

Total calls - total count of successful calls

Missed calls - count of missed calls

Average call time - average call duration

Average wait time - average Agent waiting time

Projects - projects to which the Agent is logged in

Change status - select one of the defined statuses. However, the operator can receive calls in the Ready state only (default state upon login).

Actual status - display the current operator status and status time. The calling user number/name is also displayed during an active call.


CallsOverview

The tab displays the queued calls waiting for an available Agent.Overview

The project name (from which the call came), up time and calling user number are included in the description.

Missed

The tab displays the missed calls.Missed

The project Id, up time and calling user number are included in the description.


Having displayed the details, you can call back or delete the missed call. If the list includes more missed calls from one and the same calling user, all records assigned to

the given number will be removed after answer or deletion, Make sure that the 2N®NetStar routing rules are defined properly to make successful CallBacks.

User Settings

Status change - status change to one of the defined statuses

Password change - user password change

Logout


ManagerIn addition to having all Agent properties, Manager can log in without a station, view Statistics in the tool bar and change current Agent statuses.


StatisticsThe tab displays detailed statistic data of the Contact Centre. Statistics on Statisticsprojects, Agents and time are available. Generated statistics can be saved in reports or downloaded as csv files.

Create Report

General Settings

Report name - complete the report name to save the report.

Report description - enter a detailed description for the statistics report to be saved.

Report for:

All projects - the generated statistics will include data of all projects in the database.

All projects and direct calls - in addition to project data, the generated statistics will include data on direct calls too, i.e. the calls that were routed directly instead of via the Contact Centre object.

Active projects only - the generated statistics will include data from active projects only (as set in the project configuration).

Ended projects - the generated statistics will include data of inactive projects.


Direct calls - direct call statistics.

Manually select projects

Date range

Last week

Today

Current week

Custom date range

Agent selection

Custom statistics

Agent statistics

Per project activity - count of all / received / missed calls that were routed to the selected Agent in the given project.

Overview - count of calls (total, missed, received, outgoing ...) plus average wait time and average call time for selected Agents.

Agent states - total Agent login time and status time.

Agent state activity - Agent state changes in time.

Activity per hour - total call count per Agent per hour in a day.

Activity per day - total call count per Agent per day in a month.

Activity in a week - total call count per Agent per day in a week.

Activity per month - total call count per Agent per month in a year.

Project statistics

Overview - count of all / missed / below SLA / above SLA calls, maximum and average call and wait time, etc.

Activity per - total count of calls to a project per hour. The SLA value houris included for the given hour in brackets.

Activity per week - total count of calls to a project per week. The SLA value is included for the given day in brackets.

Activity per day - total count of calls to a project per day in a month. The SLA value is included for the given day in brackets.

Call statistics

Load per hour - total count of all / received / missed / below SLA / above SLA calls to selected projects per hour in a day, maximum and average call and wait time, etc.

Load in a week - total count of calls to selected projects per day in a week.

Load per day - total count of calls to selected projects per day in a month.

Load per month - total count of calls to selected projects per month in a year.

Overview - listing of all calls routed to selected projects in the given period of time.

Reachability - statistic data on the Contact Centre reachability (time


Reachability - statistic data on the Contact Centre reachability (time necessary for incoming call answer, etc.).

Calling numbers - list of all calling numbers to selected projects.

Generate, Schedule and Save

Save report - save the generated statistics (report) for later use. View the saved reports on the tab. Click the asterisk to select the report to be Saved Reportsdisplayed and automatically updated on the tab. Complete the report Reportname and description to save the report.

Public - make the report accessible to other users with the Manager and Admin roles.

Schedule report - prepared for later implementation.

E-mail report - prepared for later implementation.

Choose action

Generate report

Download rough data (CSV)

Submit - generate a report or prepare a file for download as configured.

Saved Reports

The tab displays a list of all reports created.Saved Reports

Name

Statistics count

Report description

Download report

Send report to e-mail ( prepared for later implementation)

Report settings

Report details

Delete report

Click the asterisk to view details on the selected report on the tab.Report


Report

AdminIn addition to having all Manager properties, Admin may edit companies and projects and add/remove project users. Admin cannot log as an active user to the Contact Centre (cannot receive calls).


2N® NetStar Records Manager Manual

The allows for a simple management and rapid 2N NetStar Records Manager®filtration of recorded calls according to various categories, for example number (called and calling), date and time. Immediately after login, each user will see its call records and, depending on the allocated user rights, any other user call records (Administrator, e.g.). You can play your records via your Internet browser, make

comments using the web interface or download the 2N® NetStar Records Manager

records to your PC for later use.

After installation on the server, each user will be able to log in easily via its unique URL in the Internet browser. Make sure that there is a functional Internet connection to make the system work properly.

Content:

1. 2N® NetStar Records Manager Installation Wizard

2. 2N® NetStar Records Manager Configuration



Installation

Install the directly on a hardware PC or virtualisation 2N® NetStar Records Manager

platform (VMware, Oracle VM VirtualBox, etc.).

Caution

Make sure that the following minimum PC/virtualisation platform

configuration requirements are met while installing the 2N® NetStar

to avoid system malfunction!Records Manager

64bit processor

4 GB RAM

100 GB free hard disk space

Caution

The size of recordings made by is approximately 1 MB per 1 2N® NetStar

minute record. Therefore, check the capacity of your hard disk.

Note

Make sure before installing the onto a 2N® NetStar Records Manager

virtualisation platform that VT-x or AMD-V or any other similar virtualisation function is enabled according to the hardware PC processor type .

Example of Virtualisation Platform Installation

Connect the received ISO file with the installation to 2N® NetStar Records Manager

the CD/DVD drive in the virtualisation platform that meets the minimum configuration requirements mentioned above. Set booting from the CD/DVD drive and start the virtual PC. When the ISO file has been read, the following screen will be displayed:


Select the first option on this screen and the first option on the next screen (Use all space) and click .Next


When the installation is completed, the virtual PC gets restarted. If the booting sequence was changed in the virtual PC configuration, we recommend you to return Booting from hard disk to the first place.

After booting from hard disk is completed, you will see the login window:

Log in to complete the installation process: login and password .admin 2n

Upon login, the post-installation script will start automatically to complete the 2N®

installation.NetStar Records Manager

Caution

The post-installation script takes a few minutes. Be very careful while

entering data and do not interrupt or skip over the script to avoid the 2N

malfunction.® NetStar Records Manager

Language is the first script selection.

Having selected the language version, enter the server hostname.

Select the interface to be used for connection (this IP address will then be entered into the Internet browser if necessary).


If the DHCP server is available in the network, you can keep the assigned IP address or set the static IP address. If you select No, you will be invited to enter a new IP address, network mask, default gateway and primary and secondary DNS.

Now choose the type of user connection to the : IP 2N® NetStar Records Manager

address or Domain name. If you select the latter option, you will be asked to enter the domain name.

Caution

Make a corresponding record in your DNS server if the users are connected via their domain names.

Press any key to continue. Enter the user password to authorise the admin 2N®

server storage. Use these access data for storage mapping NetStar Records Manager

in . The password can differ from the server 2N® NetStar 2N® NetStar Records Manager

access password.

Note

Do not change the admin password for authorisation of the 2N® NetStar

server storage until the post-installation script is Records Managercompleted in the configuration menu (console) displayed upon every Admin login. To change the password, select 7 in the configuration menu and enter the following:


smbpasswd -a admin

Having entered the command, you will be invited to enter a new password.

Now create the folder for saving records.

The service will be enabled when all the required data have been set.

Press any key to complete installation. Now log in via your Internet browser and start

using the . The configuration menu below will be 2N® NetStar Records Manager

displayed upon Admin login.


Caution

Please set the correct time (via the NTP server - option 4, e.g.) before

using the as correct time setting is crucial 2N® NetStar Records Manager

for some functions.


2. 2N® NetStar Records Manager ConfigurationSection content:

First Login

Having completed installation, enter the server IP 2N® NetStar Records Manager

address or domain name into your Internet browser.

Note

Use Internet Explorer 8 and higher, Mozilla Firefox or Google Chrome for your web login.

For the first login, use the server Superadmin account: 2N® NetStar Records Manager

name and password . Click the logged-in user name in the right-hand superadmin 2nupper screen corner to change the Superadmin password after login if necessary.

The start screen is displayed after a successful login.


Click in the upper tool bar and then to upload the NetStars Upload licence 2N®

licence.NetStar Records Manager

The following data are displayed after licence upload:

SN - serial number2N® NetStar

Intercoms - count of licensed intercoms2N® Helios

NetStar servers - count of licensed platforms2N® NetStar

Date - last licence verification date

Expiration - licence time validity


Add 2N ® NetStar

Select in the upper tool bar and click to add .NetStars Add 2N® NetStar

NetStar - enter the connection name.

IP - enter the IP address. 2N® NetStar

Port - define the communication port (default port is ).2N® NetStar 6543

System user - set the user for authentication (we recommend you 2N® NetStar

to create a special user in for this purpose).2N® NetStar

System password - enter the system user password.

(confirm)System password - re-enter the system user password for confirmation.

Storage - select a folder from the list to the right to which recordings shall be saved.



Add Intercom

The server adds intercoms automatically based on the 2N® NetStar Records Manager

replies to the sent broadcast message. Thus, make sure that the intercoms are located in one and the same broadcast domain or that the broadcast message is resent by network elements.

Having added the intercoms, click Edit to complete the settings and activate video recording from the intercom(s).

Intercom - enter the intercom name.2N® Helios


NetStar - select the with which the intercom is 2N® NetStar 2N® Helios

registered.

Telephone number - enter the phone number.2N® Helios

Recording - activate / deactivate recording.

Caution

Make sure that the phone number matches the number under which the intercom is registered with to avoid video recording error.2N® NetStar

Use the field in the upper right-hand corner to search intercoms according to the Findfollowing parameters (Intercom, IP, NetStar, Telephone number):

Intercom - enter the name.2N® Helios

IP - enter the IP address.2N® Helios

NetStar - enter the IP address of the with which the 2N® NetStar 2N® Helios

intercom is registered.

Telephone number - enter the phone number. 2N® Helios

Availability - display information on availability.2N® Helios

Step

Intercom edit

Activate / Deactivate intercom

Delete intercom

Arrange the intercoms in any column. Click on the column name to display 2N® Helios

an arrow: arrow up means descending order and arrow down means ascending order.


Users

The tab provides a list of all users from all the s added. The users Users 2N® NetStar

are read automatically when the licence is added and can be updated 2N® NetStar

manually by clicking the Update icon. Each user is automatically assigned the lowest role (Agent) at the beginning. Superadmin or Admin can change the user roles. The Admin, Manager and Agent roles have the following rights:

Admin - has access to all records within the it is assigned to and 2N® NetStar

may set user roles.

Manager - has access to its records and records of the users assigned to its group.

Agent - views its records only.


Note

User login data and passwords are obtained from . Change 2N® NetStar

the data and password via the menu in the NsAdmin Users – User rights

configuration tool. A subsequent password change via the 2N® NetStar

web interface changes the password in Records Manager 2N® NetStar

too.

Storage Mapping in 2N ® NetStar

Storage mapping in via the NsAdmin configuration tool is the next step. 2N® NetStar

Add a physical storage in the Global data – Storage Manager menu displayed using the right mouse click (Add).

Caution

Make sure that the network storage name does not include diacritics and spaces.

Enter the storage name and confirm.

Set Network as Storage type.

Access port - set the server address in the 2N® NetStar Records Manager

\\ip_address\share format.

Free size (B) - define what space shall be used; enter -1 to use the whole disk capacity.

Login - enter admin.

Password - enter the password used for installation (refer to Section 1 of this manual).

Connect attempt in (s) - define when connection attempt shall be repeated after storage failure.

After storage mapping, a row will be added to the table. If all the Detected Storages parameters have been set correctly, the column will include the value.State Operating

Now right-click on , select and select the Voice Record Add physical storage mappingcurrently added storage.


Having added the storage, select the and Move evidence of files to storage Keep files options.removed from records

Enter the name of the record-saving directory created during installation into the Pathfield.

Finally, set . Refer to the menu for Record format .wav Global data - Global parameterssettings.



LoginLog in with the user name and password assigned to you.

AgentThe upper tool bar provides the following options:

Records

User Settings - change the user password if necessary.

Log out - log out.

The start screen displays the tab including the list of records. Records

Records


Use the function to search records according to Name, Calling, Called, Note or FindTime.

The searched string need not be precise, you can enter just a part of the string. Example: Enter '20' into the field to find all records including '20' in any column Find(Name, Calling, Called, Note, Time).

The following buttons are located next to the field:Find

Download - download records to a PC.

Edit - edit a note.

Delete

Click the buttons to operate the currently filtered records.

There four fields below the buttons ( ) for more Time from, Time to, Calling, Calleddetailed search and a button for search confirmation. Use this type of search for Filtersearch in defined columns and an absolute match.

The last part of the tab provides a seven-column list of all or filtered-out Recordsrecords:

Name - record name

A / V - audio or video

Calling - calling number

Called - called number

Note - note to record

Time - recording time

Step - buttons for operations with selected records

Arrange the records in any column. Click on the column name to display an arrow: arrow up means descending order and arrow down means ascending order.

The column includes four buttons for operations with selected records.Step


Click the first button to play a record, the second to edit a note, the third to delete a record and the fourth to save a record into a PC.

There are tool bars above and below the list including the following buttons:

Previous records - is applied after 10,000 records

First - first list page

Previous - page preceding the current list page

Number of pages from 1 to 200 - list page number

Next - page following the current list page

Last - last list page

Next records - is applied after 10,000 records

One list page displays . When page 200 is reached (10,000 records), the 50 records and buttons will become applicable and page Previous records Next records

numbering will be 1 to 200 again.

ManagerIn addition to having all Agent properties, the Manager can also view records of the users assigned to its group. Admin or Superadmin assigns users to groups.

AdminAdmin has all of the above mentioned Manager properties. Moreover, it can view all

records within one and use the option in the upper tool bar.2N® NetStar Users


Users

The tab includes all users of the selected .Users NetStar2N®Use the field in the right-hand upper corner to search users according to the FindUser, Role or NetStar (IP address/port) parameters.

The searched string need not be precise, you can enter a part of the string only. Example: Enter '20' into the field to find all records including '20' in any column Find(User, Role, NetStar).

Click the , button next to the field to update the records in accordance Refresh Find

with the selected (in case users are added/deleted/renamed in 2N® NetStar 2N®

).NetStar

The user list includes four columns:

User - user name


User - user name

Role - user role

NetStar - address and port2N NetStar®Step - click Edit to change a user role

Arrange the records in any column. Click on the column name to display an arrow: arrow up means descending order and arrow down means ascending order.


2N® NetStar User Manuals

Content:

1. Incoming and Outgoing Calls

1.1 Incoming Calls

1.2 Outgoing Calls

1.3 In-call Functions


2.1 Call Transfer

2.2 Call Parking

2.3 Ringing Take-Over

2.4 Conference Services

2.5 Call Forwarding

3. Secured Services

3.1 Station Activation

3.2 Login to Port

3.3 Bundle Login

3.4 PIN Setting

3.5 Private Calls


4.1 Time and Date Setting

4.2 Alarm Clock

4.3 Time Conditions

5. VoiceMail

5.1 VoiceMail Welcome Note

5.2 Forwarding to VoiceMail

6. Profiles

6.1 Profile Activation

6.2 Profile Presentation Text Setting

7. Other Services

7.1 Recording Own Messagess




9. Table of Service

10. 2N® StarPoint manual

10.1 Basic Information

10.1.1 Introduction

10.1.2 Getting Started

10.1.3 Telephone Description

10.1.4 Accessories

10.1.5 Terms and Symbols

10.2 Basic Operation

10.2.1 Outgoing calls

10.2.2 Incoming calls

10.2.3 During a Call

10.3 Telephone Menu and Functions

10.3.1 Main Menu

10.3.2 Call Register

10.3.3 Messages

10.3.4 Contacts

10.3.5 Profiles

10.3.6 Settings

10.4 Button Functions

10.4.1 Basic Terminal Buttons

10.4.2 Programmable Buttons - Setting

10.4.3 Programmable Buttons - Functions

10.5 2N® OptiSet Terminals

10.6 2N® OpenStage Terminals

10.7 Glossary

10.8 Important Warnings and Certifications


11.1 Představení

11.1.1 O telefonech 2N® OpenStage

11.1.2 Popis jednotlivých typů

11.1.3 Příslušenství

11.2 Základní ovládání

11.2.1 Odchozí volání

11.2.2 Příchozí volání

11.2.3 Během hovoru



11.3 Menu telefonu a jeho funkce

11.3.1 Hlavní menu

11.3.2 Výpis volání

11.3.3 Zprávy

11.3.4 Telefonní seznam

11.3.5 Profily

11.3.6 Nastavení telefonu

11.4 Funkce tlačítek

11.4.1 Základní tlačítka terminálu

11.4.2 Programovatelná tlačítka

11.4.3 Funkce programovatelných tlačítek

11.5 Důležitá upozornění


12.1 Základní informace

12.1.1 O telefonu

12.1.2 Popis telefonu

12.1.3 Příslušenství 2N® OpenStage

12.2 Základní ovládání 2N® OpenStage 40T

12.2.1 Odchozí volání

12.2.2 Příchozí volání


12.3 Menu telefonu a jeho funkce

12.3.1 Hlavní menu

12.3.2 Výpis volání

12.3.3 Zprávy

12.3.4 Telefonní seznam

12.3.5 Profily

12.3.6 Nastavení telefonu

12.4 Funkce tlačítek

12.4.1 Základní tlačítka terminálu

12.4.2 Programovatelná tlačítka

12.4.3 Funkce programovatelných tlačítek

12.5 Důležitá upozornění


13.1 Incoming Calls

13.2 Outgoing Calls

13.3 Call Transfer


13.3 Call Transfer


1. Incoming and Outgoing CallsThis section describes the main call establishing and processing principles provided by

the communication system.2N® NetStar


1.1 Incoming Calls

1.2 Outgoing Calls


1.1 Incoming CallsWhenever a call is coming from a public/GSM/IP/private network and/or from internal subscribers, your phone starts ringing.

Receiving Calls

To receive a call, just pick up the handset.

If your phone is equipped with a HandsFree button, push the button and start talking without lifting the handset. Before using the HandsFree function, make sure that your phone is equipped with a microphone and a loudspeaker.

If your phone is equipped with a headset, push the HandsFree button to receive the call. If the HandsFree button is not used for controlling the headset, follow the manufacturer's instructions.

To receive a VoIP call using your phone, follow the above listed steps. If you are using a software phone (e.g. a SJ-phone), then follow the particular software control instructions.

Rejecting CallsIf you do not want to speak to the caller, you can use the call rejecting service. The caller hears the busy tone. In case more stations than one are ringing (when the call is routed to the user and all the user stations are ringing), the caller hears the ringing tone until you reject the call for the last user station. You can reject a call using the

ISDN phones, phones and some software phones for VoIP 2N® StarPoint

communication, or on an external GSM station.

For an ISDN phone, use the ESC or another button for this purpose.


For a phone, use an Escape-programmed button. If you have not 2N® StarPoint

programmed a button for this function, push the Forward button once or the Back button twice and then press the confirmation button to reject the incoming call.

Most software phones used for VoIP communication have a Decline button, which is also used for rejecting calls.

Ignoring CallsYou can also ignore an incoming call, letting it ring until another user answers it (in the case of routing to multiple user stations), or the calling party or the communication system terminates it (after a 180s default timeout in the latter case).

Rejecting All Incoming Calls (DnD)

You can assign the Do Not Disturb function to a phone button to reject 2N® StarPoint

all incoming calls. Outgoing calls are not affected by this function. The LED next to the button indicates whether or not the function is enabled.

1.2 Outgoing CallsTo call a subscriber in a public/GSM/IP/private network, dial the subscriber number on your phone.

Call Set-Up Options

Lift the handset and dial the subscriber number.

Press the HandsFree button and dial the subscriber number.

Dial the required number and lift the handset or press the HandsFree button (using this option you can correct a wrongly dialled number).

If your phone is equipped with a headset, then all of the three above listed options are available to you. Most probably you dial the number and then press the HandsFree button to activate the headset. If the HandsFree button is not used for controlling the headset, follow the manufacturer's instructions.

If you are using a software phone (e.g. SJ-phone), then follow the relevant software control instructions. Mostly the number is entered from the numeric keypad or directly by pressing the software phone's virtual buttons with the mouse.

Calling Internal Subscriber

Dial the number of the internal (communication system) subscriber using one of the options described above.


For a phone, you can use the internal phone directory function 2N® StarPoint

and search the number according to the subscriber name.


Calling External Subscriber

Dial the number of the external subscriber using one of the options described above.

To access a public or private network it is mostly necessary to dial a network access code before dialling the subscriber number, typically '0'. To make a call over the VoIP dial '9' in most cases. Your service organisation will inform you which access types are available to you.

You can also use any of the options described below to call an internal/external subscriber (Speed Dialling, Hotline, Redial, Log Dialling, Direct Access). To use some of the options ask your service organisation for settings.

Speed Dialling

Speed dialling is a simplified, mostly single-button dialling of a station, 2N® NetStar

user or group. Speed dialling must be set by the administrator.

HotlineHotline is the destination that is automatically dialled after handset lifting or HandsFree activation and dialling timeout expiry. The destination can be another station, user, group, public network subscriber, or the ringing table.

RedialThe Redial service (the R button on most phones) enables you to redial the last called number regardless of whether or not the call has been set up successfully.

With the phones, the last called number is the last successfully routed 2N® StarPoint

one (the call was answered or the called station was alerted).

Note

Certain types of analogue and ISDN phones have their own database of missed and received calls. These calls are governed directly by the logic of these phones.


Log DiallingLog dialling means dialling the numbers recorded in the AOC database. The list of calls (log) includes three types of calls – Missed, Received and Outgoing. The service is available for:

StarPoint2N® phone users;

Other phone users;2N® NetStar Assistant

Software applications such as the CTI phone or CRM system.

Note

Certain types of analogue and ISDN phones have their own database of missed and received calls. These calls are governed directly by the logic of these phones.

Direct AccessDirect access means dialling an external network number by circumventing the LCR rules. This can be done by direct access to a bundle or a specific port. Direct access can, for instance, be used for calling private destinations or cross-connections. Most often, however, it is used for port access to a port for service purposes (line function check). This option must be set by the communication system administrator and, typically, it uses a double digit access code.

Fast IntercomDo you need to quickly convey information to your colleagues, assistants or supervisors? The Fast Intercom will call them, inform them of an incoming call and

answer the call itself. All you have to do is speak. The service is only available on 2N®

phones. To set the Intercom buttons use (see the StarPoint 2N® NetStar Assistant 2N®

manual).NetStar Assistant

Push and hold the Intercom button on your phone. The called 2N® StarPoint

station ( ) gives a short ring and is automatically answered. You 2N® StarPoint

can start talking. To terminate the call just release the Intercom button. You can have more Intercom buttons for various destinations.


If the called station is a phone other than , the call rings until the 2N® StarPoint

user answers it.

Calling Line Identification Restriction (CLIR)The purpose of the Calling Line Identification Restriction (CLIR) function is to hide the caller's Id. The service is active for a selected station until disabled. The enabled status can be indicated upon any handset pick-up replacing the dialtone.

Note

This function must be supported by your mobile provider or public telephone network operator. Otherwise, the call will be disconnected.

Enable CLIR

*43

Disable CLIR

#43

Example: Suppose you need to hide your identification. Pick up the handset and dial *43. The CLIR activation is announced by a voice message and the call is terminated. The CLIR function is now enabled for all the subsequent outgoing calls from this station.

Note

If you use the CLIR function frequently and are equipped with a 2N®

key system phone, assign this function to any of its buttons. StarPointYou will be informed of the active status by a shining LED.



Temporary Call HoldTo put a call on hold temporarily (while talking to a colleague, searching data on your PC, etc.), push the FLASH button on your phone and do not hang up. The call is held by the communication system, the caller hears music and you can arrange for what you need. To take the call off hold, push the FLASH button again (see the subsection below).

You can also use the HOLD button on your phone. In this case, the call hold is governed by the phone's logic and you usually cannot hang up. To return repress the HOLD button.

Multiple Call HoldDo you have one call on hold and need to put your current call on hold too to call another subscriber? Then put multiple calls on hold, switching the call slots using the FLASH button.

By default, three call slots are available to each station. Basically, it means that you can make up to three calls at the same time using one phone. However, the case is not a conference call, but two calls on hold and one active call.

Note

For phones, you can assign the call slots to the phone 2N® StarPoint

buttons using the application.2N® NetStar Assistant

Example: Suppose you are speaking to a user and want to speak to another. Push the FLASH button to put the active call on hold. You get into the second call slot and can dial another number. Dial the subscriber number, the subscriber answers the call and you can start talking. To return to the first call, push the FLASH button twice. By pressing the button once you put the second call on hold and proceed to the third call slot for further dialling. By pressing FLASH again you return to the first call on hold.

Note

There may be more call slots than one per station. Ask your administrator for settings.


Turning Off Microphone – MUTEThe MUTE function that is provided by most phones. Push the appropriate button (often labelled with a crossed microphone symbol or MUTE sign) to turn off the microphone and the opposite party cannot hear you whilst you can hear them.



This section describes the connection services of the communication 2N® NetStar

system.


2.1 Call Transfer

2.2 Call Parking

2.3 Ringing Take-Over

2.4 Conference Services

2.5 Call Forwarding

2.1 Call TransferAre you talking with a subscriber and do you want to transfer the call to another subscriber? Then you can do so in one of the following ways. In all the cases it does not matter who initiated the original call (the caller or the person called).

Call Transfer with Hang-UpPush FLASH during the call and dial the subscriber number. The other party hears music (or a predefined voice message) while waiting for connection.

If you hear the ringing tone after dialling, you can hang up. This transfers the call to the selected subscriber and the caller hears the ringing tone instead of music.

If you hear the busy tone after dialling, the call cannot be transferred as the selected subscriber is busy (or the communication system has rejected the call for other reasons). You can return to the caller by pressing FLASH on your phone and complete the call. Or, you can transfer the call to another subscriber, put the call on hold by pressing the HOLD button, or park it (Call Parking, chapter 3.2). Dial:

F nn

where F is FLASH and nn is the number of the subscriber you want to transfer the call to.

Note

Unless answered by the other subscriber within a timeout, the call returns to the transfer-initiating user and alerts his or her station.


Call Transfer without Hang-UpPush FLASH during the call and dial the subscriber number. The other party hears music (or a predefined voice message) while waiting for connection.

If you hear the ringing tone after dialling, wait until the called subscriber answers and then either hang up to transfer the call, or push the FLASH button on your phone to return to the original call (the second call is thus put on hold and the subscriber hears music or a voice message).

If you hear the busy tone after dialling, you cannot transfer the call as the selected subscriber is busy. Repress the FLASH button on your phone to return to the original subscriber and complete the call. Or, transfer the call to another subscriber, push the HOLD button to put the call on hold, or park the call (Call Parking, chapter 3.2). Dial:

F nn

where F is FLASH and nn is the number of the subscriber you want to transfer the call to.

Call Transfer to PSTNThe service is similar to the above mentioned two ones. The difference is that it is necessary to dial the public/private network access code before dialling the subscriber number. You are recommended to use the call transfer mode without hang-up for this transfer option so that you can keep control of the call until it is answered by the called user. Dial:

F x nn

where F is FLASH, x is a public/private network access code and nn is the number of the subscriber you want to transfer the call to.

Return to Transferred CallThis is an extended service of the Call Transfer without Hang-Up service. You can only return to the caller if no connection with the dialled subscriber has been set up and you have not hung up. Push the FLASH button (thus terminating the other subscriber's alerting) for return. When the transferee has already answered the call, the Call Swapping service is used.


Call SwappingTo swap multiple calls associated with your phone, use the FLASH button. Inactive subscribers always hear music.


Note

This function depends on the count of call slots per station. For details refer to Subs. 2.2 – Multiple Call Hold.

2.2 Call ParkingCall parking means setting aside a call for a rather long time for later unparking by any user. With a call parked, you can use your station without limitations for both incoming and outgoing calls.

Call Parking on Central Parking PlaceDo you want to park your current call on any of the 10 parking places for later unparking by you or any other user? Then dial the following:

F*42n

where F is FLASH and n is the number of the parking place where you want to park the call.

If the communication system ignores your dialling, you are delivered an error message informing you that either the parking place is full or you have entered a value outside the 0–9 range. Try to park your call on another parking place or any time later.

If you succeed in parking the call, the call is terminated on your side and you can use your phone without limitations. The caller is played music or a predefined voice message.

If the caller hangs up, the call is released from the parking place and cannot be recovered.

If you fail to unpark your call within a timeout (180s by default), your phone starts ringing to remind you of your parked call.

Note

Your communication system administrator may modify the parking time.


Call UnparkingTo unpark a call parked by you or other users, you have to know the number of its parking place. To unpark a call dial:

#42n

where n is the number of the parking place from which you want to unpark the call.

Example: Suppose you want to unpark a call from parking place 5, where a colleague of yours has parked it. Lift the handset and dial the service access code #42. You are asked to enter the number of the required parking place. Enter 5. Now you are automatically connected to the call waiting on parking place 5.

2.3 Ringing Take-OverDo you need to take over a call from a busy or absent colleague just by lifting your handset and dialling a service? Make your administrator set the Ringing Take-Over service.

Call Take-Over from Own GroupYou take over unanswered calls from the group to which your station belongs according to their arrival order. You always take over the station that was the first to ring (FIFO). Contact your administrator for your group specification and setting modifications as necessary.

*#

Example: Suppose you want to take over a call coming to a station that belongs to your ring group. Lift the handset and dial *#. You are automatically connected to the caller and the original station stops ringing. If more calls than one are alerting your ring group stations you always take over the first incoming call.

Call Take-Over from Selected StationIf you know the number of the ringing station, you can take over its call without bothering about which call was the first to ring. In this way you can even take over the calls that come to the stations that do not belong to your group.

**nn#


where n is the number of the station you want to take over the call from.

Example: Suppose you want to take over a call coming to station 101 without knowing whether or not the station belongs to your ring group. Lift the handset and dial **. The communication system invites you to enter the number of the station to be taken over. Enter 101 and press # for confirmation. You are automatically connected to the caller and station 101 stops ringing.

Call Take-Over from Selected GroupThis service helps you take over a call from a group irrespective of whether or not you are a member of the group. You always take over the first ringing call in this group. To take over a call you must know the number of the group concerned. Contact your administrator for the group numbers and setting modifications as necessary. Dial:

#*nn#

where n is the number of the group you want to take over the call from.

Example: Suppose you want to take a call coming to a station that belongs to group 17. Lift the handset and dial #*. The communication system invites you to enter the group number. Type 17 and press # for confirmation. You are automatically connected to the caller and the original station stops ringing. If more calls than one are coming to the station, you always take over the first one.

2.4 Conference ServicesDo you need to communicate with multiple colleagues or trading partners at the same

time? offers you a conference calling option. The advantage is that all 2N® NetStar

conference participants can hear one another as opposed to Call Swapping.

In addition to standard conference calls, supports conference rooms too.2N® NetStar

Note

You can establish a conference call with up to 32 subscribers as programmed! For details contact your administrator.


Joining in Call

Are you calling a colleague of yours via the communication system to 2N® NetStar

settle an urgent matter and he or she is busy now? In that case use the Joining in Call service to settle the matter in time. Using this service you can join in a current call or conference and become a full-fledged participant. You can talk to your colleague and/or his or her call partner. Dial:

*9#nn#

where nn is the number of the station on which you want to join in the conversation.

Example: Suppose you are dialling your colleague's number 123456 via 2N® NetStar

and get the busy tone. Hang up and after lifting the handset and getting the dial tone again enter the service access code *9#. The communication system invites you to enter the number of the station you want to join in. Type 123456 and press # for dialling end confirmation. If the dialling is correct, several short tones (or a voice message) are played to the participants and you can start communicating.

Conference CallDo you need to talk to multiple colleagues at the same time? Set up a conference call. To do this, dial the service access code and the selected conference participants. The selected stations will be called by the communication system and join in the conference as soon as they answer the call. The conference will continue until the last-but-one subscriber hangs up. Dial:

*1#S1...Sn#*

where S1 to Sn (n=2,3,..., 32) are the numbers of the stations you want to call together within a conference call.

Example: Suppose you want to set up a conference between stations 101, 102 and 103 from station 100. Dial the service access number and the following message will be played: "Enter the conference participants separated with an asterisk." Enter the selected station numbers one-by-one and separate them with an asterisk. After entering the last station, press the # symbol (i.e. 101*102*103#). Do not hang up. All the stations start ringing. Having answered the call, the subscribers become participants to the conference and you, as the initiator, can quit the conference or actively participate in it as a full-fledged member.


Note

All successful connections are billed to the station that initiated the conference.

If other stations join in the current conference using the Joining in Call service, the call is billed to the incoming stations.

Adding to ConferenceThe Add to Conference service is used for either connecting three subscribers into a conference or adding another subscriber to a currently active conference call. Dial:

*0#

Note

Conference calls are subject to licence.

Set Up ConferenceAre you talking with a colleague, having put another call on hold, and want to join the three calls together within a conference? No problem. Just put the other call on hold too and use the Add to Conference service.

Example: Suppose you are talking to a colleague and have put an earlier call on hold (the held subscriber hears the Music on Hold). To joint all the call together in a conference, push FLASH to put the two calls on hold (the subscribers hear the Music on Hold) and get the tone again. Dial *0#. Now all the three subscribers are participants of a conference call.

Add to ConferenceAre you a conference call participant and do you want to invite another colleague for advice? Just put the call on hold, call the selected colleague, put the selected colleague's call on hold too and use the Add to Conference service.

Example: Suppose you have set up a conference call and want colleague 1006 to take part in it too. Push FLASH to get the dialtone while the other participants can talk on. Dial your colleague's number, i.e. 1006. The colleague answers the call. Repress FLASH and put the colleague's call on hold. After you hear the dialtone again, dial *0#. Now all of you are taking part in the conference.

Conference Room


Conference RoomThe conference room can be used for variable purposes. It helps a duly authorised subscriber call a conference or subscribers join in the conference progressively. The conference room is active within a certain time interval and can be made accessible to selected users only. Dial:

*2#nn#

where nn is the conference room access code.

Note

Conference rooms are always subject to licence.

Call Together Conference Room SubscribersDo you need to call together selected subscribers to become participants in a conference call? Call the conference room and the subscribers will be called together automatically by the communication system.

Example: Suppose you are calling together selected subscribers for a conference call. Dial the Call Conference Room service, enter the conference room access code and press # for confirmation (e.g. *2#1234#). If you are assigned the Dial others right, the other selected stations and destinations are dialled subsequently.

Progressive Subscriber Join-InDo you want to set up a conference call with a rather high number of trade partners without requiring their presence at a given time? Send them the conference room access code and the conference call time to access the conference room progressively according to their time availability.

Example: Suppose you want to join in a conference call held in conference room 123. Lift the handset and dial *2#. When invited so, enter the access code 123 and press # for confirmation. Now you are in the conference room. If you are the first (or last) conference participant, you are played music. Whenever another subscriber joins in, you will be connected automatically.

Note

For details refer to the communication system configuration program manual.


The users can select basic conference room settings via the 2N® NetStar

application.Assistant

2.5 Call ForwardingDo you need to be available for incoming calls or want your colleague or assistant to answer your calls while you are away? Then use the Call Forwarding service.

After activation, calls are forwarded to the selected destination (station, user, group, etc.). Incoming calls will alert the selected phone instead of your phone. The calling subscriber will hear the standard ringing tone.

The call forwarding costs are split between station A and the station that has forwarded the call (station B). Station A pays for the call between stations A and B while station B pays for the call between stations B and C. If the call between B and C

is free (e.g. stations B and C are internal stations), call forwarding is free 2N® NetStar

of charge. enables you to set several types of forwarding services 2N® NetStar

differing in the first number of the service access code.

Services beginning with '1' relate to user forwarding.

Services beginning with '2' relate to station forwarding.

Services beginning with '3' relate to VoiceMail forwarding as described in detail in the VoiceMail section.

Unconditional User ForwardingBy enabling this service you forward all incoming calls unconditionally to the destination of your choice (Call Forwarding Unconditional, CFU). An incoming call will alert your assistant's phone, if you have chosen this option for example, instead of your phone. The calling subscriber will hear the standard ringing tone.

Enable CFU

*11nn#

where nn is the number you want to forward the call to.

Disable CFU

#11


Example: Suppose you want to forward all of your incoming calls to number 555. Lift the handset and enter the service access code *11. You will be asked to enter the number for immediate forwarding. Enter 555 and press # for confirmation. You will be informed that the immediate forwarding has been set.

User Forwarding on BusyBy activating the service you forward a call to a destination of your choice (station, user, group, etc.) whenever your station is busy (you are talking to someone else) or unavailable for whatever reason (station defect, etc.). The calling subscriber is not queued but forwarded immediately to the selected destination. The caller hears the standard ringing tone.

Enable CFEC

*12nn#

Disable CFEC

#12


Example: Suppose you want to forward all of your incoming calls that cannot be answered due to the user not being available to number 666. Lift the handset and enter the service access code *12. You will be asked to enter the number for call forwarding on busy. Enter 666 and press # for confirmation. You will be informed that the call forwarding has been set.

Note

CFEC means Call Forwarding on Error Cause. The call is forwarded not only due to a busy status but also in the event of error causes.

User Forwarding on No AnswerBy activating the service (Call Forwarding on No Answer, CFNA), you forward a call to a destination of your choice (station, user, group, etc.) when you fail to answer the call within a preset timeout (30s by default). Your phone will be ringing for the defined time interval and then will be forwarded. The calling subscriber will hear the standard ringing tone. The time period can be set by your administrator.

Enable CFNA


*13nn#


Disable CFNA

#13

Example: Suppose you want to forward all of your incoming calls that are not answered within the defined CFNA timeout to number 777. Lift the handset and enter the service access code *13. You will be asked to enter the number for CFNA. Enter 777 and press # for confirmation. You will be informed that the call forwarding has been set.

Unconditional Station ForwardingBy activating this service you forward all of your station's incoming calls unconditionally to a destination of your choice. An incoming call will alert your assistant's phone, if you have chosen this option for example, instead of your phone. The calling subscriber will hear the standard ringing tone.

Enable service

*21nn#


Disable service

#21

Example: Suppose you want to forward all of your incoming calls to number 555. Lift the handset and enter the service access code. You will be asked to enter the number for immediate forwarding. Enter 555 and press # for confirmation. You will be informed that the call forwarding has been set.


Station Forwarding on BusyBy activating the service you forward a call to a destination of your choice (station, user, group, etc.) when you fail to answer the call within a preset timeout (30s by default). Your phone will be ringing for the defined time and then will be forwarded. The calling subscriber will hear the standard ringing tone. The time period can be set by your administrator.

Enable service

*22nn#


Disable service

#22

Example: Suppose you want to forward all of your incoming calls that cannot be answered due to the user not being available to number 666. Lift the handset and enter the service access code. You will be asked to enter the number for Call Forwarding on Busy. Enter 666 and press # for confirmation. You will be informed that the call forwarding has been set.

Station Forwarding on No AnswerBy activating this service you forward a call to a destination of your choice (station, user, group, etc.) when you fail to answer the call within a pre-set timeout (30s by default). Your phone will be ringing for this time period and then will be forwarded. The caller will hear the standard calling tone. The time period can be set by your administrator.

Enable service

*23nn#


Disable service

#23


Example: Suppose you want to forward all of your incoming calls that you have failed to answer in the defined time period to number 777. Lift the handset and enter the service access code*23. You will be asked to enter the number for CFNA. Enter 777 and press # for confirmation. You will be informed that the call forwarding has been set.

User Forwarding to PSTN / PNBy activating this service you forward a call to a selected destination in the public (PSTN) or private (PN) network. The forwarding procedures are the same for the busy and no answer causes. The destination is a PSTN/PN number including the access code.

Enable services

CFU

*11xnn#

CFEC

*12xnn#

CFNA

*13xnn#

where x is the PSTN/PN access code and nn is the number you want to forward the call to.

Cancelling All User ForwardingThe service cancels all active forwarding settings for the user that is assigned to the service-calling station. It does not apply to the station forwarding settings.

#10

Cancelling All Station ForwardingThe service cancels all active forwarding settings for the service-calling station. It does not apply to the user forwarding settings.


#20

Cancelling All Station and User ForwardingThe service cancels all active forwarding settings for the user using the service-calling station and also all forwarding settings for this station.

##0

Note

The service does not cancel the VoiceMail forwarding settings. To cancel VoiceMail forwarding, dial #30 as described in the VoiceMail section.


3. Secured Services

This section describes the PIN-secured services of the communication 2N® NetStar

system.


3.1 Station Activation

3.2 Login to Port

3.3 Bundle Login

3.4 PIN Setting

3.5 Private Calls

3.1 Station ActivationThis service allows the user to deactivate one of the user stations to remove it out of routing. With this service on, the user is not allowed to make outgoing calls to a public

(PSTN) or private network (PN) via communication system. The station is 2N® NetStar

not alerted by an incoming call to the user. The subscriber calling to this station hears

the busy tone. To enable/disable the service use the application 2N® NetStar Assistant

(see the manual) or the administrator tool.

3.2 Login to PortThe service helps a user log in/log out a station to/from a selected port. The service can be useful in plants where the user's worksite is not stable. It can also be used in hotels where it helps operate a phone station in the guest's room.

Station Login to PortAfter implementing the service, the required station logs in to the port to which the service-requesting phone is physically connected. After login, the station owns the port (main station), which means that the settings of the logged-in station are transferred to the port. Multiple stations can be logged in to a port. The last logged-in station is used for outgoing identification. All of the logged-in stations are available for incoming calls. If a station has logged in somewhere else, it automatically logs out from the original port.

Login

*66nn#PIN


where nn is the number of the logged-out station and PIN is the user´s Personal Identification Number.

Example: Suppose you want to log in station 101 to a port. Lift the handset of the station that is physically connected to the port to which you wish to log in station 101. Dial the service access code *66. You will be asked to enter the number of the station to log in. Enter 101, press # for confirmation and, after being so requested, enter your PIN.

Station Login to Port as GuestAfter implementing the service, the required station logs in to the port to which the service-requesting phone is physically connected. After login, the station is a guest on the port to which it has logged in. Multiple stations can be logged in to a port. The last station logged-in as the main station (service *66) is used for outgoing identification. All of the logged-in stations are available for incoming calls. If a station has logged in somewhere else, it automatically logs out from the original port.

Login

*67nn#PIN


Example: Suppose you want to log in station 101 to a port. Lift the handset of the station that is physically connected to the port to which you wish to log in station 101. Dial the service access code *67. You will be asked to enter the number of the station to log in. Enter 101, press # for confirmation and, after being so requested, enter your PIN.

Station Logout from PortAfter implementing the service, the required station logs out from the port to which the service-requesting phone is physically connected. The station can be re-logged in any time to any port. This service logs out both the main and guest stations.

Logout

#66nn#PIN



Example: Suppose you want to log out station 101 from a port. Lift the handset of the station that is physically connected to the port from which you wish to log out station 101. Dial the service access code *66. You will be asked to enter the number of the station to log out. Enter 101, press # for confirmation and, after being so requested, enter your PIN.

3.3 Bundle Login

A bundle is one of the many routing objects of the communication 2N® NetStar

system. Basically, it is represented by two or more routing objects to which incoming calls are routed either linearly or cyclically. With the linear strategy, calling to the first item in the bundle is always carried out first. With the cyclical strategy, calls are routed to the object which comes directly after the object routed in the previous step. You can also set for a bundle whether it should respond to a busy routing object and a busy status. All bundle settings are carried out by your administrator. However, every user has the opportunity to log in to any bundle in the communication system (as a station or user) provided the user knows its number. Another way to log in to a bundle

is enabling the appropriate profile using the application.2N® NetStar Assistant

Station Login to BundleTo log in a station to a bundle dial the service access code from the station. Having been so invited by the communication system, enter the number of the bundle to which you want to log in the station. The service is PIN-secured by default. The logout procedure is similar.

Log In Station to Bundle

*64PINnn#

where PIN is the Personal Identification Number of the user station is assigned to and nn is the number of the bundle to which you want to log in your station.

Example: Suppose you want to log in station 1234 to bundle 88. Dial the service access code *64 and you will be asked to enter your PIN. Dial 1234 and you will be asked to enter the bundle number. Enter 88 and confirm #. The station will automatically log in to bundle 88.

Log Out Station from Bundle

#64PINnn#


where PIN is the Personal Identification Number of the user and nn is the number of the bundle to which you want to log out your station.

Example: Suppose you want to log out station 1234 to bundle 88. Dial the service access code #64 and you will be asked to enter your PIN. Dial 1234 and you will be asked to enter the bundle number. Enter 88 and confirm #. The station will automatically log out from bundle 88.

User Login to BundleTo log in a user to a bundle dial the service access code from the station that is assigned to the user. Having been so invited by the communication system, enter the number of the bundle to which you want to log in the station. The service is PIN-secured by default. The logout procedure is similar.


Log In User to Bundle

*65PINnn#

where PIN is the Personal Identification Number of the user and nn is the number of the bundle to which you want to log in user.

Example: Suppose you want to log in user 1234 to bundle 88. Dial the service access code *65 and you will be asked to enter your PIN. Dial 1234 and you will be asked to enter the bundle number. Enter 88 and confirm #. The user will automatically log in to bundle 88.

Log Out User from Bundle

#65PINnn#

where PIN is the Personal Identification Number of the user and nn is the number of the bundle from which you want to log out user.

Example: Suppose you want to log out station 1234 to bundle 88. Dial the service access code #65 and you will be asked to enter your PIN. Dial 1234 and you will be asked to enter the bundle number. Enter 88 and confirm #. The user will automatically log out from bundle 88.

3.4 PIN SettingThis service helps you set and change the PIN. The PIN can be set from any user

station and is identical for all of the user stations. It is used for some of the 2N®

services and has to be entered by the user if requested so by the system. If NetStarnot requested, the services can be used without the PIN. The default PIN for all users is 1111.

If the PIN setting procedure is successful, the communication system confirms the new PIN with a message. If not, you are notified of having entered an incorrect PIN on each step and invited to re-enter the PIN. The third erroneous attempt is followed by PIN setting termination. The PIN should consist of four digits. If shorter, push an asterisk for termination.

Change PIN

*70 old new new

Delete PIN


*70 old*


Set New PIN after Deletion

*70 new new

Example: Suppose you want to change your old PIN 1111 to a new one 9876. After lifting the handset and hearing the dialtone, enter the service access code *70. You will be asked to enter the old PIN, so enter 1111. The communication system will then ask you for the new PIN, so enter 9876. The communication system will now ask you to repeat the new PIN, so re-enter 9876. If the PIN setting procedure is successful, the communication system will confirm the new PIN setting. From now on you should only use the new PIN for secured services.

Note

If you delete your PIN as described above and fail to enter a new one, you will not be able to use any of the PIN-secured services!!!

3.5 Private CallsThe service enables you to label selected outgoing calls as private. Such calls are then easier to identify in the invoicing software. The service allows all users to make private calls including those users that do not have a stable station. This is primarily appreciated by the users who have a limited number of terminals in the company. The service is automatically active for one call (until hang-up).

Private Call from User StationThis is a service for a user with an assigned station. The service requests the PIN by default but the PIN request can be cancelled by the administrator. After dialling the service and entering the PIN, you will receive a continuous dialtone from the communication system and you can dial the private number to any network using the standard routing rules of your communication system (e.g. start with '0' to access the PSTN, e.g.). Dial:

*71PIN

where PIN is the user´s Personal Identification Number.

Example: Suppose you want to call your wife at home number 555 666 777 but you do not want your company to pay for the call. After lifting the handset and getting the dialtone, enter the access code *71. You will be requested to enter your PIN. After


doing so, you will get a continuous dialtone from your communication system. Now you can dial the number. Since it is a call to a private network, you will most likely be requested to dial the access code '0' (or any other code) first and only then 555 666 777.

Private Call from Other than User StationThis is a service for a user without an assigned station or a user who needs to make a call from a colleague's station. By default, the service requests the station number and the PIN. This can be cancelled by the administrator upon request. After dialling the service and station numbers and entering the PIN, you will receive a continuous dialtone from the communication system and can dial the private number to any network using the standard routing rules of your communication system (e.g. start with '0' to access the PSTN, e.g.). The station you are calling will identify your station as the one from which you dialled the service. Dial:

*72nn#PIN

where nn is the number of the station to be billed and PIN is the user´s Personal Identification Number.

Example: Suppose you want to phone your wife at home number 555 666 777 from your colleague´s phone but you do not want your colleague to pay for the call. After lifting the handset and getting the dialtone enter the access code *72. You will be requested to enter the number of the station to which the call will be billed. Enter the number of your station, 101 e.g., and press # for confirmation. You will then be asked to enter the PIN of the user to be billed (i.e. the PIN of the user to which station 101 is assigned). After doing so, you will get a continuous dialtone from your communication system. Now you can dial the number. Since it is a call to a private network, you will most likely be requested to dial the access code '0' (or any other code) first and only then 555 666 777.



This section describes the time related services of the communication 2N® NetStar

system.



4.2 Alarm Clock

4.3 Time Conditions


To set date and time, use either the administration console or the 2N® NetStar 2N®

application. In either case, you have to know the login name and NetStar Assistantpassword of the user with administrator rights. Another possibility is to use the dedicated services.

Set TIMEThis service helps you set the communication system time simply by using your phone. Dial:

*51 hhmm

where hh represents hours and mm minutes. Single-digit parameters are entered with a zero.

Example: Suppose you want to change the communication system time to 8:15. After lifting the handset and hearing the dialtone, enter the service access code *51. You will be asked to enter the new time value. Enter 0815 and, after receiving the last digit and playing a message on the time change, the communication system disconnects the call. The time has been successfully set.

Note

The service will not work in case time synchronisation from the NTP server is enabled in your communication system.


Set DATEThis service helps you set the communication system date simply by using your phone. Dial:

*52 ddmmrrrr

where dd represents days, mm months and yyyy years. Single-digit parameters are entered with a zero.

Example: Suppose you want to change the communication system date to June 5, 2007. After lifting the handset and hearing the dialtone, enter the service access code *52. You will be asked to enter the new time value. Enter 05062007 and, after receiving the last digit and playing a message on the date change, the communication system disconnects the call. The date has been successfully set.

4.2 Alarm ClockThere are two ways how to set the alarm clock for your phone. The default alarm ringing time is 180s. If you are talking, the alarm clock starts ringing 180s after the call end. The alarm is stopped by lifting the handset or after 4 cycles have been completed.

One-Time Alarm ClockUse this service to set the unrepeated alarm clock mode for a certain time and date values. If you enter time only, the alarm clock setting is valid for the day on which the service is activated.

Enable alarm clock with date

*50 hhmm*DDMM#

Enable alarm clock without date

*50 hhmm#

Disable alarm clock

#50


where hh means hours, mm minutes, DD days and MM months. Single-digit parameters are entered with a zero.

Example: Suppose you want to set the alarm clock to May 9 at 6.30 am. After lifting the handset and hearing the dialtone, enter the service access code *50. You will be asked to enter the alarm time. Enter 0630. Now you will be asked to enter the date. Enter 0905. Now the communication system announces that the alarm has been set successfully and terminates the call.


4.3 Time ConditionsThis service helps you control calls variably in a given time interval. For instance, you can route incoming calls differently in the day and night modes and process outgoing

calls variably during the day depending on the provider's optimum tariffs. 2N® NetStar

provides an unlimited number of time conditions to be set. The time conditions are set and allocated by your administrator. Ask your service organisation for the settings.

Note

Users can set time conditions of their own using the 2N® NetStar

application and use them subsequently for use profile Assistantswitching.


5. VoiceMail

This section provides description of the communication system VoiceMail 2N® NetStar

services.

Do you need to work undisturbed yet ready to receive any call from your customers or business partners? Then use the VoiceMail service. Just dial the service access code from your phone to activate one of the call forwarding options for incoming calls and adapt the pre-recorded message to your specific situation. An administrator-defined count of VoiceMail records are kept in the communication system for a defined period of time. The records can also be sent to an email address of your choice in the wav format.



5.2 Forwarding to VoiceMail


The following services are used for the communication system VoiceMail 2N® NetStar

welcome note administration.

Recording Welcome NoteTo record the message to be played to welcome callers after service activation, enter the following string:

*35

Example: Lift the user station handset and dial the service access code *35. You will be invited to record a welcome note. Record the message and press *. The call will be terminated. The new message has been recorded and saved successfully. From now on this new welcome note will be used for any new call coming to VoiceMail.

Playing Welcome NoteCan't you remember your VoiceMail welcome note? Don't worry. Enter the following code to check your welcome note and decide whether or not to change it.

*36


Example: Lift the user station handset and dial the service access code *36. The current welcome note version will be played to you.

Deleting Welcome NoteTo remove your welcome note and use the communication system default message, enter the following string:

#35

Example: Lift the user station handset and dial the service access code #35. The current welcome note will be deleted and the communication system default message will be played to you next time you use VoiceMail.

5.2 Forwarding to VoiceMailThese forwarding options are governed by the rules described in the Call Forwarding section. The difference for VoiceMail is that the user's VoiceMail box is always the destination.

Unconditional Forwarding to VoiceMailThe first option is Call Forwarding Unconditional (CFU), which means that all incoming calls are forwarded to VoiceMail.

Enable CFU

*31

Disable CFU

#31

Forwarding to VoiceMail on BusyCall Forwarding on Error Cause (CFEC) is another option, meaning that calls are forwarded if your station is busy and/or unavailable for whatever reason (station failure, etc.).

Enable CFEC

*32


Disable CFEC

#32

Forwarding to VoiceMail on No AnswerThe last type is Call Forwarding on No Answer (CFNA). You can set the time between

the start of ringing and the moment the call is forwarded to VoiceMail using your 2N®

.NetStar Assistant

Enable CFNA

*33

Disable CFNA

#33


Cancelling All Forwarding to VoiceMailIf you are not sure which forwarding option you have set, cancel all active VoiceMail forwarding settings at once and set the one you want.

#30


6. Profiles

This section describes the use of profiles in the communication system.2N® NetStar

Do you hate complicated settings for such user parameters as forwarding, ringing volume or display backlight for variable periods of a day? Then the user profiles are exactly what you have been looking for! Once you set them, you can switch and modify them as you want until you are completely satisfied. Moreover, the communication system can also log in your station to various bundles depending on

the active profile. To create a profile use the phone menu or the 2N® StarPoint 2N®

application. Profiles can be combined with time conditions thus NetStar Assistantproviding the user with an even higher comfort.


6.1 Profile Activation

6.2 Profile Presentation Text Setting

6.1 Profile ActivationTo activate a profile dial the service access code and enter the profile number. The service can also be secured with a PIN.

Activate Profile

*60n

where nn is the number of the profile to be activated for the given user.

Example: Suppose you want to activate profile 3. Dial the access code *60 from the station that is assigned to this user. You will be asked to enter the number of the profile to be activated. Enter 3. If the PIN request is disabled, the call ends and the selected profile is active.

Deactivate Profile

#60n

where nn is the number of the profile to be deactivated for the given user.


Example: Suppose you want to deactivate profile 11. Dial the access code #60 from the station that is assigned to this user. You will be asked to enter the number of the profile you want to deactivate. Enter 11. If the PIN request is disabled, the call ends and the selected profile is inactive.

6.2 Profile Presentation Text SettingA presentation text can be set for each profile to be displayed to the calling

subscribers (using the key system phones). The text can be set using 2N® StarPoint

the SMS only (not by a call). Basically, two types of presentation text settings are available depending on where the relevant SMS is sent from. Always set the presentation text for the currently active profile (or a no-profile status).

*61

Setting from Internal Station

You can send the new SMS directly to the service access number from an internal station.

Example: Suppose you want to set the "I'm out for lunch" presentation for the Lunch profile (number 5). Dial *605# to activate the profile. Write an SMS ("I'm out for lunch") and send it to *61. Whenever a call comes within the active Lunch profile, the

calling subscriber's phone will display the "I'm out for lunch" text.2N® StarPoint

Note

Similarly, you can send the SMS using the or tray 2N® NetStar Assistant

icon applications. However, these applications provide easier profile activation and presentation text setting procedures.

Setting from External Station

You cannot send the SMS directly to the service access number from an external station because you have to send it to the communication system first. For this purpose, your administrator will help you set the text router for routing SMS. The SMS will have a more complex form and will be sent to the number of the SIM card installed in the communication system.

Example: Suppose you want to set the "I'm out for lunch" presentation for the Lunch profile (number 5) but you are out of your office. First send an SMS in the *605# format to the appropriate communication system SIM card number (774 406 458, e.


g.) from the external station to activate the Lunch profile. Now send another SMS "*61_I'm out for lunch" to the same SIM card number from the same external station

Whenever a call comes within the active Lunch Profile, the calling subscriber's 2N®

phone will display the "I'm out for lunch" text.StarPoint


7. Other Services

This section describes all the remaining services of the communication 2N® NetStar

system.




The communication system also provides a complete set of user voice 2N® NetStar

messages in variable language mutations. You can add them to the communication system as files or record them using the Record Own Message service.

Record Own MessageUse this service to record voice messages of your own into the communication system. Thus, you can easily update the IVR. To record a message, you have to know the progress tone identification code as communicated by the administrator.

*56

Example: Suppose you need to change the voice message for progress tone 123. Lift the handset and dial *56. You will be asked to enter the progress tone Id. Enter 123 and press # for confirmation. You will be invited to record a new message for the progress tone. Start speaking after a sound and press # to complete recording. The message will be played back to you. Press * to replay the message and # to terminate the call.

Replay Active MessageUse this service to play the currently active message for a selected progress tone. You have to know the progress tone Id within the communication system again.

#56

Example: Suppose you need to know the current message for progress tone 123. Lift the handset and dial #56. You will be asked to enter the identification code. Enter 123 and press # for confirmation. The communication system will start playing the currently active progress tone message in a loop. Hang up to terminate the call.


Restore Default MessageUse this service to restore the default message for the basic communication system progress tones.

*57

Example: Suppose you need to reset the default message for progress tone 123. Lift the handset and dial *57. You will be asked to enter the identification code. Enter 123 and press # for confirmation. The user message will be deleted and the default message will be reset for the progress tone.



This section describes how to operate the services using SMS messages.2N® NetStar

Apart from the methods described above, some of the services can also 2N® NetStar

be operated using text messages sent from internal or mobile stations. It is necessary to send the SMS in a specific form as requested by your COMMUNICATION SYSTEM administrator. In general, remember that the service access codes can, in the event of SMS setting, be sent in the format specified by the administrator while the remaining part of the text message, which bears the information, must be entered in the format defined in the previous sections. The following examples are given to clarify this. The services that can be operated in this manner are designated with 'S' in the last column of Table 11 (see the explanatory note under the table).

Example: Suppose you want to set time to 8:15. The SMS access to the service is based on the key text 'Time'. Your time setting SMS will thus be Time0815 and should be sent to the number specified by the administrator.

Example: Suppose you want to forward all calls to the user of station 101. The SMS access to the service is based on the key text 'CFU user'. Your forwarding setting SMS will thus be CFU user 101 and should be sent to the number specified by the administrator.

Example: Suppose you want to activate alerting of a station by calling the user. The SMS access to the service is based on the key text '*55'. Your ringing setting SMS will thus have the form *55 and should be sent to the number specified by the administrator from the station to be set.


9. Table of Service

This section provides an overview table of all the services.2N® NetStar



For all the above listed services a tone dialling phone has to be used.

Explanatory notes:

No symbol – Dialling during dialtone

C – Dialling during call

S – Can be set using an SMS.


10. 2N® StarPoint manualContent:

10.1 Basic Information

10.1.1 Introduction



10.1.4 Accessories


10.2 Basic Operation




10.3 Telephone Menu and Functions

10.3.1 Main Menu


10.3.3 Messages

10.3.4 Contacts

10.3.5 Profiles

10.3.6 Settings

10.4 Button Functions






10.7 Glossary



10.1 Basic InformationHere is what you can find in this section:

10.1.1 Introduction



10.1.4 Accessories


10.1.1 Introduction

The telephone sets are digital key system phones designed for the 2N StarPoint 500® system. All of them (except for the phones) are 2N® NetStar 2N® StarPoint 500 Entry

equipped with a two-row display, programmable buttons, the HandsFree function, and other options and accessories. The telephones are available in the arctic or anthracite colour versions.

StarPoint 500 Types:2N®

Figure: , , 2N StarPoint® 500 Entry 2N StarPoint® 500 Economy/Standard 2N® StarPoint 500 Advance

A telephone for common traffic – – is a simple digital 2N® StarPoint 500 Entry

telephone without a display, with 8 programmable buttons and one-way Speakerphone (it is not equipped with a microphone for HandsFree but uses the handset microphone).

A mid-range office telephone – – is the basic 2N® StarPoint 500 Economy, Standard

type of a digital key system phone with a two-row display and 12 programmable buttons. The model offers a one-way Speakerphone (is not equipped with a Economy


1.

2.

3.

HandsFree microphone); the model has a two-way Speakerphone (is Standardequipped with a HandsFree loudspeaker and microphone). The model is not Economyequipped with interfaces for optional modules (e.g. button modules).

A managerial telephone – – is the most advanced of all 2N® StarPoint 500 Advance 2N

telephones. The phone is equipped with a backlit two-row display, 19 ® StarPoint 500

programmable buttons and an interface for additional modules. A head set can be connected too. The apparatus provides a two-way Speakerphone (a loudspeaker and microphone are available for the HandsFree function).

Phone Package Contents:

a telephone set

a telephone socket cable

User Manual and Warranty Certificate

button tagname templates


Putting in Operation

Unpack the telephone set.

Connect the handset to the telephone set using a twisted cable (from the bottom – see the figure below).

Connect the telephone set to the telephone socket using the cable included in the delivery.

Warning

Be sure to connect the phone to a dedicated socket 2N® StarPoint 500

only.


Button Tagnames

Use the to create button tagnames automatically. Click on 2N® NetStar Assistant

on the button displaying screen to create a complete Prepare for printingtemplate in the PDF format automatically.

You can also use a blank template as included in the delivery to create button tagnames.

Tip

Refer to for details on button 10.4.2 Programmable Buttons - Settingprogramming.



Use the and buttons to browse through the phone menus and list items. Push

the confirmation button to select a function. Use the and buttons to adjust the ringing and call volume (handset or HandsFree). Any volume change applies to the current call only, the next call will have a volume level as set in the

phone menu (or ).2N® – NetStar Assistant

Telephone Display

All the telephones (except for Entry ones) are equipped with a two-2N® StarPoint 500

row display. The display contrast is adjustable in eight steps; time is displayed in 12h or 24h formats. Refer to for display parameter details.10.3.6 Settings

Display default:

Station number – this item depends on your PBX setting. The station number, name, or both can be displayed.

Current profile – this item depends on your PBX setting.

To work more comfortably with your telephone, adjust the display tilt in 2N® StarPoint

six steps:


Programmable Buttons

The telephones are equipped with programmable buttons. Use the 2N® StarPoint 500

to program the buttons. The system administrator defines the 2N® NetStar Assistant

basic settings, the user may adjust the button settings as required. Refer to 10.4.2 for more details.Programmable Buttons - Setting

Tip

Refer to for available 10.4.3 Programmable Buttons - Functionsprogrammable button functions.

10.1.4 Accessories

Optional acccessories can be connected to the phone bottom for 2N® StarPoint 500

convenience: additional button modules or head sets.


Button Modules

A 16/90-button console can be added to the Standard and 2N® StarPoint 500

Advance phones to increase the count of available buttons. Connect multiple consoles serially to get an extremely high number of buttons. Program the console buttons like

your telephone buttons. 18-button module is only available for the 2N® OpenStage 15T

and terminals.40T

Tip

Every 90-button module needs a power supply of its own. Every other 16-button module needs a power supply of its own.

Figure: Console, 16 buttons – 2N® StarPoint

Figure: Console, 90 buttons – StarPoint2N®


Figure: Console, 18 buttons – 2N® OpenStage


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Button Console Connection

Connect the console mechanically to the phone.

Interconnect the telephone and the console using a cable.

Connect the included mains adapter to the dedicated bottom socket for power supply.




Safety


Warning


Caution


Tip


Note




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10.2 Basic OperationHere is what you can find in this section:





Outgoing Call Setup

Pick up the handset. You can hear the dialtone.

Dial a number. Having dialled the last digit, wait for automatic call setup.

If stored in your or group directory, the called subsriber's name is also displayed.

The call time will be displayed when the called subscriber answers.

If the called user is busy, you will get the busy tone and the message Call endgets displayed.

Tip

Or, you can enter the called number on your phone diplay first and then pick up the handset to set up the call.

Push the HandsFree button to use the HandsFree function instead of lifting the handset. The HandsFree button indicator signals activation. Push the HandsFree button again to terminate the current call.

Tip

If the HandsFree button is not programmed on your phone, just press

the confirmation button after dialling a number to activate HandsFree .

Outgoing Call to Call Register RecordThe telephone stores records on missed, incoming and outgoing calls, the so-called call register. Using these records, you can call the subscribers whom you called or who called you before. The call register can contain up to one thousand records.


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Response to Missed CallSituation:

The phone is signalling missed calls. You want to reply to these calls.

Procedure:

Push to enter the list of missed calls.

Or, use the pre-programmed missed call accessing button if available.

If the missed call is not displayed, push to enter the main menu, select Call and then .register Missed calls

Now you can see the list of missed calls on your display. Use and to browse through the list items.

Having selected the item to be called (using the confirmation button), pick up the handset or push the HandsFree button.

Call to Previously Called NumberSituation:

You want to make a call to a number you called before.

Procedure:

Use the outgoing call history for calling a previously called number.

Push the outgoing call accessing button if pre-programmed to enter the list. If

the button is not programmed, push to enter the phone main menu, select and then .Call register Dialled numbers

Use and to browse through the list items.

Pick up the handset, or press the HandsFree button to dial the selected number.


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Call to Previously Calling NumberSituation:

You want to make a call to a number from which you were successfully called before.

Procedure:

Push the answered call accessing button if pre-programmed to enter the list. If

the button is not programmed, push to enter the phone main menu, select and then .Call register Received calls

Use and to browse through the list items.

Having selected the item to be called (using the confirmation button), pick up the handset or push the HandsFree button.

Outgoing Call from Telephone DirectoryTo facilitate dialling, program a telephone directory access button. Use the directory to search items according to their names or name parts. Pick up the handset, press

for confirmation, or push the HandsFree button to dial the telephone number for the selected list item. Refer to for more details on phone 3.3 Telephone Directorydirectory search.

Speed DiallingYou can program the so-called speed (quick) dialling buttons for frequently used numbers. Just push the button once to dial the selected telephone number. Refer to

for one-button dialling settings.10.4.2 Programmable Buttons - Setting

Calling Line Identification RestrictionUse the (Calling Line Identification Restriction) button to restrict your CLIRidentification. Refer to for CLIR button 10.4.2 Programmable Buttons - Settingprogramming.

Remember to press this button before dialling a number. The active CLIR mode is signalled by the button LED indicator. CLIR remains active until you re-press this button. If you activate CLIR during dialling, it will be active for the next call.

IntercomIntercom is a special type of call where a predefined telephone number is dialled by pressing the button and the called station receives the call automatically in Intercomthe HandsFree mode. The function provides flexible communication – you can speak almost the moment you press the button.


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Tip

The function works with the telephones only. If a 2N® StarPoint 500different telephone type is used, the call will not be answered automatically.

How to use Intercom:

Press and keep pressed the Intercom pre-programmed button.

Start speaking when you hear a beep.

The called subscriber answers the call automatically switching into the HandsFree mode. The called subscriber is notified of the call with an acoustic signal.

Keep the button pressed during the whole call.

Release the button to terminate the call.

Refer to for Intercom button programming.10.4.2 Programmable Buttons - Setting


Answering Incoming Call

An incoming call is signalled by displaying the calling station name and/or telephone number or the called number if possible. Names from your private phone directory or the system administrator directory are assigned to the numbers.

Pick up the handset to answer the call. Press the confirmation or HandsFree button to answer the call in the HandsFree mode.

Ringing Volume ControlSuppose you do not want to answer or reject a call yet do not want to be disturbed by your telephone's loud ringing. In that case, either lower the volume or mute the call as follows.

a) Push to adjust the ringing volume.


b) Push the or arrows, select and press to mute ringing of an Ringer offincoming call.

Rejecting Incoming Call

Use the or arrows, select and press for confirmation. The Reject callcalling subscriber will get the busy tone.

Or, you can use the programmable button. Refer to Esc 10.4.2 Programmable Buttons for Esc button programming.- Setting

The diagram below shows a complete incoming call menu.

Do Not Disturb (DND)You can use the ( ) function to work without being disturbed by Do Not Disturb DNDincoming calls. If this function is enabled, all incoming calls are rejected. Every rejection is followed by a missed call record on your display, allowing you to call the number later. Outgoing calls are set up without limitations.

Pre-program a button for the mode. The active state is signalled by a Do Not Disturbshining LED indicator at the button. Press the pre-programmed button again to cancel the mode. Refer to for DND button 10.4.2 Programmable Buttons - Settingprogramming.


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Switching to HandsFree During a CallSuppose you are speaking with your handset picked-up, but suddenly you have to put the handset aside without ceasing to speak. Push the button. The HandsFreetelephone transits into the HandsFree mode as signalled by the button LED. Now you can hang up and go on speaking. Push the button with your handset hung-HandsFreeup to terminate the call.

Tip

The function only works with the HandsFree or Speakerphone supporting terminals.

Handset/HandsFree Volume Control

Use the and buttons to adjust the volume of the handset or HandsFree mode during a call. The change will apply to the currently active mode (handset/HandsFree).

Microphone MutingDo you need to consult a colleague of yours or just not to be heard during a call? Use the function. Microphone muting is signalled by the button LED flashing. Re-Mutepress the button to switch the microphone on again.

Call TransferSuppose you want to transfer your opponent to another telephone number during a call.

Call Transfer with Hang-Up Procedure:

Push the or programmable button.Flash Call slot

You can hear the dialtone in your handset (the opponent hears music).

Dial the telephone number to which you want to transfer the call.

Hang up after receiving the ringing tone. The two subscribers will get connected.


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Call Transfer without Hang-Up Procedure:

Push the or programmable button.Flash Call slot

You can hear the dialtone in your handset (the opponent hears music).

Dial the telephone number to which you want to transfer the call.

Wait until the called subscriber answers the call.

Now you can push or the respective button to switch between the Flash Call slotactive and held calls.

When you terminate the active call, the held subscribers will be connected.

Call Transfer with Speed Dialling Procedure:

Push the programmable button during a call to which the State/Speed diallingsubscriber number that should receive the transferred call is assigned.

The active call is put on hold and a new call is set up to the required number.

Having received the ringing tone, hang up (Call transfer with hang-up), or wait for answer and then hang up to complete the transfer (Call transfer without hang-up).

10.3 Telephone Menu and FunctionsHere is what you can find in this section:

10.3.1 Main Menu


10.3.3 Messages

10.3.4 Contacts

10.3.5 Profiles

10.3.6 Settings

10.3.1 Main Menu

Press the confirmation key when the telephone is at relax to enter the main menu.

Use the and arow keys to browse through the menu items. Press to enter the submenus. You can program an button to accelerate your return to ESChigher menu levels.


Tip

If you press at relax, you will get directly to the List of called .numbers

Tip

If you press at relax, you will get directly to the menu Contacts(Telephone directory).


Menu Structure

The key system phone main menu includes lists of missed, incoming 2N® StarPoint

and outgoing calls. Each list contains up to 1,000 records on latest calls. You can access the call lists both from the main menu and by pressing a programmable button.

The lists are also accessible in the user interface.2N® NetStar Assistant


Missed CallsIt is useful to program the call history access button particularly for missed calls. After programming the button, you will be notified of a missed call by a flashing LED and also on your phone display.

Record Processing OptionsAll call register items are displayed when you enter the appropriate list of calls. Use

the and arrows to browse through the records. Push the confirmation button to display the submenus.


Call – press the key to set up a call to the selected record.

Send message – press the key to move to the SMS editing section where the addressee's number has already been entered.

Details – display additional information on the selected call (name, number, type, date, time, duration).

Delete – delete the record. Confirmation is required before deletion.

Save to list – save a record number into the user telephone directory under a selected name.

Received Calls and Dialled NumbersThe way of control of these two menus is completely identical with that of the missed call menu. You can make calls and/or send text messages to the records and view details on the records.

Delete Recent Call ListUse this option to easily delete a selected call history list or all lists at once. Confirmation is required before deletion. Once deleted lists cannot be recovered.


Tip

By deleting the telephone call history you also delete the same items

displayed in the interface.2N® NetStar Assistant

SettingsThe menu offers the following display options for your call history records:

Name and time – the name and call date, time and duration are displayed.

Number and time – the number and call date, time and duration are displayed.

Name list – names are displayed only.

Number list – numbers are displayed only.


10.3.3 Messages

Menu Structure

You can send and receive text messages using the telephones. Refer to 2N® StarPoint

for sending SMS and to for receiving SMS. The figure below Create Message Inboxshows the menu structure. Push the following sequence to get into this Messages

menu at relax: > > .

Tip

You can also use the web application for 2N® NetStar – Assistant

sending and receiving SMS. Refer to the 2N® NetStar Assistant Manualfor more details.


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Create Message

Use the and arrows to select the item in the Create message Messages

menu and press for confirmation.

Now type any text using your numerical keypad. Press the keys repeatedly to type different letters. Push the button to switch to digits or capital letters. To #enter such special characters as *, # a + push the * button repeatedly. Punctuation and other special symbols are available under the button.1

Having completed the message, push for confirmation.

Now choose whether to send the message to a number, email, or a selected telephone directory contact, or return into the editing mode. If you choose Send

, use the and buttons to browse through the contacts.to contact

Having chosen a number, email address or contact, press to send the message.

InboxTo read a received message:

Push while an incoming message is signalled on the display.

Use the programmable button with the function. Refer to Received messages for button programming.10.4.2 Programmable Buttons - Setting

Or, access the messages through the telephone main menu. Enter the following

sequence at relax: > > > > .

Message Processing Options

Press the and arrows to browse through the received messages. All new

messages are denoted before the sender's name until read. Press to read the New

message and and to scroll down in long messages. In addition to a text, a message contains some details, such as the sender's name and number, station type

and date&time. Push again while reading your message to display the details.


Answer message – here you can easily send a reply to the number that sent the

message. Type the reply and push to send the message.

Resend message – here you can alter and forward the message to another user.

Delete message – here you can delete the message. Confirmation is required before deletion.

Bulk Message Delete

Enter the following sequence at relax to enable mass deleting of messages: >

> > 2x > . The following options are available:


Use the and buttons to browse through these options. Push for confirmation.

Delete all – delete all user messages. Confirmation is required before deletion.

Delete all read – delete all user messages that have been read (those which are not marked New). Confirmation is required before deletion.

Delete all unread – delete all user messages that have not been read (those which are marked New). Confirmation is required before deletion.

Tip

Being deleted, the messages are completely removed from the system

and cannot be recovered even using the 2N® NetStar – Assistant

interface. Therefore, be careful while using this function.

Message SettingsThe menu offers several display options for received messages. They are as follows:

Name and time – the name and receiving time are displayed in the list of received messages.

Number and time – the number and receiving time are displayed in the list of received messages.

Name list – senders' names are displayed only.

Number list – senders' numbers are displayed only.

10.3.4 Contacts

Telephone Directory in General

In the system, you can be assigned telephone directories by the system 2N® – NetStar

administrator or create your own telephone directories using your telephone or the 2N

web application enviroment. Subsequently, you can search the ® – NetStar Assistant

directories and make calls or send SMS to selected items.


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Access to Telephone DirectoryIn general, you can access the telephone directory in your telephone as follows:

Push when your phone is at relax.

Use a programmable button with the function. Refer to Phone directory 10.4.2 for button programming.Programmable Buttons - Setting

Or, access the directory through the telephone menu. Enter the following

sequence at relax: > 2x > 2x .

The first two items (alphabetically) are displayed when you enter the phone directory.

Again, use and to browse through the items.

Telephone Directory SearchYou are recommended to use the search function for extensive directories. Press any key on the numerical keypad while you are in the directory to activate the function.

Letters are entered using the numerical keypad – each numerical key is assigned defined letters (see the button tagnames). Each letter in the sequence is typed by a specific count of button presses. For example, letter requires two presses of the E 3button. Do not enter the next character for searching until the display cursor moves. You can use the button if pre-programmed to delete characters.ESC

Whenever a character is entered, the telephone directory records are filtered automatically and you can browse through the filtered items using the arrows.

The first item containing the required character string is always displayed:

Use and to access the following items. If no item matches the string entered, the 'No match' message is displayed:

Push the programmable button repeatedly to terminate filtering.ESC


Record Processing Options

Use and to browse through the phone directory items. Push to display the contact submenu.

Call – press the key to set up a call to the selected record.

Tip

Remember that phone directory email items cannot be called but can only be sent messages.

Send message – press the key to enter the message editing section with the sender's number pre-defined.

Details – display complete information on the contact (name, number, prefix, email).

Edit – edit contact details sequentially. Push to move to the next contact detail.

Delete – delete the phone directory items in singles. Confirmation is required before deletion.

Tip


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By deleting a global contact from your telephone you delete the same from the list of all users and so be very careful while using this function.

Menu Structure

BrowseSelect this item to enter the telephone directory.

Add Number and EmailAdd number – add an item to the phone directory to be called or sent messages. To add an item:

Enter the name and push for confirmation.

Enter the contact number or URL and push for confirmation.

Select a prefix from the menu using the and buttons or confirm None

with .

The record will be stored in your user directory.

Add email – add an item to the phone directory to be sent only text messages, which are transformed into email messages. To add an item:


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Enter the name and push for confirmation.

Enter the contact email address and push for confirmation.

The record will be stored in your user directory.

SettingsThe menu offers two display options for the phone directory records. They are as follows:

Name and number – the contact name and number are displayed in the directory.

Name list – contact names are displayed only.

10.3.5 Profiles

About Profiles

A variety of properties can be assigned to stations or users (profiles) in the 2N®

system for convenience. Subsequently, you can switch the profiles depending NetStaron time or user selection in order to change station forwarding, pre-programmable buttons or telephone/HandsFree volume quickly and easily. All user changes are added to the curently active profile. Each user may create up to eight profiles, which, together with the basic user setting, represents nine unique configuration settings.

Tip

The web application provides the user with 2N® NetStar Assistant

extended profile editing options including time condition creation and

links to selected profiles. Refer to the for 2N NetStar Assistant Manual®more details on web interface configuration.

Menu StructureTo access the profile menu:

Use the programmable button with the function. Refer to Profiles 10.4.2 for button programming.Programmable Buttons - Setting


2. Find the menu in the main menu. Enter the following Profiles 2N® StarPoint

sequence at relax: > 3x > .

See the figure below for the menu structure. Push and to browse through the

menu items and available profiles. Press for selection.

Profile Options

Having selected a profile in the menu and confirmed it with , you get into Profiles

a profile processing submenu. Use and to browse through the profile options. The figure below shows the profile processing structure.


Activate profile – enable the selected user profile.

Tip

You can also assign user profile activation to a programmable button.

Delete profile – delete the selected user profile. Confirmation is required before deletion.

Rename profile – rename the selected profile. A profile name may not occur more than once for one user.

Automatic SwitchingYou can switch profiles manually on your telephone, using a service, or bind them with time conditions and activate automatic switching. Time condition binding can be

made by the system administrator and/or the user through the 2N® NetStar Assistant

web application, where the global time conditions can be used or new user conditions

can be created as necessary. Refer to the for more 2N NetStar Assistant Manual®details on profile configuration.


To activate automatic profile switching from your telephone, select 2N® StarPoint

in the menu using and and press for Automatic switching Profilesconfirmation. Automatic profile switching according to time conditions will be activated automatically.

Tip

By activating a profile manually in the automatic profile switching mode you disable automatic switching.


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Create New Profile

A profile may be created by the system administrator or a user using the 2N®

terminal or the web application. Each user may create StarPoint 2N® NetStar Assistant

up to eight profiles.

To create a new profile:

Use the and buttons to find the selection in the Create new profile

menu and push for confirmation.Profiles

Enter the new profile name and confirm with . A profile name is not allowed to occur more than once for one user.

A new profile will be created if you have entered a correct name and have not exceeded the limit of 8 profiles.

10.3.6 Settings

Menu Structure

Use the following sequence at relax to enter the menu: > 2x Telephone settings

> . The figure below shows the main items of the menu. Use the and

arrows to browse through the items. Push to confirm selections.


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Tip

All the settings included in this section can also be made from the 2N®

web application, which provides a higher user comfort. NetStar Assistant

Refer to the for more details.2N NetStar Assistant Manual®

Volumes

Use the and arrows to adjust the volume levels in all submenus. Push for confirmation.

Handset volume – set the handset volume level (i.e. how you can hear the opponent with your handset picked-up).

HandsFree volume – set the HandsFree volume level (i.e. how you can hear the opponent in the HandsFree mode with your handset hung-up).

Ringing volume – set the phone ringing volume (i.e. how loud the telephone should ring to announce an incoming call).

Keys volume – set the phone button volume (i.e. how loud a keystroke should be signalled in the handset).

Presentation TextPresence text is a short message to be displayed to calling users in case the calling

party uses a display-equipped terminal. Typically, each user profile has 2N® StarPoint

a unique presence message announcing whether or not the user is currently available.

To set the presence text:

Find in the menu and press for Presence text Telephone settingsconfirmation.

Type the presence text using the numerical keypad buttons. Use the button to #type capital letters and digits. Press the button to enter special characters.1

Confirm the complete text with .

Tip

If you want to set a presence text for a profile, make sure that the profile is activated. Refer to for details on profile activation.10.3.5 Profiles


Other

Use and to find the item in the menu and push Other Telephone settingsfor confirmation. See the figure below for the menu structure and options.

Display Contrast

Use the item to change the terminal display contrast. Push and Display contrast

to execute a change. You can currently check the effect of the changes. Confirm

the selected level with .

Time SchemeUse the item to change the time format to be displayed: twenty-four or Time scheme

twelve hours. Push and to execute the change and confirm the selection with

.


Active LanguageUse the item to change the terminal user language. Push and Active languageto select one of the PBX supported languages. English and Czech are supported by

default. Confirm your selection with .

Tip

If you want to use a language that is currently not supported by your system, contact your PBX administrator.

Standard Ring ToneUse the item to change your terminal ringing tune. Push and Standard ring tone

to execute the change. You can currently check the effect of the changes thanks

to sound examples. Confirm the selected tune with .

Information TypeUse the item to change the information displayed in the bottom Information type

display row when the terminal is at relax. Push and to select the required

information type and push for confirmation. The following options are available:

Name – the station name is displayed to the left in the bottom row.

Name and number – the station name is displayed to the left and the station number to the right in the bottom row.

Name and profile – the station name is displayed to the left and the current profile to the right in the bottom row.

Number – the station number is displayed to the left in the bottom row.

Number and profile – the station number is displayed to the left and the current profile to the right in the bottom row.

Profile – the current profile name is displayed to the left in the bottom row.

The complete information on a telephone at relax also includes the current date and time in the selected format.


Incoming ViewUse the item to change the displayed information to signal an Display incoming

incoming call. Push and to select a new item and confirm the setting with . The following options are available:

Called – the called number is displayed.

Called, calling – the calling name (or number) and called number are displayed alternately.

Called, calling list – the calling name (or number) and called name (or number) are displayed alternately.

Key TimeoutUse the item to change the interval for entering subsequent characters Editing delay

in the system. Push and to select one of seven pre-defined 2N StarPoint®

levels. Confirm your selection with .

System InformationTo display system information on the PBX, select the item in the System information

menu using the and arrows and press for Telephone settingsconfirmation. The current PBX firmware version and uptime value are displayed.


10.4 Button FunctionsHere is what you can find in this section:





Numerical Keypad Buttons

The numerical keypad is used for dialling numbers, typing text 2N® StarPoint

messages and entering contacts into the telephone directory. Therefore, multiple characters are assigned to each key. Press a key repeatedly to proceed to the next character. The table below gives a clear overview.

Dialling numbers – you cannot switch the terminal into the editing mode while dialling since it is useless. If you need special characters, press * and #.

Editing text messages – you need the most comprehensive set of characters to edit text messages. The editing mode is enabled by default and you can change it any Abctime by pressing #.

Entering contacts – the editing (Abc) and numerical (123) modes are enabled automatically depending on whether you are entering a contact name or a contact number. Press # to switch the mode any time.


Menu Browsing

Use the , and buttons to browse through the menu. The former two ones are used for browsing through menu items and often for selecting a level. They can also move the cursor through the text during editing text messages and other items. The last-named button is designed exclusively for confirmation of selections and settings.

Tip

You are recommended to assign the function to a programmable ESCbutton to easily quit the current configuration, delete characters and edit items.

Volume Control

Use the and buttons to control the volume level of your 2N® StarPoint

terminal. These buttons control the volume of ringing, handset and HandsFree according to the mode that is currently enabled in the terminal.

Volume adjustment:

Ringing – adjust the ringing volume for incoming calls.

Handset – adjust the handset volume if the handset is picked-up, i.e. durign a call or outgoing call setup.

HandsFree – adjust the HandsFree volume in the HandsFree mode, i.e. during a call or outgoing call setup in the HandsFree mode.

Programmable Buttons

All terminals are equipped with programmable buttons, which can be 2N® StarPoint

assigned any function. To increase the number of available buttons, connect button consoles with 16/90/18 programmable buttons.

The basic function of some programmable buttons is pre-defined by your PBX administrator, but you can change their settings according to your needs.

Tip

Refer to for button programming. 10.4.2 Programmable Buttons - SettingRefer to for functions that can 10.4.3 Programmable Buttons - Functionsbe assigned.


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Button Function Assignment

The system administrator or a user may program the 2N® NetStar Assistant 2N®

programmable buttons. To assign a function to a programmable button:StarPoint

Log in to the web interface.2N® NetStar Assistant

Select the menu in the upper toolbar.Telephones

Select the type in the menu.2N® StarPoint

Click on the button to be programmed in the terminal figure to open the window as shown in the figure below.

Select the required function in the menu. Complete the other active Functionwindows if necessary (station name, user name, number, etc.) and click on

for confirmation.Change

The button is now ready for use.

Tip

Refer to for details on the 10.4.3 Programmable Buttons - Functionsfunctions to be assigned.



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Tip

The functions of some buttons are pre-defined by your system administrator. Typically, they include (Esc), , Back Redial Access to

, (Flash), and . Therefore, missed calls Call holding Call slots HandsFreeyou do not have to assign these functions any more.

Button Tagnames

The system is equipped with the automatic button tagname 2N® NetStar Assistant

creating function.

Procedure:

Click on the item on the main telephone setting screen.Prepare for printing

This creates and displays a complete template in the PDF format. The template

consists of tagnames defined and displayed in the 2N® NetStar Assistant

environment.

Tip

Make sure that a suitable PDF file browser has been installed for template viewing and printing (e.g. AdobeReader).


Activate ProfileUse the function to easily activate and deactivate a user profile.

Use:

Push the button with this function to activate the profile. The LED indicator of the respective button is shining while the profile is active. Re-push the same programmable button to deactivate the profile, or activate another profile, thus deactivating the current profile automatically.


CLIR (Identification Restriction)CLIR means Calling Line Identification Restriction. Use the function to hide your outgoing identification during call setup.

Use:

Push the function button before dialling the number (before picking up the handset). The active CLIR state is indicated by a shining button LED. Identification will be restricted for all subsequently set-up calls. Re-push the button to cancel the function.

Tip

The resultant effect of CLIR depends on additional settings of your PBX for calls outside the PBX. Consult your system administrator in the event of error.

DefaultThe programmable button with this "function" helps you use the functions defined by the PBX administrator. These functions can be "inherited" by connected terminals unless the user sets a function of its own that is always preferred to the global settings of the PBX.

Esc (Back)Use the function primarily to easily quit a menu, cancel current actions, delete edited symbols and reject incoming calls. It is one of the most important functions of the terminal, which improves the user comfort above all.

Use:

Whenever the terminal gets into the state in which this function button can be used, the respective LED indicator will go on.

Flash (Call Holding)Use the function to put a call on hold for call transferring or consulting purposes. The function also enables active and held calls to be switched.

Use:

Push the function button during a call. The call will be put on hold and you will hear the dialtone and can dial another station number. The opponent will hear music. Having set up the other call, you can push the same button to switch the calls.

HandsFree/Speakerphone


HandsFree/SpeakerphoneUse this function to make HandsFree calls. Some terminals are not equipped with an external microphone and cannot make full use of this function. For such terminals the

function is available. This function uses the standard handset Speakerphonemicrophone instead of an external microphone and the call quality is usually lower.

Use:

If you push the button during an incoming call, you will answer the call in the HandsFree mode, which is signalled by the button LED. Re-push the button to terminate the call.

If you push the button during a call with your handset picked-up, the telephone switches into the HandsFree mode too. This is signalled by the button LED. Now you can hang up the handset without terminating the call.

Call SlotUse the function to make multiple calls from one terminal by holding the previous call and setting up a new call using another call slot. By default, each terminal has three call slots.

Use:

Push the unused ("switched-off") call slot button during a call. The active call will be put on hold as with the Flash function, you will hear the dialtone and the held user will hear pre-defined music. If there are more calls in your terminal call slots, you can switch them using the respective call slot buttons. The active call slot LED is shining while the held call slot LED is flashing. The unused call slot LED is off.


IntercomIntercom is a special type of call where a predefined telephone number is dialled by pressing the Intercom function button and the called station receives the call automatically in the HandsFree mode. The function is suitable for flexible communication – you can speak the moment you push the button.

Tip

The Intercom function works with the telephones 2N® StarPoint 500

only; if a different telephone type is used, the call is not answered automatically.

Use:

Press and keep pressed the button with the Intercom function. Start speaking when you hear a beep. The called user answers the call automatically in the HandsFree mode. The called party gets an acoustic signal before the call. Keep the button pressed during the whole call. Release the button to terminate the connection.

MuteUse the function to temporarily mute the microphone during a call.

Use:

Push the function button during a call. The opponent cannot hear you. The muted state is signalled by a flashing button LED indicator. Re-push the button to re-activate the microphone.

Call RecordingUse the function to enable call recording. This function is based on the PBX administrators's permission to record.

Use:

Push the button with the recording function any time during a call. The active state is indicated by the button LED. Re-push the button to stop recording.


Do Not Disturb (DND)Use the function to switch the terminal into the DND (Do Not Disturb) mode. In this mode, outgoing calls can be set up, but incoming calls are rejected. Missed calls are shown on the telephone display.

Use:

Push the button with the DND function. The active state is indicated by the button LED. Re-push the button to deactivate the function.

ProfilesUse the function to set a quick access to the menu.Profiles

Answered CallsUse the function to set a quick access to the list of answered calls.

Received MessagesUse the function to set a quick access to the list of received messages.

RedialUse the function to redial the last successfully dialled number (incorrectly dialled numbers are ignored).

Use:

Push the button with the Redial function and the terminal sets up a call to the last-dialled number automatically. If the handset is hung up at the moment, the call is set up in the HandsFree mode.

State/Speed DiallingThe button with the State function only enables you to monitor the activity of a selected user, station or port. In that case, the button LED indicates the following states:

Object at relax – the LED is off.

Object ringing – the LED is flashing quickly.

Object busy with call – the LED is shining continuously.

You can also use the State button for . This function helps you dial a Speed diallingselected number by pressing a single button. You can assign both user numbers and service access codes to the button.


Use:

Push the pre-defined button to dial the number. If the handset is hung up at the moment, the call is set up in the HandsFree mode.

Tip

The state and speed dialling functions can be combined as necessary. For example, you can monitor a station other than the speed-dialled one.

Phone DirectoryUse the function to set a quick access to the phone directory.

Outgoing CallsUse the function to set a quick access to the list of outgoing calls.

Missed CallsUse the function to set a quick access to the list of missed calls.



About 2N ® OptiSet Phones

The telephone sets are digital key system phones designed for the 2N® OptiSet 2N®

system. They represent an earlier type of telephones intended for common NetStaroffice operations. The phones are equipped with programmable buttons and, with the

exception of , a two-row display.2N® OptiSet Entry

2N ® OptiSet Types

Figure: 2N® OptiSet Entry, OptiSet Standard, OptiSet Advance2N® 2N®

Description of Types

2N ® OptiSet EntryThe telephone:

is intended for basic use by undemanding users;

is not equipped with a display, which excludes telephone-based configuration and/or use of a telephone directory;

is equipped with three programmable buttons;

cannot be extended with an additional button module;

does not support HandsFree and/or Speakerphone.

2N ® OptiSet StandardThe telephone:

is equipped with a two-row display;



is equipped with twelve programmable buttons;


supports HandsFree.

The , and buttons are used for menu browsing

and configuration. The terminal is controlled identically as the 2N® StarPoint

terminals.

2N ® OptiSet AdvanceThe telephone:

is intended for demanding users, assistants and operators.


is equipped with twelve programmable buttons;

can be extended with an additional button module;

supports HandsFree.

The , and buttons are used for menu browsing

and configuration. The terminal is controlled identically as the 2N® StarPoint

terminals.



About 2N ® OpenStage Telephones

The telephone sets are digital key system phones designed for the 2N OpenStage® 2N

system. They represent an innovative type of terminals, which can be widely ® NetStar

used thanks to a variety of types.

2N ® OpenStage Types

Figure: ,2N® OpenStage 10T 2N® OpenStage 15T

Figure: 2N® OpenStage 20T, 2N® OpenStage 40T


Description of Types

2N ® OpenStage 10TThe telephone:

is intended for basic use by undemanding users;

is not equipped with a display, which excludes telephone-based configuration and/or use of a telephone directory. Therefore, the phone is not provided with menu browsing buttons.

is equipped with three programmable buttons;


has dedicated buttons for:

– HandsFree activation;

and – volume control.


is equipped with a display;

is equipped with eight programmable buttons;



– access to call history;

– access to SMS messages (reading/viewing);


and – volume control.

Control:

The buttons are used for menu browsing and configuration.

corresponds to

corresponds to

corresponds to

2N ® OpenStage 20T



is equipped with an adjustable display;

has no programmable buttons;






and – volume control;

– call holding (Flash);

– Redial;

– Do Not Disturb (DND);

– Mute function.

Control:

The cross buttons are used for menu browsing and configuration.

corresponds to

corresponds to

corresponds to



is equipped with an adjustable six-row display;

has six programmable buttons;






and – volume control;

– call holding (Flash);

– Redial;

– Do Not Disturb (DND);

– Mute function;

– head set activation.

Control:

The cross buttons are used for menu browsing and configuration.

corresponds to the Back (Esc) button. It is used for rejecting calls, deleting texts or quitting menus.

corresponds to

and correspond to

corresponds to


10.7 Glossary

2N NetStar Assistant® – a user web interface accessible via a web browser; helps

the user view call history, program telephone buttons, set profiles, manage phone directories, and so on. Your PBX administrator will communicate the interface address and login data to you.

BLF module – a large button module (operator's console) with 90 programmable buttons.

CLIR – Calling Line Identification Restriction.

DND (Do Not Disturb) – all incoming calls are rejected and outgoing calls are set up without limitations in the DND mode.

Escape (Back) – return to a telephone menu.

Flash (Call Holding) – used for accessing services during calls for call forwarding purposes, for example.

Handsfree/Speakerphone – a type of making calls where an external microphone and a loudspeaker are used for speaking. The Speakerphone mode uses an external loudspeaker and the handset microphone.

Intercom – a function for flexible communication – calls are answered automatically by the called station.

KEY module – a small button module (operator's console) with 16 programmable buttons.

Mute – temporary microphone muting during a call.

Redial – redialling of the last successfully called number.

Station – a terminal device (telephone set, e.g.) assigned to the user and represented by a telephone number.

Telephone line – a set of technical parameters providing connection of a telephone station to a PBX (cables, sockets, etc.).

Telephone directory – a list of users/stations and their numbers as defined by the administrator or user.

Dial selected number – the pre-selected dialling executed by the button press. A telephone number or service access code can be selected.



The telephone may not be operated in explosive environments.

Connect the telephone to dedicated sockets only.

Keep the telephone away from aggressive substances and staining liquids (such as coffee, tea, wine, etc.). Never use petrol, solvents, alcohol containing detergents and similar substances for telephone cleaning.

Use a soft cloth dampened with water or diluted washing-up detergents to clean the telephone. Microfibre cloth is recommended for display cleaning.

Certificate of ComplianceThese devices meet all technical requirements concerning radio and telecommunications equipment pursuant to Act. No. 22/1997 Coll. in the latest wording and Governmental Decree No. 426/2000 Coll. and have been awarded a Certificate of Compliance.

CE Marking

The CE marking certifies compliance with EU Directive 1999/5/EG.

Environmental Marking

This marking certifies that the device was manufactured in compliance with the ISO 14001 Environmental Management Standard. This guarantees raw material, energy and waste minimisation during manufacturing.

The manufacturer reserves the right to modify the product in order to improve its qualities.

Use the product for no purposes other than those specified in the User Manual.When the service life of the product or a part thereof is exhausted, dispose of it in accordance with applicable environmental regulations.


11. Manuál 2N® OpenStage 30TThis page has not yet been translated to English.

11.1 PředstaveníThis page has not yet been translated to English.

11.1.1 O telefonech 2N® OpenStageThis page has not yet been translated to English.

11.1.2 Popis jednotlivých typůThis page has not yet been translated to English.

11.1.3 PříslušenstvíThis page has not yet been translated to English.

11.2 Základní ovládáníThis page has not yet been translated to English.

11.2.1 Odchozí voláníThis page has not yet been translated to English.

11.2.2 Příchozí voláníThis page has not yet been translated to English.

11.2.3 Během hovoruThis page has not yet been translated to English.

11.3 Menu telefonu a jeho funkceThis page has not yet been translated to English.

11.3.1 Hlavní menuThis page has not yet been translated to English.


11.3.2 Výpis voláníThis page has not yet been translated to English.

11.3.3 ZprávyThis page has not yet been translated to English.

11.3.4 Telefonní seznamThis page has not yet been translated to English.

11.3.5 ProfilyThis page has not yet been translated to English.

11.3.6 Nastavení telefonuThis page has not yet been translated to English.

11.4 Funkce tlačítekThis page has not yet been translated to English.

11.4.1 Základní tlačítka termináluThis page has not yet been translated to English.

11.4.2 Programovatelná tlačítkaThis page has not yet been translated to English.

11.4.3 Funkce programovatelných tlačítekThis page has not yet been translated to English.

11.5 Důležitá upozorněníThis page has not yet been translated to English.


12. Manuál 2N® OpenStage 40TThis page has not yet been translated to English.

12.1 Základní informaceThis page has not yet been translated to English.

12.1.1 O telefonuThis page has not yet been translated to English.

12.1.2 Popis telefonuThis page has not yet been translated to English.

12.1.3 Příslušenství 2N® OpenStageThis page has not yet been translated to English.

12.2 Základní ovládání 2N® OpenStage 40TThis page has not yet been translated to English.

12.2.1 Odchozí voláníThis page has not yet been translated to English.

12.2.2 Příchozí voláníThis page has not yet been translated to English.

12.2.3 Během hovoruThis page has not yet been translated to English.

12.3 Menu telefonu a jeho funkceThis page has not yet been translated to English.

12.3.1 Hlavní menuThis page has not yet been translated to English.


12.3.2 Výpis voláníThis page has not yet been translated to English.

12.3.3 ZprávyThis page has not yet been translated to English.

12.3.4 Telefonní seznamThis page has not yet been translated to English.

12.3.5 ProfilyThis page has not yet been translated to English.

12.3.6 Nastavení telefonuThis page has not yet been translated to English.

12.4 Funkce tlačítekThis page has not yet been translated to English.

12.4.1 Základní tlačítka termináluThis page has not yet been translated to English.

12.4.2 Programovatelná tlačítkaThis page has not yet been translated to English.

12.4.3 Funkce programovatelných tlačítekThis page has not yet been translated to English.

12.5 Důležitá upozorněníThis page has not yet been translated to English.



This section describes the service, which allows external 2N Mobility Extension®stations (outside the communication system) to make use of all advantages of the 2N®

internal stations. With the you are available under NetStar 2N® Mobility Extension

your company number even if you are out of your office. Moreover, you can make outgoing calls including outgoing identification as if you called from a fixed station.


13.1 Incoming Calls

13.2 Outgoing Calls

13.3 Call Transfer

13.1 Incoming Calls

An incoming call to your phone number in the system will make your 2N® NetStar

Mobility Extension station ring like your internal station. An incoming call to your mobile phone will be identified by the GSM gateway number or a public line number.

Note

Upon an incoming call to the Mobility Extension station, an outgoing call

from to your mobile phone will be set up (and charged 2N® NetStar

accordingly).

ME Station Ringing On/OffTo enable the Mobility Extension station ringing during an incoming call, dial the ME service access code from the mobile phone to which the calls are routed. After a pre-recorded message is played, select the activation code mentioned below. The deactivation procedure is the same.

Enable ringing

*55

Disable ringing


#55

Example: Suppose you want to turn on ringing on your GSM phone. Call the communication system from your station. Your Mobility Extension will be recognised automatically and a pre-recorded message "Mobility Extension, dial please" will be played for system login confirmation. Now dial the service access code *55. You will be informed of the ringing enable and the call will be terminated.

Note

You can use the ringing enable/disable function for any user station and thus cancel the station's ringing temporarily in the event of call routing to the user.

13.2 Outgoing Calls

Outgoing calls here mean outbound calls set up via the system. This type 2N NetStar®of communication guarantees company identification for all outgoing calls and lower call costs for outgoing international calls.

Outgoing Calls to Internal StationTo call from your Mobility Extension, dial the Mobility Extension service access code.

Your login to the system will be confirmed by a pre-recorded message 2N® NetStar

and subsequently by the continuous dialtone from your communication system. Now you can dial a number as if you used a fixed internal line.

Example: Suppose you want to talk to an internal subscriber number 2345. Call the communication system using your mobile phone. Your login will be confirmed by a pre-recorded message "Mobility Extension, dial please". Now dial the internal subscriber number 2345. The call will be connected.

Outgoing Calls via communication system to PSTNYou can make outgoing calls to a public network (depending on your rights) as if calling to an internal line. For the outgoing call you are identified by the phone

number you are assigned within the communication system.2N® NetStar

Note

For an outgoing call from the Mobility Extension station, you are


For an outgoing call from the Mobility Extension station, you are

charged the call from a mobile station to the 2N® NetStar

communication system and the communication system is 2N® NetStar

charged the outgoing call to the called number.

Example: Suppose you want to talk to an external subscriber with number 654654654. Call the communication system using your mobile phone. Your login will be confirmed by a pre-recorded message "Mobility Extension, dial please". Now dial the external line access code (typically '0') and the called subscriber number, i.e. 0654654654.

13.3 Call TransferWhen you use the Mobility Extension station, you can transfer calls as if you used an internal fixed line. Unlike with fixed stations, use the code 7* for transfer to the Mobility Extension, which is an equivalent to the FLASH key. Calls can be transferred

to phone numbers both inside and outside the communication system. 2N® NetStar

Using the call transfer code 7* you can also enable the Call Swapping or Return to Transferred Call services from your Mobility Extension.

Call Transfer to Mobility ExtensionLike with an internal station, transfer to the Mobility Extension can be with or without hang-up. One and the same transfer code is used in either case but the transferring procedures are slightly different. Dial:

7*nn

where nn is the number of the subscriber you want to transfer the call to.

Transfer with Hang-UpExample: Suppose you want to transfer (with hang-up) an active call from your Mobility Extension. During the call dial 7* and the selected phone number. The caller hears music and is waiting. If you hear the alert tone, you can hang up to transfer the call to the selected subscriber and the person under transfer hears the alert tone. If you hear the busy tone, no call transfer can be made as the selected destination is busy or unavailable. In that case return to the person to be transferred by re-entering 7* and either end the call or transfer it to another subscriber.


Transfer without Hang-UpExample: Suppose you want to transfer (without hang-up) an active call from your Mobility Extension. Dial 7* and the selected phone number during the call. The caller hears music and is waiting. If you hear the alert tone, wait till the transferee answers the phone and then either hang up to complete the transfer, or enter 7* to return to the original call. If you hear the busy tone, the transfer cannot be completed as the selected destination is busy or unavailable. In that case return to the person to be transferred by re-entering 7* and either end the call or transfer it to another subscriber.

Returning to Transferred Call or Call SwappingUse this service when you have put one call on hold and are speaking to another subscriber in an active call. Dial:

7*

Example: Suppose you are calling to the communication system from your Mobility Extension and have put one call on hold and are making another call. To activate to the subscriber on hold push 7*. Now you are speaking to the subscriber originally put on hold and the other subscriber hears the Music on Hold. By re-pressing 7* you swap the two calls.

Active Call TerminationBasically, there are two situations in which you can use this service. In the first case you have one call on hold and want to terminate the current call, and in the other case you have an active call and want to dial another destination after terminating the active call.

9#

Return to Call on Hold with Active Call TerminationWhen you have put a call on hold and want to terminate the active call to return to the one on hold, use the Active Call Termination service. Using this service you disconnect the active call (the opposite party hears the busy tone) and get connected to the call on hold.


Example: Suppose that two calls have been established on your Mobility Extension

(both via ). One of the calls is on hold (the subscriber hears the Music on 2N ® NetStar

Hold). To terminate the active call and get through to the call on hold, dial the code 9#.

Active Call Termination with Further DiallingThe case is quite different if you have established one call only and want to terminate

it. After a standard hang-up, you have to call the communication system 2N NetStar®for a new call. To save the call setup costs, use the Active call Termination service.

Example: Suppose you have established just one call on your Mobility Extension and want to terminate it to be free for another internal call. Press 9# during the call. The current call is disconnected and you get the communication system dialtone again. Dial the next subscriber number and a call will be set up like the first call.


Supplementary Information

This section provides supplementary information on the product.2N NetStar®Here is what you can find in this section:

1. Troubleshooting


3. General Instructions and Cautions


1. Troubleshooting

For the most frequently asked questions refer to .faq.2n.cz

http://faq.2n.cz/



Europe

2N NetStar® conforms to the following directives and regulations:

Directive 1999/5/EC of the European Parliament and of the Council, of 9 March 1999 - on radio equipment and telecommunications terminal equipment and the mutual recognition of their conformity

Directive 2006/95/EC of the European Parliament and of the Council of 12 December 2006 on the harmonisation of the laws of Member States relating to electrical equipment designed for use within certain voltage limits

Directive 2004/108/EC of the Council of 15 December 2004 on the harmonisation of the laws of Member States relating to electromagnetic compatibility

Commission Regulation (EC) No. 1275/2008, of 17 December 2008, implementing Directive 2005/32/EC of the European Parliament and of the Council with regard to ecodesign requirements for standby and off mode electric power consumption of electrical and electronic household and office equipment

Directive 2011/65/EU of the European Parliament and of the Council of 8 June 2011 on the restriction of the use of certain hazardous substances in electrical and electronic equipment

Regulation (EC) No. 1907/2006 of the European Parliament and of the Council of 18 December 2006 concerning the Registration, Evaluation, Authorisation and Restriction of Chemicals (REACH), establishing a European Chemicals Agency, amending Directive 1999/45/EC and repealing Council Regulation (EEC) No. 793/93 and Commission Regulation (EC) No. 1488/94 as well as Council Directive 76/769/EEC and Commission Directives 91/155/EEC, 93/67/EEC, 93/105/EC and 2000/21/EC

Directive 2012/19/EC of the European Parliament and of the Council of 4 July 2012 on waste electrical and electronic equipment.

Industry CanadaThis Class B digital apparatus complies with Canadian ICES-003. / Cet appareil numérique de la classe B est conforme a la norme NMB-003 du Canada.

FCCNOTE: This equipment has been tested and found to comply with the limits for a Class B digital device, pursuant to part 15 of the FCC Rules. These limits are designed to provide reasonable protection against harmful interference in a residential installation.


This equipment generates, uses and can radiate radio frequency energy and, if not installed and used in accordance with the instructions, may cause harmful interference to radio communications. However, there is no guarantee that interference will not occur in a particular installation. If this equipment does cause harmful interference to radio or television reception, which can be determined by turning the equipment off and on, the user is encouraged to try to correct the interference by one or more of the following measures:

Reorient or relocate the receiving antenna

Increase the separation between the equipment and receiver

Connect the equipment into an outlet on a circuit different from that to which the receiver is connected

Consult the dealer or an experienced radio/TV technician for help.

Changes or modifications to this unit not expressly approved by the party responsible for compliance could void the user's authority to operate this equipment.


3. General Instructions and CautionsPlease read this User Manual carefully before using the product. Follow all instructions and recommendations included herein.

Any use of the product that is in contradiction with the instructions provided herein may result in malfunction, damage or destruction of the product.

The manufacturer shall not be liable and responsible for any damage incurred as a result of a use of the product other than that included herein, namely undue application and disobedience of the recommendations and warnings in contradiction herewith.

Any use or connection of the product other than those included herein shall be considered undue and the manufacturer shall not be liable for any consequences arisen as a result of such misconduct.

Moreover, the manufacturer shall not be liable for any damage or destruction of the product incurred as a result of misplacement, incompetent installation and/or undue operation and use of the product in contradiction herewith.

The manufacturer assumes no responsibility for any malfunction, damage or destruction of the product caused by incompetent replacement of parts or due to the use of reproduction parts or components.

The manufacturer shall not be liable and responsible for any loss or damage incurred as a result of a natural disaster or any other unfavourable natural condition.

The manufacturer shall not be held liable for any damage of the product arising during the shipping thereof.

The manufacturer shall not make any warrant with regard to data loss or damage.

The manufacturer shall not be liable and responsible for any direct or indirect damage incurred as a result of a use of the product in contradiction herewith or a failure of the product due to a use in contradiction herewith.

All applicable legal regulations concerning the product installation and use as well as provisions of technical standards on electric installations have to be obeyed. The manufacturer shall not be liable and responsible for damage or destruction of the product or damage incurred by the consumer in case the product is used and handled contrary to the said regulations and provisions.

The consumer shall, at its own expense, obtain software protection of the product. The manufacturer shall not be held liable and responsible for any damage incurred as a result of the use of deficient or substandard security software.


The consumer shall, without delay, change the access password for the product after installation. The manufacturer shall not be held liable or responsible for any damage incurred by the consumer in connection with the use of the original password.

The manufacturer also assumes no responsibility for additional costs incurred by the consumer as a result of making calls using a line with an increased tariff.

Electric Waste and Used Battery Pack Handling

Do not place used electric devices and battery packs into municipal waste containers. An undue disposal thereof might impair the environment!

Deliver your expired electric appliances and battery packs removed from them to dedicated dumpsites or containers or give them back to the dealer or manufacturer for environmental-friendly disposal. The dealer or manufacturer shall take the product back free of charge and without requiring another purchase. Make sure that the devices to be disposed of are complete.

Do not throw battery packs into fire. Battery packs may not be taken into parts or short-circuited either.


2N TELEKOMUNIKACE a.s.

Modřanská 621, 143 01 Prague 4, Czech Republic

Phone: +420 261 301 500, Fax: +420 261 301 599

E-mail: [email protected]

Web: www.2n.cz

v4.4

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2N NetStar