® 2N NetStar Communication System Manual NS Admin Version 3.1.0 www.2n.
The 2N TELEKOMUNIKACE joint-stock company 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.
Content 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5 1.1 About Help . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1.2 About Application . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1.3 Connecting to PBX . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1.4 Configuration Menu . . . . . . . . . . . . . . .
6.4 Licences . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 110 6.5 Language Packages . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 113 6.6 Services . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 114 6.7 Conference Rooms . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 116 6.8 Progress Tones . .
1. Introduction Here is what you can find in this chapter: 1.1 1.2 1.3 1.4 1.
1.1 About Help Document format Document version: 3.1.0 This document serves as a Help and Manual for the configuration of the 2N NetStar communication system by the NSAdmin program. 2N reserves the right to modifications. Conventions The following fonts are used in the text: #Hypertext Hyperlink to another place in the document or outside the document. #Document contents item All page headings are listed in the table of content.
1.2 About Application About Application NSAdmin is a configuration tool that helps configure the 2N® NetStar communication system, version 2. The application is designed for an x86 platform using the WINDOWS operating system connected with 2N® NetStar through a network. It is controlled by a mouse and, secondarily, a keyboard. NSAdmin uses the TCP connection or modem and communicates with 2N® NetStar basically via port 6992.
port Id and find the port in the trace. General Set colours of virtual ports – here define the colour for each virtual port or disable this function. Set colours of tables – here define the background colours for tables or disable this function. Set colours of logins – here 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.
1.3 Connecting to PBX Icons of connection section The 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 – use this option to create a group of PBXs on the same level as the selected object or nested into the existing group.
Figure: View of Possible PBX Connection Structuret To change the PBX connection settings use the Properties icon. PBX properties The 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 – use this parameter to define the path to the folder for the configuration to be saved. Autosave – use this option to enable automatic database saving in the off-line mode.
Figure: View of PBX Property Settings Connection properties The 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 – here enter the name of the selected connection. Modes – use this parameter to define whether the connection will support the on-line, off-line or both modes. Download trace – use this option to disable or restrict trace downloading from the PBX.
Figure: View of Connection Property Settings Connecting to PBX After an automatic or manual initiation of a PBX connection, the dialogue shown in 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 Try again button.
Figure: View of Connection Course Dialogue If you are unable to connect to your PBX, please check whether: 1. 2. 3. 4. 5. 6. 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.
1.4 Configuration Menu Main menu After 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 – use this option to logout 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.
import/export dialogue. The csv and xml files are supported. Database import – click on this option to display the database importing window. Select From file or From PBX. If you choose From PBX, all the PBXs available for database import will be displayed. Use the Rule parameter to specify which of the original settings should not be overwritten by database import. The option is available in the off-line mode only. Database export – click on this option to export the PBX database into an xml file.
Figure: View of Configuration Tool Tags and Windows The Trace and Database sections situated in the lower part of the screen above the status bar are important configuration tool components. Trace helps you monitor calls and analyse configuration errors if any. Database provides a direct view of the data stored (depending on the connection mode).
1.5 PBX Activation What you need To activate and configure 2N® NetStar you need the 2N® NetStar system, a computer running the supported Windows version, a keyboard and a mouse. The PC and 2N® NetStar PBX have to be interconnected in a LAN. Furthermore, it is necessary to display 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.
Step 2: Hardware activation After the first connection to the selected PBX according to the Connection to PBX chapter, 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 Dialogue If the wizard is displayed, you can define the basic configuration settings of the virtual BRI ports. If you are not sure, you can push the Next button to proceed to the next 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's Hardware Detection Operations The 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.
Figure: View of Wizard's Localisation Setting Operations Step 4: Time setting In 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's Time Setting Step 5: PBX function selection In 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 GSM gateway, where some steps are omitted and configuration starts as late as the SMTP.
Step 6: Creation of groups, users and extensions In 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.
Figure: View of Wizard's Extension Creation or Intra–Plant Structure Import Step 7: Settings for Assistant This configuration step includes just two functions with the following meanings: Launch web server – this item launches the internal PBX web server, to which you can log in by entering the CPU IP address from your web browser. Generate default logins – this item generates logins for the users created in the preceding step. With them, you can log in to the web server as a user.
Step 10: Creation of routers The 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.
2. Hardware Here is what you can find in this chapter: 2.1 2.2 2.3 2.
2.1 Basic Service mode The 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 OFF and save the changes. Having returned from the service mode successfully, you can see in the section Detected rack in the column Status the state RUN.
Profiles Extenders Extender channels Trunk positions Main case case – digital Main case – analog Detectors Players 0 4 128 128 128 32 32 32 1 0 0 256 64 32 64 64 2 4 32 224 64 32 64 64 3 4 64 192 64 32 64 64 4 4 128 128 64 32 64 64 5 0 0 224 64 64 64 64 6 4 32 192 64 64 64 64 7 4 64 160 64 64 64 64 8 4 128 96 64 64 64 64 Table: Benefits and Disadvantages of Hardware Profiles 28
2.2 Boards Arrangement of HW Unfolding the HW – Rack menu you can see the rack fitting as shown in the figure below. Figure: View of PBX Basic Unit Panel Push the buttons 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.
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 port/resource – use this option to remove and delete a virtual port once forever. You will not be able to use this port any more. Regenerate name – use this option to rename a selected virtual port according to its physical port.
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. PBX HW configuration print The buttons to the right of the basic unit/extender figure help you print out the current PBX hardware configuration. Click on one of the buttons to display the print setting options.
Addressing The position of each board is specified in the R : C : B format and the position of a port in the R : C : B : P format. The characters have the following meanings: R – rack number; C – rack unit number; B – unit board number; P – board port number. Currently, R takes up the value of 0 and C ranges from 1 to 5, the basic unit being 1, the first extender 2 and so on up to the fourth extender with number 5. The board positions (B) in the basic unit are numbered 1 to 14 from the left.
2.3 Synchronisation Upon 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 HW – Synchronisation menu. The left-hand one contains all digital virtual ports that can be selected for synchronisation, i.e. all PRI and BRI virtual ports in the TE mode.
2.4 Board and Port List The Hardware – Board list menu contains a list of boards that are physically present in the PBX. The board list has four columns with the following meanings: Address – shows the physical board address within the PBX according to the Boards chapter. 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.
3. Virtual Ports Here is what you can find in this chapter: 3.1 3.2 3.3 3.4 3.5 3.6 3.7 3.
3.1 BRI and PRI BRI Refer to the Boards section for the meaning of the virtual port. 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/PRI menu in the Stack tag. A list of all BRI virtual ports is displayed on the left and a window for the port parameter setting is available on the right. The configuration parameters are divided into logical parts.
upper level, a red exclamation mark appears on the port in the Hardware – Boards menu and a red text is displayed in the upper stack status field. This error status gets changed after the SLIP rate falls below the lower level. The interval between these two values represents hysteresis. Settings for BER – select BER as error to enable acceptable interface error parameters.
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 – use this option to 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 – use this option to enable call establishment without an explicitly set B–channel.
Tab Expert Cause mapping – is an additional function for modification of outgoing causes for a selected virtual port. It may be useful while adapting NetStar causes to the specific conditions of your network. You can choose a particular internal NetStar cause in the left–hand column and assign a cause to it to be sent to your network in the right–hand column. PRI Refer to the Boards section for the meaning of the virtual port.
SLIP range parameters. Use this option in the TE mode only. If the SLIP rate gets over the upper level, a red exclamation mark appears on the port in the Hardware – Boards menu and a red text is displayed in the upper stack status field. This error status gets changed after the SLIP rate falls below the lower level. The interval between these two values represents hysteresis. Settings for BER – select BER as error to enable acceptable BER range parameters.
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 – use this option to 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 – use this option to enable call establishment without an explicitly set B–channel.
Expert tab Cause mapping – is an additional function for modification of outgoing causes for a selected virtual port. It may be useful while adapting NetStar causes to the specific conditions of your network. You can choose a particular internal NetStar cause in the left–hand column and assign a cause to it to be sent to your network in the right–hand column.
3.2 Cornet Cornet is a digital virtual port for the StarPoint key phones with proprietary signalling (UPN interface). The Stack tag 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 subtag of the Properties tag. For more details on Softphone extensions refer to Chapter Setting Properties.
Digital interface diagnostic 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.
3.3 ASL The ASL virtual port is used for connecting common analogue telephones or fax machines. 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 status This field displays information on the stack and its current status.
Incoming parameters (phone is dialling) Call type – determines the preferred type of communication on this port. Choose one of the Voice, FAX, A3.1kHz Audio and 56kb Modem options. DTMF dial enabled – check this item to make the carrier detect DTMF dialling from an analogue phone. Pulse dial enabled – check this item to make the carrier detect pulse dialling from an analogue phone. FLASH length [ms] – the parameter sets the maximum time of the FLASH signal transmitted from a local phone to the PBX.
3.4 CO The CO virtual port is an analogue virtual port for connection of a CO (central exchange) 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.
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 Reverse polarity, Tariff pulse or Simple options. Tariff pulse type – defines the tariff pulse sending source. Choose 12 kHz, 16 kHz or none.
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 alerting detection.
ACIM [3:0] 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.
3.5 GSM The Virtual ports – GSM menu provides a list of all GSM virtual ports of the PBX. The parameters are divided into logical sections. Basic Stack status This 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 – use this option to enable roaming for a GSM virtual port.
Signal diagnostics Signal measuring – select this option to enable signal level measuring for a selected carrier. Signal monitoring – here enable signal level monitoring for a selected carrier. If the signal level drops below the value specified in the Poor signal level parameter, a red exclamation mark appears on the port in the Hardware – Boards menu and a red text is displayed in the upper stack status field.
GSM interface parameters CLI mode – use this parameter to 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 restricted regardless of the SIM card or calling user settings.
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, PIN REQUESTED means that the SIM card requires the PIN code to log in.
Expert tag AT commands You 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 Timeout column sets the time during which the answer to the command entered is awaited. The Result columns includes a brief statement on whether or not the command was successful. For specific answers see the Answers for selected section.
USSD USSD commands In this section, you can enter the USSD commands (codes) for pre-paid SIM recharging or credit info display, for example. Click on New to enter the required command and on Repeat to execute the last-entered command. Press Cancel to terminate the current command processing. Refer to the Answer window for the command effect on the USSD code. Network name – the parameter shows the name of the network to which the SIM card is currently logged in. Command – shows the last-entered USSD command.
3.6 SIP SIP Gateway The 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 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.
Figure: View of SIP Gateway Configuration Menu Connect to gateway Host – here 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.
IP filter The 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.
Figure: View of Codecs Setting Menu SIP Proxy The SIP Proxy virtual port is used for connecting SIP terminals to the PBX through terminal registration. All the parameters are divided into logical sections. Basic Stack 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.
Common parameters Port – here fill in PBX port for the SIP Proxy – terminal communication. Realm (Domain) – defines 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.
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 Codec setting options. DTMF according to RFC2833 – use this option to enable DTMF transmission according to RFC2833. With this option on, you can set the Payload type for DTMF transmission using the link below the name. Fax T.
Be sure to set the terminal type and MAC address correctly to make use of the SIP Provisioning function for the 2N® StarPoint IP T2x phones.
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 – use this option to define the way of codec negotiation. If this option is not checked, codecs are offered already in the Invite message. Use short headers – select this option to enable using short headers (e.g.
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 – use this parameter to generate some noise into the call. Since analog line users are used to some background noise, similar noise is simulated here for comfort. Mask lost packets – use this option to activate lost packet masking optimisation.
Miscellaneous In–call mark receivin Mode – use this parameter to set the supported DTMF receiving method for calls. Generating of INFO message DTMF – select one of the two available DTMF transmission modes using the SIP INFO method. The modes provide different formats of the DTMF character transmitting message. Keep–alive Interval – defines the keep–alive packet sending interval. The default value is 10s. STUN server – allows the NAT clients (i.e.
3.7 SMTP SMTP The Simple Mail Transfer Protocol (SMTP) is an Internet protocol used for e–mail 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 routing tag.
Cram_MD5 – Account name – provides the name of the e–mail account registered by the selected SMTP server. Required by all methods. Password – account access password, required by all methods. SMTP server (SMTPD) The SMTP server processes incoming e–mails. 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.
3.8 Virtual Port Options Introduction The Virtual ports menu helps you configure all virtual port types and virtual ports. In the Virtual ports – All menu you can see all virtual ports regardless of their type. For easier 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 Virtual ports submenus.
there. Basic The Basic tag includes the following parameters: Name according to the physic port – use this option to 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 – use this parameter to assign a virtual port to a specific virtual port type, which represents another hierarchical level for some parameters.
that arrive in the PBX with the Unknown subtype. The other subtypes are retained. Normalising takes place as defined in the Localisation menu. 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 routers This section is used for assigning a selected AutoClip router to a virtual port.
calling number and name matching. Insert calling station name – define whether the calling station name shall be added to the outgoing INVITE message. Own channel count – shows the count of voice channels that can be served by the virtual port. Licences needed In this section, you can check easily whether the Mobility Extension or Call Recording licence is required on the virtual port. If a licence is required yet absent or insufficient, it is in red letters here.
the opposite port that generates the disconnect tone. Setting options Default – provides fall–down to the next level (virtual port type). Yes – enables use. No – disables use. Reset condition– enables 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.
During alerting – when the PROGRESS_IND message comes after the opponent's alert signalling. Overlap Overlap 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 – this parameter enables overlap sending in the port–to–PBX direction. It is primarily used for ISDN virtual ports. Overlap receiving – has not been implemented yet.
Select tariff rate Click on the Set free minutes/SMS button to display a dialogue and select one of the tariff rates as defined in the Accounting and tariff rates menu. In addition, you can assign 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 Used tariff rate optio n. If you do so, you will lose all data saved on free minutes with the given tariff rate via this virtual port.
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.
Remove – removes the selected file from a storage. Remove all – deletes all files from a selected storage. Stack The Stack tag is described in Chapter 3. Virtual Ports depending on the stack type.
4. SIM Here is what you can find in this chapter: 4.
4.1 SIM Cards The Virtual ports – GSM – SIM menu includes a list of all PBX SIM cards. This menu is opened 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 tags. Basic Card serial number – this parameter shows the SIM card identification code detected.
5. Network Here is what you can find in this chapter: 5.1 5.2 5.3 5.
5.1 Network Interface The Network – Network Interfaces menu helps you manage all network interfaces available 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.
5.
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 Synchronisation result field together with information about the next planned synchronisation attempt.
TFTP Root Storage What Is TFTP? The Trivial File Transfer Protocol is a very simple file transfer protocol, containing just basic FTP functions. The TFTP works over the connectionless UDP protocol. A single file 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.
Rename – here rename a file within the root storage. Add file – here add a PC file to the root storage. Save file – here save a root storage file into your PC. The meanings of the table columns are as follows: Name – displays the file name within the root storage. Size – displays the size of the file added. Changed – displays the date and time of the last file update. Attributes – gives additional information on the file.
TCP-IP Communication port The TCP/IP Communication port menu is used for management of ports through which you can access your PBX. Basically, you can only Add or Remove a port in this menu, enabling/disabling the authorisation requirement. It is only port 6992 that requires authorisation after initialisation.
System Services Communication ports Ports 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.
DHCP Server The DHCP server is used exclusively for assigning IP addresses and other parameters to SIP terminals with the specified MAC address in NetStar. The menu consists of two basic sections. A field for setting ranges of subnet IP addresses to be assigned is in the left part of the screen. More parameter settings for the selecetd subnet are available on the right.
Add DNS server(s) – the option is only active for subnets with no DNS defined. Edit value – use this option to edit the existing values. It has the same function as a double click on the selected parameter. Remove option – use this option to remove a parameter from the selected subnet configuration.
Directory Service (LDAP) The LDAP will be launched in one of the following versions of the 2N ® NetStar PBX.
5.
Remote Control (SNMP) What is SNMP? The Simple Network Management Protocol (SNMP) forms a part of the Internet protocol 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.
Rights Name – define the name of the right to be created. This name is displayed in the Users selection. Context – use a text string to identify the SNMP module within the client address. This parameter need not be filled in. Full match context – use this option to enable requirement of full match including context. It is mostly unnecessary. Security model – in this parameter choose either a specific security model (SNMP v1, SNMP v2c, USM = SNMP v3) or the Whatever option.
MIB files Add – use this option to add a selected MIB file to the MIB database. Delete – use this option to delete a selected MIB file. Recompile – use this option to recompile a selected MIB file. File – this column shows the path to the MIB file source. This path is relevant for the Recompile option. Status – this column shows the current status of a MIB file. The options are Compiled, Not compiled and Not found.
Figure: View of Listening Port Setting Section Notification Client address – here define the client's IP address or domain name to which notifications are filtered as mentioned below are sent. Client port – here define the client port to which notifications are sent. Used local port – here 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.
Figure: View of Notification Configuration Menu User/Community – here define the SNMP user that corresponds to the USM for SNMP v3 and Community for the other versions. Filter – here 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.
Event Reporter Find the Event reporter in the Network – Supervision Services – Event Reporter menu. Here you can set the basic rules for sending info SMS on system parts. This function is subject to licence! Event– use this parameter to 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 checking purposes. Set the keepalive sending period in the Planned events in the Global data menu.
Relay action – the option is not accessible until the virtual port is selected in the Used relay block. Only then you can select the required relay action. Switch on – this option helps close the relay of the below–specified port whenever some of the conditions defined in the Event parameters block is met. Switch off – helps open the relay of the below–specified port whenever some of the conditions defined in the Event parameters block is met.
Figure: Pohled na nastavení Reportéra událostí Send as User User – here define the user to be presented as the message author. Send to User User – use this parameter to define the user to which the message shall be sent. Save to User – use this parameter to enable/disable saving messages at the user regardless of the user settings, or respecting the user settings.
SNMP Notification – here specify the SNMP user for notifications. The SNMP block is not available yet. Used relay Port – here specify the port whose relay is to be closed whenever some of the conditions defined in the Event parameters block is met. Event parameters The block is accessible if any of the above mentioned options (PBX restart, Port ready, Port error, etc.) is selected in the Event parameters.
5.4 DB connectors The Network – DB connectors menu is used for setting communication with the External Routing Machine (ERM server). The ERM server partially replaces or complements the 2N® NetStar internal routing mechanisms. Having received a call/SMS routing request, the PBX sends a query to the ERM server. If a matching record is found in the ERM server database table, the ERM server sends back a parameter specifying further call/SMS routing in the PBX.
be stored. Valid record time in cache – set the validity time for a record in the cache. Actual count of records in cache – this field displays the count of records currently stored in the cache. Click on Clear cache to delete all the cache records. Port – set the port number for PBX - ERM server communication. Check IP address – having ticked off this option, complete the Checked IP address parameter to enable communication with the ERM server with this IP address only.
6. Global Data Here is what you can find in this chapter: 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 Progress Tones 6.9 Ring Tones 6.10 AutoClip Parameters 6.11 Storage Manager 6.12 Scheduled tasks 6.13 Status Control parameters 6.14 DTMF 6.15 Causes 6.16 Time Parameters 6.
6.1 Global Parameters Disable all new calls Tick 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.
Example: Figure: View of Global Parameter Configuration Menu Global prefixes Global prefixes are primarily used for Analogue and VoIP virtual ports for easier dialling (CallBacks) even to public networks from the list of missed calls. The prefix is not added where the CLI has the Internal subtype. Assign the respective prefixes to the virtual ports using the Added prefix for external CLIP included in the Basic tag.
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 Added prefix for external CLIP parameter. If the number 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.
6.2 Emergency Calls This menu helps you route emergency calls properly when the PBX is in one of the pre–defined emergency modes. Of course, this setting does not solve the PBX error states. In error states, analog CO lines and an analog 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 – here specify all necessary emergency numbers.
6.3 Localisation Destination selection In 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 – here choose a Localisation (country) from the list and the appropriate country 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.
Normalise CLIP Normalise CLIP – check this option to cut automatically the Calling Party Number (CPN) to the shortest known format according to the CLIP routing Localisation setting. If this option 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 – use this parameter to define the PBX numbering plan length.
6.4 Licences Licence files This 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: 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, Not loaded, Bad CPU, etc.). E1 ports – shows the count of licensed ports for ISDN PRI.
Figure: View of Licence Features Table The most important licences The 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).
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 Packages The list of available language packages finds itself in the Global data – Language packages menu. In addition to default packages, new languages packages can be installed 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 Language.ini file (a common text file). Change the Language ID into the number corresponding to the required localisation.
6.6 Services Service Division The services supported by the NetStar PBX can be divided into three groups – User, Extension and Others. User – this group consists of user call forwarding settings, including forwarding to VoiceMail, VoiceMail progress tone/message recording, playing and deleting, PIN changing, user bundle login and so on.
changing, profile activation, etc.). Description of Selected Services For some services, more parameters should be defined in addition to progress tones or messages. Below are some of them: Private call Here set the destination type in the Destination field to Nothing (the calling user settings are used) or Router (select a router). Call parking Here define the Maximum parking time. The default value is set to 180s. The parked user hears the Music on Hold.
6.7 Conference Rooms For conference room settings refer to the Global data – Conference rooms menu. Use this 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.
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.
Destination for addresses Use 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).
6.8 Progress Tones Introduction The "Progress" 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 tags.
Progress configuration Action– here choose one of the listed commands to define the meaning of the row. Repeat – use this command to 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.
Language pack files Language pack files list This 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 list This 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 in the Progress list tag.
keeping its uploaded file in the PBX. Related progress list This 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 in the Progress list tag. Other sections The Information about progress and Progress configuration sections are common for all tags and their parameters. For the configuration options refer to the Progress list section.
Related progress list This 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 in the Progress list tag. Other sections The Information about progress and Progress configuration sections are common for all tags and their parameters. For the configuration options refer to the Progress list section. Audio inputs Audio inputs This section displays all the audio inputs of the PBX that can be used as progress tone sources.
6.9 Ring Tones To set the ringing tones use the Global Data – Ring Patterns menu. Each ringing tone consists 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.
6.10 AutoClip Parameters AutoClip Routing AutoClip 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.
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.
6.11 Storage Manager Find the storage manager in the Global data – Storage manager menu. This menu helps 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 storages Logical storages represent the basic storing units for all PBX services and functions.
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.
List of files If 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.
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. //192.168.122.110/netstar_data). 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.
6.12 Scheduled tasks This menu helps you schedule your database back-up, PBX restart, UMTS board restart or sending keepalive messages informing of the PBX operation. Click on Add in the context menu to add an event. Select the event type, name and repetition mode in the next window. Database back–up This option helps you schedule your database back-up intervals easily, especially in case of incidental data loss or configuration changes.
Figure: Daily Database Back–Up Configuration Example Do action immediately – make the selected event be executed the moment the button is pressed.
6.13 Status Control parameters The Global data – Status Control parameters menu is used for defining states of the Status Control objects created in the Routing – Routing objects – Status Control objects menu. Figure 1 shows an overview of the parameters to be defined. You can add and remove the rows using the right-hand mouse button context menu Figure: Pohled na menu Status Control parametry The meanings of the columns are as follows: State – define the state of the Status Control object.
6.14 DTMF Refer to the Global parameters – DTMF menu for DTMF profile settings. Select the profile for DTMF detection using this menu.
6.
Cause Objects In this menu, you can create sets of causes to be used for modifying bundle parameters. The menu is divided into two parts. You can add, remove and rename objects on the left and edit the selected objects on the right. The following options are available on the right-hand side: Name – name of the selected cause object. Respond to – use this parameter to specify the object's behaviour with respect to the causes entered.
User Causes In this menu, you can add user causes to be used within other objects if necessary (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.
Cause Mapping Tables In this menu, you can specify changes in selected causes. By assigning causes 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.
Dekadicky 0 Význam 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.
6.
Date and Time In this menu you can find the current date and time of your PBX including the time zone. Figure below shows a basic view of the Date and time menu. The date format is year/month/day and time is displayed in the 24–hour format. Figure: View of Date and Time Setting Menu Push the Set date and time button to display a dialogue box as shown in figure below. Select a calendar item or use the arrows in this window to change the date. Type the day/year values to set the date.
Time Conditions To define the time conditions use the Global Data – Time Parameters – Time Conditions menu. The menu is divided into two parts. A list of available time conditions is on the left and can be created, removed or renamed here via the context menu. On 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.
1. define the time limit, use the From and To checkboxes and Date and Time fields in the upper part of the time limit setting window. 2. The other fields except for Interval negation define a repeat rule for each part of the definition. An interval is valid for a selected time point if: a. Holiday is not checked or the selected time point represents any of the defined holidays; b. No weekday is checked or the selected time point represents a checked weekday; and simultaneously c.
Holidays To define holidays and important days use the Global Data – Time Parameters – Holidays menu. The menu is divided into two parts. A list of available holidays is on the left and the setting options are on the right. To add a holiday, choose the Add opti on in 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 Valid every year item below the calendar.
6.
Administration Settings What is Assistant? The Assistant is a web application for user account supervision. The web server 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 Assistant menu you can find three submenus for an easy Assistant managing and active session monitoring.
User Relations In the Assistant – User Relations submenu you can find the list of all active sessions. There 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. Routing Here is what you can find in this chapter: 7.1 7.2 7.3 7.4 7.5 7.6 7.
7.1 Routers Router The router is a set of rules used for incoming call routing through the PBX. Routers are defined in the Routing – Routers menu, which consists of two windows. The left window displays a list of available routers. The right–hand window helps you configure a selected router. On the left–hand side of the menu you can use the context menu with the following options: Add– use this option to initiate a router adding dialogue. Then enter the name and type of the new router.
Update from file – this option has a similar function as Update, but in this case you can choose a source file of your own. The existing routers are not deleted but completed with missing records. Export to file – use this option to 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 – use this option to make a copy of the currently selected router.
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.
Examples 1. The instruction t1p(5)3,,*6 means that after the other party answers the call, you dial digit 1, wait for five seconds, dial digit 3, wait for two seconds and, finally, dial * and digit 6. 2. The instruction 1,2,,3p(3)456 means that digit 1 is dialled followed by a one–second 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.
By Called Number Subtype This 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.
By Call Type This 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 Port This router is based on routing according to the incoming port (the call comes into the PBX through this port).
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).
7.2 External Routers External routers use the External Routing Machine (ERM server) for call and SMS routing. The ERM server partially replaces or complements the 2N® NetStar internal routing mechanisms. Having received a call/SMS routing request, the PBX sends a query to the ERM server. If a matching record is found in the ERM server database table, the ERM server sends back a parameter specifying further call/SMS routing in the external router.
Figure: Pohled na nastavení externího routeru DB connector – this option helps you select the DB connector for communication with the ERM server. The external router does not work without DB connector assignment. Parameter – this column gives a string of characters to be compared with the string sent back by the ERM server. Alphanumerical characters can be used. Destination type – this column specifies the type of the destination to which the call is to be routed.
7.3 Complex Routers This menu is used for complex routing of incoming calls through the PBX.
7.4 Switch Routers The Switch Router function is available in 2N® NetStar firmware versions 3.1.1. and higher. Switch Router helps you modify call/SMS routing via the 2N® NetStar PBX using a service called Set switch router. Make a call or send an SMS to the service to select a switch router and one of its predefined parameters, thus specifying the call/SMS destination. The Routers – Switch routers menu consists of two windows. The left window displays a list of available switch routers.
Figure: Pohled na nastavení přepínacího routeru Router number – router identification, entered into the service during router selection. Active row – currently active parameter of the switch router. Show comments – tick off to display the Comment column in the table. Enter a comment related to the row without affecting the call or SMS routing process. The comment is displayed automatically next to each switch router row in the 2N ® NetStar Assistant.
more sophisticated call routing rules in dependence on time. Thus, you can route calls with the same input conditions to different destinations at different times. Default destination – if a match is not found in the Parameter column for the value returned by the ERM server, the call is routed to the default destination (item below the routing table): Default type - this parameter specifies the type of the destination to which the call shall be routed. All the PBX routing objects are available (if created).
7.
Bundles Bundle The bundle is a routing object that enables to route an incoming call to one (or all) of the 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.
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.
Bundle conduct Cause object – use this option to select one of the cause objects as pre–defined in the Global data – Causes – Cause objects menu. These objects represent a set of causes to be responded to by the bundle. When one of the cause objects has been selected, the Respond to busy and Respond to reject options are disabled automatically.
The table consists of two columns with the following meanings: Destination type – in this column 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.
Figure: View of Bundle Configuration Menu – Advanced Service Login to Bundle The Station/User Login to bundle services have been enhanced with the option to specify 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.
Refer to the User Manual for details on the Login to bundle service.
DISA DISA The DISA (Direct Inward System Access) routing object is used for automatic call acceptance 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).
Figure: Příklad konfigurace objektu DISA se strategií Ihned The menu consists of the following parameters: Tone – use this option to choose a suitable progress tone from the list. Add progress tones and messages of your own in the menu in Chapter 6.7, Progress tones, if desired. Destination after DTMF dial – use this option to set a router to be used for call routing by the PBX upon DTMF dialling. DTMF – use this option to set whether the DTMF detector should be allocated for the DISA.
Alerting strategy This strategy represents a new DISA concept. When a call comes to the port, it is immediately routed to the preset Alerting destination and this destination is being alerted till the end of timeout. The timeout is set by the Timeout parameter. The call is not 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.
Id – choose a router of the selected type.
Ring Groups Ring Group The ring group is a routing object that is used for routing an incoming call or SMS message 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 Global Parameters menu, the Unselected as missed item. The ring groups are also used as user groups for taking over calls.
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 – use this column to select an object of the selected type.
Advanced settings Send CLIP – this is a table facilitating incoming call identification. Having passed the table, the call Id is modified as required. The Send as parameter helps you set two identification display modes. Select Display to display the Number/URI as the CLIP on the telephone, but the original CLIP will be stored in the CPN history. Select Force to modify both the CLIP displayed during ringing and the CLIP stored in the CPN history.
Figure: View of Ring Group Configuration Menu – Advanced 176
Ring Tables Ring Table The ring table is a routing object used for sequential routing of incoming calls to multiple 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 Setting To set a ring table use the Routing – Routing objects – Ring tables menu.
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 – this command routes an incoming call to the object defined in the remaining table columns.
Wait – use this command to 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.
Figure: View of Ring Table Configuration Menu – Advanced 180
Modems Modem Connection Modem 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 ISDN modem with protocol X.75.
Figure: Nastavení parametrů připojení pro vzdálený dohled prostřednictvím ISDN modemu Modem Setting Trace send enabled - use this option to 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 - use this option to enable a login dialogue request for modem connection.
Sets Set The set is a routing object that is used for an easy object sequencing. For example, sequencing 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.
has been changed since it arrived in the PBX and there is a True setting somewhere in 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 False parameter is set for the subsequent objects somewhere in the set, the call is routed according to this changed number until an object with the True selection is found. Time condition – use the time conditions to change a set in time.
Audio Inputs and Outputs What is Audio I/O? The Audio I/O ports are routing objects that cooperate with the audio ports of the Audio/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 Ports The Audio/IO/Relay board can be equipped with two or four stereo jack ports with the diameter of 3.
Figure: View of Audio I/O Configuration Menu Example 1 – Broadcast To use the audio port for broadcasting set the selected port onto Output in the Boards 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 Tone parameter.
Binary Inputs and Outputs What Is Binary I/O? The Binary I/O ports are routing objects cooperating with the binary ports on the Audio/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.
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 width [ms] parameter and then re-deactivated. If activated earlier, it is only deactivated at the end of the pulse. Disconnect pulse - the relay is deactivated for the time defined in the Pulse width [ms] parameter and then re-activated. If deactivated earlier, it is only activated at the end of the pulse.
Figure: View of Binary I/O - Switch Configuration Menu Example Switch 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 Action at pick up parameter is set to Connect. The other actions are ignored. To deactivate the switch, route the SMS with a different text through the text router to a different binary object than that used for activation.
Detector Setup Detector status - this parameter displays the current status of the detector (Active, Inactive, Unknown). With the Unknown option, the assigned binary port is probably configured as an output or the port or board is unavailable Tone connected - sets 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 Timeout and Play whole tone parameters.
Figure: View of Binary I/O - Detector Configuration Menu Messages for events Sending events - displays the current state of event sending. If such sending is enabled, the messages can be sent and the Enable button is inactive. If the sending is stopped, the messages are not sent and the Enable button is ready for use. Active detector state - use this option to enable a message about the active state of the detector. Within this section you can define the message text to be sent.
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.
Figure: View of CallBack Configuration Menu for calls SMS CallBack Name – only displays the name of the selected object. CallBack delay [s] – shows the delay between the CallBack requesting SMS reception and CallBack execution. Delay in SMS content – the delay data can be omitted in the SMS if set so here. Alerting destination – set the destination for routing the numbers included in the SMS.
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 Ring detection timeout. In that case, the CallBack function is not activated and the call is automatically routed according to the Destination after timeout.
Status Control Objects A Status Control object is a routing object used for keeping the defined state (information) on the basis of received information. Received information here means the called number or text message. The state of the Status Control object is determined by the called number or received SMS. Use the Routing – Routing objects – Status Control objects menu to create the Status Control objects.
only to the users who are assigned directly to the group (or subgroup) to which the bundle is assigned.
7.6 Identification Tables What Is Identification Table? The identification tables are used for changing the calling extension numbers. To create and modify them use the Routing – Identification tables menu. To view an identification 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 on the left. To configure a selected identification table, use the right–hand window.
Identification Table Setting In 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: Calling party determination, New identification determination, Advanced settings and Default destination. The table rows are arranged according to priorities. To change a row priority use the arrows on the right–hand side of the screen.
Time condition You 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 Time Conditions menu.
7.7 AutoClip Routers AutoClip Router The AutoClip routers are used for automatic routing of incoming calls and SMS messages 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.
Figure: View of Identification Table Basic Settings Strategy – use this option to 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 – choose this option to 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.
limits in the AutoClip parameter set. Last change with – this column defines whether the record was created/changed by a call or message. Scheme – this column shows the CPN scheme for each record. Select Number or 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.
8. Users Here is what you can find in this chapter: 8.1 8.2 8.3 8.4 8.
8.1 Users and Groups User Creation To set users use the Users – Users and Groups menu. In this menu you can also manage groups and extensions. A list of available groups, subgroups, users and extensions is displayed on the left. Figure: PBX User Structure from Groups to Extensions In the context menu you can find the following options: Add user – use this option to add a user to a selected basic group/subgroup. Add extension – use this option to add a new extension to a selected user.
subgroups with users and stations easily. Collapse all – use this option to close the whole structure of groups and subgroups with users and stations easily. Moving records using the mouse, also called drag & drop, has been implemented in this menu for easier 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.
E–mail address – here fill in the user e–mail address to be used for user VoiceMail forwarding. If this field is empty, the user will not be able to use service call forwarding to VoiceMail because the voice messages created will have no target destination. 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.
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 Rename option in the context menu as described above. Number – represents a profile identifier used primarily for Profile activation. If you do not fill in this field, you will not be able to use this service. Bundle – use this option to assign a selected profile to one of the available PBX bundles.
Add – use this option to 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 Add option becomes unavailable until you create another time condition. Remove – use this option to remove table rows. One profile may only be assigned to one time condition within the time condition validity period.
Number – here fill in the user Number or URI according to the Scheme column. 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.
Forwarding CFNA – Call Forwarding at No Answer – use this parameter to 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 subtag in the Properties tag for the respective user. The default value is 30 seconds. CFU – Call Forwarding Unconditional – here set the unconditional forwarding to VoiceMail.
Notification Message – use this parameter to set the text of the SMS informing 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 – use this parameter to enable/disable saving messages at the user regardless of the user settings, or respecting the user settings.
Free minutes/SMS The tag helps you set free minutes and SMS for a selected user. The set count of minutes and SMS shall only be deducted on the ports via which the call goes out of the PBX and that are selected for call billing on port (Basic tag of the given port). All the Default OUTports are such ports by default. Select tariff rate Click on the Set free minutes/SMS button to display a dialogue and select one of the tariff rates as defined in the Accounting and tariff rates menu.
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 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.
Day of account – here 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 0 means Never (Manually) and setting 32 means Every day. Mode– use this option to select the method of transfer of old free minutes into the next accounting period.
8.2 User Rights Logins A list of all users and logins is displayed on the left-hand side of the Users – Users rights menu. The list is divided into sections according to user groups and subgroups. The 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 – use this option to create a login for a selected user. This option is active only if the user has not been assigned any login.
Basic After 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 Basic tag. This view is useful for setting similar rights in the user group. The table of rights is divided into sections with the following meanings. Basic Disable – use this option to disable a login for a period of time without deleting it.
8.3 Extension Types Extension Type Creation This tag gets displayed whenever you click on the Users – Extension Types menu. The extension types are used for easier setting of groups of extension. A list of available extension types is displayed on the left and you can set a selected extension type on the right. On the left, you can use the context menu with the following options: Add – use this option to add a extension type. Delete – use this option to delete a selected extension type.
8.4 Extensions Extension Creation This tag gets displayed when you click on the Users – Extensions menu. A list of available extensions is on the left and settings for a selected extension on the right. On the left, you can also use the context menu with the following options: Add – use this option to add a extension. After clicking on this option you will see a dialogue box as shown in Figure 1. First define the extension name.
string to be searched by the Find function. Basic Settings If you select a extension on the right-hand side of the screen, three tags will get displayed on the right: Basic, Properties and Profiles. The Basic tag contains the following parameters: Object – specifies the object type. Name – shows the name of the selected station. Station type– defines the station type. The following options are available: Normal – a normal internal station. SIP – a SIP station.
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.
Figure: View at Extension Options Extension Properties The Properties tag consists of a lot of subtags, which are described in a separate chapter for convenience. This tag is exceptional because almost all of its parameters follow the hierarchical structure. For the structure and all the parameters refer to Chapter Setting Properties.
Profiles In this tag, define the properties of a 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 on the left. When you select one of these profiles, you will see two new tags – Basic and Properties.
8.
User Phone Directories Having been created, each user is automatically assigned a private phone directory. A list of user phone directories is displayed on the left-hand side of the Phone directories – User phone directories menu. The phone directory has a limited capacity of records. The default value is 10 records per user. This limit can be changed using the Maximum user tel. nums parameter in the Basic tag in the user settings.
Group Phone Directories For 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 pattern and call forwarding columns. For the group phone directory refer to the Phone directories – Group phone directories menu.
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 Generate phone directories from users param eter in the Global Data – Global parameters menu. Every change in the name, number, scheme or subtype is automatically made in the generated phone directory too. For group phone directories refer to the Users – Phone directories – Group phone directories (Generated) menu.
Common Phone Directories To create common phone directories use the Phone directories – Common phone directories menu. You can create an 'unlimited' number of phone directories and assign them to selected groups of users. The context menu on the right–hand side of this menu offers the following options: Add – use this option to add a row to a selected phone directory. Delete – use this option to remove a selected row from a selected phone directory.
SIP Phone Directories Find the SIP phone directories in the Users – Phone directories – SIP phone directories menu. 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.
9. Setting Properties Here is what you can find in this chapter: 9.
9.1 Setting Properties Fall-Down Hierarchy All the Properties tag parameters are used according to a fall-down hierarchy of the PBX. It 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.
Properties Tag The Properties tag is situated in the menus of all routing objects as mentioned above (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 properties button. To cancel a parameter, push the Reset default properties button . The Properties tag consists of fourteen subtags, which are logically divided according to functions.
20s. Queue parameters The queue parameters are only available on the group and user levels. The Station polling timeout is the only parameter on the station level. Queue – use this option to 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.
Incoming hold CLIP – use this parameter to 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 YES is selected, the transferred call will be identified by the CLI of the transferred user (A) instead of that of the user who transferred it (B). The default value of this parameter is NO. Outgoing hold CLIP – use this parameter to display the original calling party number in the case of call transfer.
including these parameters. No port – use this option to 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 by the PBX refer to the Routers chapter. Parameters of unsuccessful sending Repeat at fail – use this option to enable repeating of a failed SMS sending attempt. An attempt may fail due to a GSM network rejection or bad signal quality.
each is used in a different situation. The call forwarding settings in this tag can be changed for a selected group of users in the Forwarding exceptions tag, which has a higher priority. Furthermore, it holds true that if extension A forwards its calls to extension B, then extension B can call to extension A without being forwarded. This function is called Boss–secretary.
Tones Use this tag to define the basic tones of the PBX to be played to the calling user. The menu is divided into three parts. The first part, Dial, helps you set various dial tones, the second part, Alert, helps you set various alert tones and the third part, Congestion, helps you set various congestion tones. To add a row defining which tone would be used for which situation use the context menu. A list of situations (states) related to specific types of tones is displayed in the Type column.
ECONOMY – indicates a 2N® StarPoint key phone – Economy type. ADVANCED – indicates a 2N® StarPoint key phone – Advanced type. ENTRY – indicates a 2N® StarPoint key phone – Entry type. BASIC – indicates a 2N® StarPoint key phone – Basic type. STANDARD – indicates a 2N® StarPoint key phone – Standard type. ISDN – indicates any ISDN terminal. GSM – indicates any GSM terminal. VoIP – indicates any VoIP terminal. Optiset Advance – indicates a 2N® Optiset key phone – Advanced type.
The outgoing calls are not limited in the Do not disturb mode. ESC – push the Escape key to reject incoming calls, return to a superior level or clear a character in an item. FLASH – push the Flash key to hold calls. If a call is on hold, you can dial another user or service number. Re–push the key to switch between two calls (one active and the other on hold).
Parameter setting The Parameters subtag offers the following parameters: Key volume – use this parameter to set the loudness of the key pushed in the handset or HandsFree. The parameter may range from 0 to 15. Ring volume – use this parameter to set the loudness of the ring tone. The parameter may range from 0 to 8. HandsFree volume – use this parameter to set the loudness of the HandsFree. The parameter may range from 0 to 15. Headset volume – use this parameter to set the loudness of the headset.
one position to another while typing a text on a StarPoint key phone. Choose one of the seven levels, starting from 'extremely fast' to 'extremely slow'. 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.
2. disabled for User A while enabled with no timeout for user B. After no answer or call rejection by the external GSM extension, an SMS at no answer is sent containing the text as defined in the bottom row of the configuration of user B. When the SMS at no answer is enabled for user A too, an SMS at no answer is sent containing the text from the upper row of the configuration of user A. Services The Services subtag helps you create individual service settings, thus replacing the global ones.
Delete oldest after reaching limit – enable deleting of oldest record files if necessary. 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 Customer subtag provides parameters for functions that have been implemented for a specific customer and so their meanings will be explained marginally only. This subtag is divided into three sections.
10. Billing and Tariffs Here is what you can find in this chapter: 10.
10.1 Billing and Tariffs This menu describes tariffs offered by network providers. The tariffs are then used for deducting free minutes and SMS messages for virtual ports. In future, the menu should facilitate accounting and least cost routing. Provider Add 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.
Destinations/Time conditions You can add a destination and time condition to each credit in this section. Destination means the target network to be dialled. Context menu options: Add – add a destination. Rename – rename the selected destinaci. Delete – remove the selected destinaci. Tariff setting – Here you can set or change the time condition for the selected destination. Tariff description Context menu options: Add – add a row to the destination describing table.
11. Configuration Examples Here is what you can find in this chapter: 11.1 Other Useful Information 11.2 Mobility Extension Configuration 11.
11.1 Other Useful Information COM port and communication program setting Basic equipment of OS Miscrosoft Windows, Hyperterminal application, is used for connection. The whole setting of this application is shown in the figure below.
Console structure Figure: View of Console Structure for Easier Orientation 248
11.2 Mobility Extension Configuration Mobility Extension Mobility Extension is an extension feature of the 2N® Netstar PBX, which enables external extensions to use also features that are not normally available as well as practically all PBX services. The Mobility Extension feature is connected with the existence of external extensions. Before you start creating external extensions and configuring their routing make sure that you have a valid licence and if so, for how many extensions the licence is valid.
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 Users – Extensions – External menu, where, via the context menu, you select Add station and, having completed the name and number, assigned the user and, if necessary, ticked off the SMS resending option, press OK to confirm station creation.
Figure 2: Dodatečné vytvoření externí stanice uživateli The third and last way is to create an external station over the user. This can be done in the Users – Users & Groups menu, where you, having selected a user, open the context menu and select Add station. Complete the station name and number and, if necessary, enable SMS resending in the dialogue window. Again, refer to Figure 2 for details.
Figure 3: View of external extension "Routing" Tag As shown in Figure 3, incoming calls from an external station are routed to DISA. Specific settings of the DISA direct inward dialling are shown in Figure 4. With this configuration, the incoming call is routed to the Default router and then a 10-second dialling timeout follows, which is then detected by the DTMF detector (its inclusion requires checking the DTMF option in the lower part).
Figure 4: View of Configuration of DISA for Mobility Extension The 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 hierachical 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.
Figure 5: Common Configuration of "ME" Tag with Enabled Transmission of Dialling of External Extension What is necessary:Routing Outgoing Calls with Mobility Extension 1. 2. 3. 4. 5. 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.
Figure 6: Part of Configuration of External Extension with Number Used for Routing The port over which the call will be made determines the setup of routing "Without port" in the "Routing" tag in the extension properties (or, as the case may be, the type of the extension when using a mass configuration by the fall–down structure). Figure 3 shows a suitable solution of routing of an outgoing call of the external extension.
Figure 7: Typical Configuration of Bundle of GSM Ports for Outgoing Routing of External Extension What is necessary:SMS at No Answer 1. an external extension; 2. "SMS at no answer" setting in Properties. These SMSs are used for information on missed calls. Make sure that the SMS at no answer tag in the Properties is set correctly on one of the hierarchical levels to send the SMS successfully. Refer to Figure 8 for a typical configuration. As you can see, the configuration is divided into two parts.
Figure 8: Typical Settings for Sending SMS at No Answer What is necessary:Outgoing SMS to External Extension 1. an external extension; 2. "No Port" Messages routing Outgoing SMS are routed according to the Message routing tag in the Properties on one of the hierarchical levels (mostly Group, User or Station) in 2N® NetStar. This tag, together with the typical settings, is shown in Figure 9. The part of configuration marked as "No port" is used for routing or redirecting of SMS at an external extension.
Figure 9: View of Typical Settings in "Messages routing" Tag for User with External Extensio The condition of a correct function of SMS sending to an external extension is the checked "Resend SMS" option in the configuration. The flow chart for SMS sending to an external extension is included in Annex 3. The procedure of forwarding of SMS received by the user at the external extension is shown in Annex 4.
Appendix Annex 1: Flow chart showing the processes for an incoming call from an external extension 259
Annex 2: Flow chart showing the processes for an outgoing call to an external extension 260
Annex 3: Flow chart showing the processes for sending SMS to an external extension 261
Annex4: Flow chart showing the processes for forwarding SMS to external extension 262
11.3 NetStar Installation Guide Setting IP Address and Time IP address Connect to NetStar with HyperTerminal tool 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 NetStar Once you are connected, press to get the login screen. The default login information is Admin and 2n. If there is a # character and no login request on the screen, type NsCon and push Enter for confirmation.
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 key. Having completed all settings, push twice to get to the default menu.
Time When you are in the initial screen for time configuration, press 1 and 3. You will get into the Time setting menu. For correct configuration, push 3 to set the time zone and then enter the number according to your location. Then you can modify time and date using options 1 and 2. You can also use an NTP server is available on the LAN. After pressing option 4 set the IP address or domain name of the NTP server (one DNS at least has to be set in the IP settings).
and create a new PBX called Test. Push OK to open the IP setting screen. Set any name you want. In our case we use Local IP to mark that the local IP will be used. Fill in the IP address into the IP address field that you set for NetStar in the first step.
After all the steps are finished, you will create the connection to NetStar. To get connected, just 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 Wizard The aim of the configuration wizard is to provide you with an easy basic installation. The ISDN BRI parameters are specified during configuration (click on Next not to use ISDN BRI).
Then the hardware is activated. When the activation is finished, you will get the screen 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 on Next to continue.
time zone settings and purpose of the NetStar. Here choose the GSM GW option.
When asked for SMTP settings, choose Next. The last screen will ask you for Router settings. Here choose the preferred LCR structure. In our case it will be Default routers. Then choose Next.
When you get to the final overview, press Finish to get to the configuration interface. To apply the configuration created by the wizard scrip, save the changes to NetStar using the saving icon.
Interface Configuration PRI ISDN The most important aspect of PRI interface configuring is the configuration of the PRI line between the NetStar and PBX systems. The first information we need is the PRI interface configuratin on 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.
To apply your new configuration, save the changes to NetStar using the saving 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 the NetStar and PBX systems. The first information we need is the BRI interface configuration on the PBX.
Set the Virtual port and Stack tags for this port. When you are in the Stack menu, set the interface mode to NT and mode to PTP. To apply your new configuration, save the changes to NetStar using the saving 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. LCR Creation The final configuration step is to create the LCR rules and configure the interfaces to work properly according to these rules.
1. modules. 2. Create a router responsible for routing calls to the GSM bundle. 3. Assign this router to a virtual port connected to the PBX. Create GSM bundle Go to Routing – Routing objects – Bundles, click on the right mouse button and choose Default to create the default set of bundles. One of them is called GSM and 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 NetStar.
Fill in the router name and keep the Called number type selection. 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 Hardware – Boards and choose a port connected to the PBX. On the bottom side of the configuration tool choose the Virtual port tag, then Properties and finally Routing. In the Routing tag set From port, Type to Router and Id to your router.
Save your new configuration to NetStar using the icon. Now NetStar is properly configured to pass calls from the PBX to GSM or PSTN through a bundle of GSM ports. Inbound calls We need to take two steps for inbound calls. 1. Create an incoming router responsible for routing calls into the connected PBX and assign the router to the virtual port through which NetStar is connected to the PSTN (GSM ports). 2. Set the DISA function for processing incoming calls.
Fill in the router name and keep the Called number type selection. Suppose that the PBX PRI port cannot be re-programmed. In this case we have to send a call request in the same format as the PSTN. In our example the company number is 020123xxx. When DISA passes the digits to our router, we 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).
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. In the Routing tag set From port, Type to DISA and Id to your new DISA.
Save your new configuration to NetStar using the icon. Now NetStar is properly configured to answer incoming calls from GSM, play DISA to them and pass dialled numbers to the PBX.
2N TELEKOMUNIKACE a.s. Modřanská 621, 143 01 Prague 4, Czech Republic Tel.: +420 261 301 500, Fax: +420 261 301 599 E-mail: sales@2n.cz Web: www.2n.cz 1949v3.1.
