Page 1 ® 2N NetStar Communication System Manual NS Admin Version 3.1.0 www.2n.cz...
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Page 3: Table Of Contents
Content 1. Introduction ........5 1.1 About Help .
Page 4 11.3 NetStar Installation Guide ........
Page 5: Introduction
1. Introduction Here is what you can find in this chapter: 1.1 About Help 1.2 About Application 1.3 Connecting to PBX 1.4 Configuration Menu 1.5 PBX Activation...
Page 6: About Help
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.
Page 7: About Application
1.2 About Application About Application ® NSAdmin is a configuration tool that helps configure the NetStar communication system, version 2. The application is designed for an x86 platform using the WINDOWS ® operating system connected with NetStar through a network. It is controlled by a mouse and, secondarily, a keyboard. NSAdmin uses the TCP connection or modem and ® communicates with NetStar basically via port 6992. As a necessary condition for using this configuration tool under the Windows XP OS, a service pack 2 and Framework v.3. have been installed. The program does not work without these components. Main menu of the configuration tool Once ...
Page 8 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.
Page 9: Connecting To Pbx
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. Create PbX – use this option to create a PBX on the same level as the selected object or nested into the existing group. Create connection – use this option to create a connection to the selected PBX. Properties – here set or change the properties of a selected object. A name is only assigned to groups. For details on PBX and connection settings see below. Delete – select this option to delete a group, PBX or connection. Auto login ...
Page 10 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. Autosave database after – here set the interval for automatic database backup. This function may only be used for the off-line mode. Delete autosave item older than – here set the maximum time for keeping old database backups in the storage folder. This function can only be used in the off-line mode.
Page 11 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. Particularly useful for modem connections. *Any of the following modes can be selected: Only new – Only the new trace is sent to the tool existent since the moment of PBX connection.
Page 12 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 ...
Page 13 Figure: View of Connection Course Dialogue If you are unable to connect to your PBX, please check whether: 1. the PBX has been switched on; 2. the PBX has been connected to the network; 3. both sides have the same IP address and port; 4. the used communication port has been opened; 5. the appropriate firmware and configuration tool versions are used; 6. the used communication port is not blocked by your antivirus software.
Page 14: Configuration Menu
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.
Page 15 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. The option is available in the off-line mode only. Help – select this option to start the help in the chosen language. Figure: View of Configuration Tool Main Menu Figure above also includes all configuration menu icons. Icon meanings from the left: Logout PBX ...
Page 16 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). We strongly recommend that you should not change the data in this view if you do not know how! This menu tag is governed by the read & write rights assigned to each login.
Page 17: Pbx Activation
1.5 PBX Activation What you need ® ® To activate and configure NetStar you need the NetStar system , a computer running the supported Windows version, a keyboard and a mouse. The PC ® and 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.
Page 18 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).
Page 19 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. ...
Page 20 The current firmware version is loaded into the PBX after the first start–up or every firmware upgrade and may cause a short delay in the GSM board activation. Step 3: Localisation setting ® Localisation setting is another important step of the NetStar configuration. In this step you can define the parameters shown in figure below and described in detail in the 6.3 Localisation chapter. In addition, you can add a language package of your own including texts and progress tones. Two language packages – Czech and English – are...
Page 21 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.
Page 22 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. The settings are identical for the other modes except for step ...
Page 23 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. Cornet extensions are used for the StarPoint key (system) phones connected to the Cornet virtual ports. You can define the first extension number and count of extensions for each group (every other extension has a number increased by one). The extensions are then assigned to ports according to their types (if possible).
Page 24 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 8: SIP domain setting This step helps you define a specific SIP domain. If this option is not selected, the CPU IP address is used as the domain. Step 9: SMTP setting Within this step you can define the SMTP server to be used by the PBX. Port 25 and the CPU Ethernet interface are set automatically for the SMTP. No security is used by default.
Page 25 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. Step 11: Data saving Changes are not automatically stored into the PBX during the Wizard process. To save the changes, use the ...
Page 26: Hardware
2. Hardware Here is what you can find in this chapter: 2.1 Basic 2.2 Boards 2.3 Synchronisation 2.4 Board and Port List...
Page 27: Basic
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 and save the changes . ...
Page 28 Profiles Extenders Extender channels Trunk positions Main case case – digital Main case – analog Detectors Players Table: Benefits and Disadvantages of Hardware Profiles...
Page 29: Boards
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. Add a board that has been detected by the PBX (using the Detected option) or a board...
Page 30 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.
Page 31 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. Click on Print to display the preview and on the button in the left-hand upper corner to print out the configuration. The buttons in the right-hand upper corner display the extender configuration previews.
Page 32 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, takes up the value of 0 and 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 ...
Page 33: Synchronisation
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. The other field contains a list of virtual ports that have been ...
Page 34: Board And Port List
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. Virtual port – shows the complete name of the carrier or resource assigned to a physical port. Stack – shows the general carrier / protocol stack (DSS1, ASL, CO, etc.). Extension – shows the list of extensions assigned to a physical port carrier. User – shows the users of extensions assigned to a physical port. State – shows the current port state. Description – provides associated information. The context menu under the right button offers the following two options: Export to CSV – use this option to export the whole table into a *.CSV file. You ® can use this export for stocktaking purposes and/or for contacting the 2N TELEKOMUNIKACE Technical Support if necessary. Move to port – use this option to move quickly to the configuration of the virtual port on the selected row.
Page 35: Virtual Ports
3. Virtual Ports Here is what you can find in this chapter: 3.1 BRI and PRI 3.2 Cornet 3.3 ASL 3.4 CO 3.5 GSM 3.6 SIP 3.7 SMTP 3.8 Virtual Port Options...
Page 36: Bri And Pri
3.1 BRI and PRI 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.
Page 37 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 ...
Page 38 MSN numbers . Assign a extension to these terminals using the Extensions tag. The terminal shall then identify itself as the selected extension within the PBX. ® NetStar is able to process (i.e. resend to an interface other than ISDN) only G.711 A-law encoded calls on the ISDN interface. Calls encoded by G.711 µ-law can only be resent between the ISDN interfaces in the PBX. Digital interface diagnostic Line state – the parameter cannot be set. It only shows the state of the first interface layer.
Page 39 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. Refer to the Boards section for the meaning of the virtual port. PRI virtual ports are assigned to physical ISDN ports for the Primary Rate Interface. For the hardware configuration of the PRI virtual ports refer to the ...
Page 40 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 ...
Page 41 Hardware – Boards menu and a red text in the upper stack status field. This option may not be combined with the Disconnect L2 if no call option. Terminals – not applied for PRI ports. ® NetStar is able to process (i.e. resend to an interface other than ISDN) only G.711 A-law encoded calls on the ISDN interface. Calls encoded by G.711 µ-law can only be resent between the ISDN interfaces in the PBX. Digital interface diagnostic Line state – the parameter cannot be set. It only shows the state of the first interface layer.
Page 42 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.
Page 43: Cornet
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 Virtual port status The upper menu field displays information on the stack type and its current status including information on the L1 and L2 states, increased bit error rate or nonsynchronous L1.
Page 44 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. The value is updated every 6 seconds and represents a weighted average per minute. Supported terminals ...
Page 45: Asl
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. With an ASL virtual port you can see the following statuses: null config on_hook off_hook error_stop error_start_req Figure: View of ASL Virtual Port Hardware Configuration Line parameters Impedance ...
Page 46 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 ...
Page 47 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. Basic Stack status This field displays information on the stack and its current status. With a CO virtual port you can see the following statuses: null config...
Page 48 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. Dial tone – Congestion ...
Page 49 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.
Page 50 ACIM AC termination [3:0] 0000 600 Ohm 0001 900 Ohm 270 Ohm + (750 Ohm || 150 nF) and 275 Ohm + (780 Ohm || 0010 150 nF) 220 Ohm + (820 Ohm || 120 nF) and 220 Ohm + (820 Ohm || 0011 115 nF) 0100 370 Ohm + (620 Ohm || 310 nF) 0101 320 Ohm + (1050 Ohm || 230 nF) 0110 370 Ohm + (820 Ohm || 110 nF) 0111 275 Ohm + (780 Ohm || 150 nF) 1000 120 Ohm + (820 Ohm || 110 nF) 1001 350 Ohm + (1000 Ohm || 210 nF) 1010 0 Ohm + (900 Ohm || 30 nF) 1011 600 Ohm + 2.16 µF 1100 900 Ohm + 1 µF 1101 900 Ohm + 2.16 µF 1110 600 Ohm + 1 µF 1111 Global complex impedance...
Page 51: Gsm
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: Only GSM Prefer GSM Only UMTS Prefer UMTS Roaming enabled – use this option to enable roaming for a GSM virtual port. Manual network selection – if not checked, the SIM card tries to log into the preferred network automatically. If checked, enter the correct ...
Page 52 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 –...
Page 53 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. Restricted – SIM CLI is always presented regardless of the SIM card or calling user settings.
Page 54 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. To log in successfully, you either enter the PIN or disable PIN requesting by the SIM card.
Page 55 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.
Page 56 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 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. State – shows information on the command processing.
Page 57: Sip
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. REALM_CONFLICT – the Realm collides with another port's Realm/Alias. STUNNING – the public IP address is being obtained from the STUN server. STUN_TIMEOUT – the STUN server is unavailable. EXPIRED – the public IP address validity has expired. SIP_REGISTERING – the gateway registration is in progress. REG_TIMEOUT – the REGISTRAR server is unavailable. REG_NOT_AUTH – the registration has not been authorised. REG_REJECTED – the registration has been rejected with an error. Common parameters Port – here define the local gateway port to communicate with the other party. Realm (Domain) – define the domain over which the gateway communicates. The ...
Page 58 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 ...
Page 59 If checked off, a link becomes available to the Advanced settings. The recommended setting is – Transfer Error correction , – Redundancy and No compression FAX detection – set whether 2N® NetStar shall detect FAX (send the re-INVITE message with T.38 in SDP) for Incoming and/or outgoing fax messages, Always or Never. NetStar versions 2.6.0 and higher also support DTMF sending through SIP Info. This function requires no setting.
Page 60 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. CREDS_OUT_ERROR – the authorisation client is unavailable. REALM_CONFLICT – the Realm collides with another port's Realm/Alias. STUNNING – the public IP address is being obtained from the STUN server. STUN_TIMEOUT – the STUN server is unavailable. EXPIRED – the public IP address validity has expired. SIP_REGISTERING – the gateway registration is in progress. REG_TIMEOUT – the REGISTRAR server is unavailable. REG_NOT_AUTH – the registration has not been authorised. REG_REJECTED – the registration has been rejected with an error.
Page 61 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. Via/Contact – here define the contents of the Via and Contact headers. The following options are available: IP address – fill in the PBX IP address.
Page 62 No compression FAX detection – set whether 2N® NetStar shall detect FAX (send the re-INVITE message with T.38 in SDP) for Incoming and/or outgoing fax messages, Always or Never. NetStar versions 2.6.0 and higher also support DTMF sending through SIP Info. This function requires no setting. Terminals This section is used for terminal management. If no terminal has been created, the VoIP phone cannot register to the SIP proxy. A registered phone is indicated by displaying the IP and MAC address for the connected terminal. Multiple phones may be...
Page 63 Be sure to set the terminal type and MAC address correctly to make ® use of the SIP Provisioning function for the StarPoint IP T2x phones. Advanced settings Figure: View of Advanced VoIP Parameter Setting Dialogue...
Page 64 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. ...
Page 65 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.
Page 66 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.
Page 67: Smtp
SMS messages, i.e. using such objects as text routers and the SMS routing tag. SMTP clients The NetStar PBX provides more than one Ethernet interface. For communication with the SMTP server, however, the PBX always uses the CPU Ethernet interface. In the Virtual ports – SMTP menu you can create SMTP clients to log into the SMTP servers and send e–mail messages.
Page 68 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. Queue length – set the count of e–mail messages to be queued and subsequently processed by the server (routed to final destinations in the PBX or resent to another interface). If you set 1, the server will not receive an e–mail processing request until it completes the preceding one.
Page 69: Virtual Port Options
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. By default, the following colours are assigned to virtual ...
Page 70 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. Enable call without extension – use this parameter to enable/disable answering of incoming calls without the CLI. This parameter is enabled by default. For example, it can be used where a terminal is connected to a certain physical port and no extension has been assigned to the virtual port.
Page 71 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. Assign the AutoClip routers for calls and messages separately but you can use one and the same AutoClip router. For details on AutoClip routers refer to Chapter 7.7 AutoClip Routers Calls – here you can assign an AutoClip router for saving records on outgoing calls. To make the function work, assign the AutoClip parameters to the calling user in the Routing – Users & groups menu on the user or use group level. To assign the ...
Page 72 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. If a licence is valid, the blue text Valid licence is displayed. Properties The Properties tag consists of a number of subtags, which are described in a separate chapter. This tag is exceptional because almost all of its parameters obey the hierarchical ...
Page 73 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. Connect resets condition – an incoming Connect message resets the tone–generating ...
Page 74 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. The selection is inactive. Overlap dialling – has not been implemented yet. The selection is inactive. First digit timeout [ms] – this parameter sets the first digit dialling timeout starting at the moment of the microtelephone pick–up. When it expires, the user cannot ...
Page 75 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. To cancel the virtual port tariff rate, push the Cancel free minutes/SMS button.
Page 76 month for the given virtual port. This count is credited to the given virtual port at the beginning of the accounting period. If the free minute count changes within a month, the port credit is not increased until the beginning of the next accounting period unless provided otherwise in the setting dialogue. Free minutes for this month – the column shows the current count of free minutes to be used in this month. The value includes free minutes transferred from the previous accouting period if any. Spent minutes – displays the current count of minutes spent in the accounting period. Free SMS for month – the column includes the count of free SMS messages per month for the given virtual port. This count is credited to the given virtual port at the beginning of the accounting period. If the free SMS count changes within a month, the port credit is not increased until the beginning of the next accounting period unless provided otherwise in the setting dialogue.
Page 77 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.
Page 78: Sim
4. SIM Here is what you can find in this chapter: 4.1 SIM Cards...
Page 79: Sim Cards
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. ...
Page 80: Network
5. Network Here is what you can find in this chapter: 5.1 Network Interface 5.2 Service Settings 5.3 Supervision Services 5.4 DB connectors...
Page 81: Network Interface
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. Having been opened, the Network – Network interfaces menu displays a list of Ethernet interfaces of the PBX on the left and selected interface settings on the right. With the CPU interface, the options are as follows: Get IP address from DHCP server – use this option to enable obtaining IP settings from the DHCP server automatically. In this case the following sections are inactive.
Page 82: Service Settings
5.2 Service Settings Here is what you can find in this section: Time Synchronisation (NTP) TFTP Root Storage TCP-IP Communication port System Services DHCP Server Directory Service (LDAP)
Page 83 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. Figure: View of Time Synchronisation Menu...
Page 84 telephone format from the selected source and sends it. 3. NetStar receives a downloading request for the <MAC address>.cfg ® configuration file and generates it for each StarPoint IP T2x phone based on the MAC address. 4. NetStar receives a downloading request for another file, searches the TFTP storage for the file and sends the file if successfully found. Configuration The context menu provides the following options: Refresh – use this option to refresh the root storage for updated view. Delete – here remove a file from the root storage.
Page 85 Enter the following string into one of the fields: tftp://PBX_IP_address/tftpPhoneBook.xml. Save the data. Go to the Users – ® Phone directories – SIP phone directories menu in NetStar and select the phone directory source (group/user). Now push the directory access button on your phone to download the directory from the PBX.
Page 86 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 or Remove a port in this menu, enabling/disabling the authorisation requirement. It is only port 6992 that requires authorisation after initialisation. Figure: View of TCP/IP Communication port Menu Configuration If the PBX is accessed through a password-protected port, the Database tag for direct configuration is not displayed for the user or administrator by default. To display it, assign the Read and Write rights to the user or administrator using the Users –...
Page 87 Changing password for root New password: <Enter> Retype password: <Enter> The password will be changed regardless of the original password. If you forget your password, you can change it any time in the same way. All you have to know is your console login name. Use the "Admin" login for the console and password "2n" (by default). If you, despite recommendations, use the TELNET and SSH protocols for login to the NetStar PBX, any software warranty provided by the manufacturer shall be null and void. The system access is logged and intended for servicing purposes only. Menu System To open or close a port using the configuration tool, go to the Network – Service setting – System Services menu and use the Enabled ...
Page 88 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. Subnet The context menu provides the following options: Add new subnet – use this option to display a dialogue to define the required subnet parameters, see below. Change range – use this option to edit the existing subnet range. It has the same function as a double click on the selected subnet.
Page 89 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.
Page 90 Directory Service (LDAP) ® The LDAP will be launched in one of the following versions of the 2N NetStar PBX.
Page 91: Supervision Services
5.3 Supervision Services Here is what you can find in this section: Remote Control (SNMP) Event Reporter...
Page 92 Default option is available, which helps introduce the default SNMP setting, including creation of the public user, right line Unrestricted and filters Internet and NetStar Traps Authentication – here define the password and way of encryption for authentication. Protocol – use the or methods to secure your password. Password – here enter the user password. Privacy – here define the password and way of encryption for data transmission. Protocol – use the or ...
Page 93 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 ...
Page 94 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 . The MIB file statuses are also indicated by the icons on the line beginnings as shown in Figure 2. Additional information – displays additional information.
Page 95 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. Notification type – here select the type of notification to be used. For SNMP v1, Traps may be selected only, for higher versions info requests are also available. Trap – is an SNMP message sent to the client about an event that should be notified. The message does not require acknowledgement.
Page 96 Figure: View of Notification Configuration Menu User/Community – here define the SNMP user that corresponds to the 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. Choose one of the following options: Authentication and privacy Without authentication and privacy Authentication only Context – use a text string to identify a SNMP module within the client address. This parameter need not be filled in.
Page 97 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.
Page 98 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. Positive pulse – helps activate the relay of the below specified virtual port for a period defined in the Relay pulse width parameter after one of the conditions specified in the Event parameters is met.
Page 99 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. SIP extensions – use this parameter to enable/disable resending messages to user SIP stations regardless of the user settings, or respecting the user settings (According to stations). Email extensions – use this parameter to enable/disable resending messages to user email stations regardless of the user settings, or respecting the user settings (According to stations). Mobility Extensions – use this parameter to enable/disable resending messages to user external stations regardless of the user settings, or respecting the user settings (According to stations).
Page 100 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 . A survey of available objects related to the event is on the left and a list of objects currently monitored by the given Event reporter is on the right. Use the arrows to move the objects from one side to the other.
Page 101: 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 ...
Page 102 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 ...
Page 103: Global Data
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 6.17 Assistant...
Page 104: Global Parameters
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. Switch on regime ME Use this option to switch your PBX into the Mobility Extension mode, which is specifically ...
Page 105 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. Unlike the frequently used identification table, this option is applied close before departure to the selected port, i.e. after pairing with the telephone directory. It would be ...
Page 106 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.
Page 107: Emergency Calls
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.
Page 108: Localisation
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. For example, the Czech Republic has number 420 and Slovakia 421.
Page 109 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.
Page 110: Licences
tag on any of the hierarchical levels, you need more licences. Setting this parameter to on the user level needs as many licences as many extensions the user has. Setting this parameter to on the carrier level needs as many licences are there are extensions logged to the carriers of the selected type. Figure: Example of Three–Licence NetStar Licences This part displays a well–arranged table showing details on a selected licence file. The field consists of several columns with the following meanings: Feature – shows the type of a licensed service, interface or object within the system. Type – defines a licence within its type.
Page 111 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 ...
Page 112 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.
Page 113: Language Packages
Figure: View of Language Package Adding Menu Column meanings: Name – shows the name of the language package. The default packages are named after their respective languages. Storage – defines the path to the package storage within the system data space. Built-in means the /opt/netstar directory and Internal means the /data/netstar directory. Together with the Directory column, this parameter gives the absolute path to the storage.
Page 114: Services
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. Extension – this group consists of extension forwarding settings, extension ringing settings for user calls, extension bundle and carrier logins, private calls, CallBack to a extension, call takeover from a extension and so on. Others – this group consists of all the remaining services – progress tones recording, ...
Page 115 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. After the time limit, the parking place is cleared and the call returns to the extension that parked it before. The parked user hears the alert tone.
Page 116: Conference Rooms
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.
Page 117 Tones Welcome to conference – this tone is played to the user after the user's joining the conference called by the selected conference room (handset pick–up). Notice on entering – this tone is played to the conference participants after joining of the user that was not dialled during the conference calling or got out of the conference and is now trying to rejoin the conference room. Alone in conference – this tone is played to the user that remains alone in the conference room (no other extension is even ringing). Alone with alerted – this tone is played to the user that is the first or only to answer during ringing to the conference room users. As soon as another user answers the phone, the tone is disconnected.
Page 118 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).
Page 119: Progress Tones
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.
Page 120 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. ...
Page 121 Delete record, keep file – use this function to delete a selected record while keeping its uploaded file in the PBX. Backup file to local disk – use this function to download a file of a selected record to your local disk. First select the file to be saved in the NetStar data space and then enter the name and storage on your local disk. Own files sources Within this section you can upload a file of your own assigning it to the created record...
Page 122 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. Tones Tones This section displays all the tones of the PBX that can be used as progress tone sources. The context menu of this section provides the following functions: Add – use this function to add a new tone.
Page 123 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. The context menu of this section provides the following functions: Add – use this function to add a new audio input. Rename – use this function to rename a selected audio input. Delete – use this function to delete a selected audio input. Audio input sources In this section you can assign a virtual port of the Audio/IO/Relay board to a selected Audio input. For each port define the language to be used for the input. Related progress list This section provides a list of all the progress tones that use the above selected audio input. You can use all the context menu functions as available in the Progress list tag.
Page 124: Ring Tones
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. To restore the default settings without removing the tones created by you, use the ...
Page 125: Autoclip Parameters
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. For these cases, the information on outgoing calls and messages is saved into AutoClip routing tables, which help find the originally calling user and route the incoming call or SMS to this user. For more details on AutoClip routing refer to Chapter ...
Page 126 receives an incoming call (Active message). Action after record call/message use: None – no action is done after use and the record may be reused for the next matching call(s)/message(s) (until its validity has expired). Restart timeout – the record validity is restarted after use and the record may be reused for the next matching call(s)/message(s) (until its validity has expired). Delete record – the record is deleted after use. Time [mins] – use this parameter to set the validity period for each record of the AutoClip router. When it is checked, the given record has an unlimited validity.
Page 127: Storage Manager
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. You can add logical storages to the PBX from a pre–defined set but cannot create logic storages of your own. Logical storages themselves have no reserved data space. Hence, you have to map one physical storage at least to each logical storage, such as the internal memory, MMC card or network disk (CIFS – Common Internet File S.).
Page 128 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. ...
Page 129 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. Create directory – use this option to create a directory within the data space of the currently used physical storage.
Page 130 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. Network type – choose either Microsoft Windows or . Used for network connections only. Login – set the login for connection to the shared space on the network disk. Used for network connections only.
Page 131: Scheduled Tasks
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. The database is stored in the physical storage defined in the Storage manager in pre-set intervals. The storage can be a MMC card or a shared directory on a network disk. The database is stored with a timestamp designating the storing date/time and the current firmware version for later...
Page 132 Figure: Daily Database Back–Up Configuration Example Do action immediately – make the selected event be executed the moment the button is pressed.
Page 133: Status Control Parameters
State color – assign one of the available colours to the object. Whenever the Status Control object state changes, its colour will change accordingly in the Operator menu of the 2N® NetStar Assistant. Should one state be assigned multiple colours in the table, the colour of the first state is used for all identical states.
Page 134: Dtmf
6.14 DTMF Refer to the Global parameters – DTMF menu for DTMF profile settings. Select the profile for DTMF detection using this menu.
Page 135: Causes
6.15 Causes Here is what you can find in this section: Cause Objects User Causes Cause Mapping Tables...
Page 136 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. Unspecified – the object shall respond to all causes unspecified in the Cause item.
Page 137 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. Desription of cause – the column defines the user description of the cause, which replaces the Cause ID in other menus.
Page 138 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 ...
Page 139 Dekadicky Význam User Private network serving the local user Public network serving the local user Transit network Public network serving the remote user Private network serving the remote user International network Network beyond interworking point Test – this option relates to column Q.850 loc and is used in the inbound direction (Stack to CP). If it is not checked off, column Q.850 loc need not match and the row is recognised according to Q.850 val. If it is checked off, both the values have to match.
Page 140: Time Parameters
6.16 Time Parameters Here is what you can find in this section: Date and Time Time Conditions Holidays...
Page 141 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. To set time, type the values or use the arrows. Standard 0–23 hour and 0–59 minute/second limitations are applied.
Page 142 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.
Page 143 1. define the time limit, use the From and 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 ...
Page 144 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 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. The holidays are not arranged alphabetically ...
Page 145: Assistant
6.17 Assistant Here is what you can find in this section: Administration Settings User Relations...
Page 146 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. Administration settings The Assistant – Administration Settings menu provides the following basic application settings: Confirm deleting – use this option to enable confirmation of record removing from the call history. If this option is checked, the user is asked for confirmation before removing a record. Default language – use this option to select the application language from a list. Currently, the list includes three languages - Czech, English and Finnish.
Page 147 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.
Page 148: Routing
7. Routing Here is what you can find in this chapter: 7.1 Routers 7.2 External Routers 7.3 Complex Routers 7.4 Switch Routers 7.5 Routing Objects 7.6 Identification Tables 7.7 AutoClip Routers...
Page 149: Routers
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. After creation, the router types are colour distinguished for convenience. Choose any of the following router types: Called number ...
Page 150 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.
Page 151 destination according to the preset rule. "0" – no more digits are awaited. ">0" – the process waits for a given count of digits (characters). "–" – the dash indicates an unknown length of the called number. Dialling should be terminated by adding a or by the timeout expiry. In the case of an 'unknown length' of the called number, the call is routed immediately upon prefix recognition and the following digits are transmitted to the destination according to the rule (generally to another router or to the public network). Otherwise, the call is not routed until the whole number has been dialled (according to the preset prefix and count of expected digits, so the number need not be complete at all). Therefore, remember to sort prefixes from the longest to the shortest ones while using the 'collision routing'. The called number can also be changed in this router type. Having passed through the router, the call can be routed to another router of the same type where, however, it is routed ...
Page 152 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 ...
Page 153 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. Local – represents a private network phone number in the local format. National – represents a public network number in the national format with prefixes. International – represents a public network phone number in the international format with prefixes.
Page 154 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). All the columns have the same meanings as the case is with the By called number subtype except for the first one. The first column defines the port. When a preset port is recognised as the incoming port, the call is routed to the preset destination.
Page 155 The routing rule is valid only during the time condition validity period. Time conditions help you create sophisticated routing schemes according to time. You can route a call to different destinations for the same incoming conditions (except for time). Default destination– if no match is found with any of the preset strings, the SMS message is routed as defined in this option (located below the routing rule table): Default type – in this part set the type of destination to which an incoming SMS message is to be routed. Choose only one of the destinations that can be used for SMS routing.
Page 156: External Routers
External routers use the External Routing Machine (ERM server) for call and SMS ® routing. The ERM server partially replaces or complements the 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, ...
Page 157 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. All the PBX routing objects are available (if created). Moreover, there are three options where the destination is not obvious at first sight: Default ...
Page 158: Complex Routers
7.3 Complex Routers This menu is used for complex routing of incoming calls through the PBX.
Page 159: Switch Routers
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 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.
Page 160 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 Assistant – use this block of parameters to set the switch router with respect to the Assistant user application. Visible in Assistant – enable displaying of the switch router in the application. If you do not tick off this option, the given switch router will not be available within the application. Group – set a group or subgroup of users authorised to work with the switch ...
Page 161 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). Default Id - this parameter helps you select a destination of the selected type. ...
Page 162: Routing Objects
7.5 Routing Objects Here is what you can find in this section: Bundles DISA Ring Groups Ring Tables Modems Sets Audio Inputs and Outputs Binary Inputs and Outputs CallBack Status Control Objects...
Page 163 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. Bundle Setting Bundles can be configured in the Routing – Routing objects – Bundles menu. A list of available bundles is displayed on the left. Add, delete or rename bundles using the context menu. You can also create predefined bundles with the Default option. The parameters ...
Page 164 this object is busy or unavailable, the call is routed to the next row or terminated (as preset). All – an incoming call is routed to all objects at the same time. Basically, the strategy substitutes the ring group function. The main difference, however, is that stations and users can login to a bundle using a service. By credit – this strategy is intended for credit–monitored bundles with virtual ports. An incoming call is routed to the virtual port of the bundle with ...
Page 165 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. Cause object for queue – select this option to route an incoming call with queue to all the bundle destinations that have not returned the cause defined in the selected cause object.
Page 166 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. ...
Page 167 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. Example Suppose you want to log in a station to the third position of bundle 151. Dial the service access number *64 from the station and enter the four–digit user PIN (1111, e.g.) when requested so. Now you will be asked to dial the bundle number. Dial 151 and press * for confirmation. Then dial the required bundle position for your station, i.e. 3, and press ...
Page 168 Refer to the User Manual for details on the Login to bundle service.
Page 169 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).
Page 170 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. The count of DTMF detectors is determined by the available ...
Page 171 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. The Alerting destination is still being alerted. If the DTMF option is checked, the DTMF detector is connected after the timeout and remains ...
Page 172 Id – choose a router of the selected type.
Page 173 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. The users who miss their calls due to absence may use the ...
Page 174 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. Figure: View of Ring Group Configuration Menu – Basic ...
Page 175 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. Use Scheme to select either number or URI and Type to set the number subtype (Unknown, Internal, Local, National or International). Force facility – refers to the called number. It is used in DSS1 messages for communication with Ericsson exchanges for billing purposes. Again, set the Scheme (Number or URI), Subtype (Unknown, Internal, Local, National, International) and Number/URI (specific number or address). Force forwarding – refers to the called number. It is used in DSS1 messages for communication with Nokia exchanges for billing purposes. Again, set the Scheme (Number or URI), Subtype (Unknown, Internal, Local, National, International) and Number/URI (specific number or address). Assistant – use this section to set the selected ring group with respect to the Assistant user application. Visible in Assistant – use this option to display a ring group within the application. If it is not checked, the ring group is not available for use.
Page 176 Figure: View of Ring Group Configuration Menu – Advanced...
Page 177 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. ...
Page 178 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. First select an object type and then an object of the selected type. Choose the extension, user, carrier, set, ring group, bundle, another ring table, AutoClip router and also such objects as DISA, VoiceMail and service. Remember that a call is answered immediately when routed to the DISA (Immediate), VoiceMail and service and it makes no sense to add other objects to the ring group! Route with queue – this command routes an incoming call to the object defined in the remaining table columns. If the object is busy, the incoming call is queued regardless of the ...
Page 179 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. Wait always – use this command to set the timeout for proceeding to the next row of the table. The incoming call is not routed to the next row before the timeout expiry. If 0 is used, the PBX does not wait and immediately proceeds to the next row (such row has no sense).
Page 180 Figure: View of Ring Table Configuration Menu – Advanced...
Page 181 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 . The figure below shows an example of modem configuration for remote access to the PBX.
Page 182 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 ...
Page 183 Sets 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 ...
Page 184 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. You can define a different time condition for each row. The rows are then valid in the time of the preset time condition validity only.
Page 185 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.
Page 186 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. When a call comes to the routing object, the selected message is played ...
Page 187 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. Binary Ports The ...
Page 188 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.
Page 189 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. This binary object, however, is assigned to one and the same binary source. But the Action at pick up parameter is set to Disconnect this time.
Page 190 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. Tone disconnected ...
Page 191 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.
Page 192 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 ...
Page 193 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. SMS format An ...
Page 194 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 . When the calling user hangs up before the timeout expiry, the ...
Page 195 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 ...
Page 196 only to the users who are assigned directly to the group (or subgroup) to which the bundle is assigned. ...
Page 197: Identification Tables
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. The context menu on the left side of the menu consists of the following options: Add – use this option to add an identification table.
Page 198 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.
Page 199 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. Advanced settings You can define advanced parameters for each identification table row – Numbering plan Screening , ...
Page 200: Autoclip Routers
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. For easier comprehension and use, examples are provided at the end of this chapter. AutoClip Router Use To set the AutoClip routers use the Routing – AutoClip Routers menu. The menu is divided into two parts. A list of available AutoClip routers is displayed on the left. Add, delete or rename the AutoClip routers using the context menu. Moreover, there is an ...
Page 201 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. Sequentially – choose this option to alert all the matching users sequentially ...
Page 202 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 Number/URI – this column shows the called party number (CPN). This number is necessary for finding a match with the calling subscriber. Therefore, make sure that the CPN is saved in the appropriate format. Always consider specific network properties and incoming normalising if applicable. Time [mins] – this parameter shows the validity time for each record. Action after call use – this parameter defines whether the record shall be kept valid or deleted after being used by a call. Action after message use – this parameter defines whether the record shall be kept valid or deleted after being used by a message. Record is used – this parameter defines whether the record shall be designated as used when it has passed through alerting, i.e. when the alerting message has been signalled, or when it has passed through active, i.e. when the call has been answered. Virtual port – this column shows the port used for routing of the outgoing call that ...
Page 203: Users
8. Users Here is what you can find in this chapter: 8.1 Users and Groups 8.2 User Rights 8.3 Extension Types 8.4 Extensions 8.5 Phone Directories...
Page 204: Users And Groups
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.
Page 205 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. For user creation, however, a dialogue is displayed for you to define ...
Page 206 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. ...
Page 207 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. Upon activation of a profile to which a bundle has been assigned, the user is automatically added to this bundle. Upon deactivation, the user is automatically removed from this bundle.
Page 208 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 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. However, different time conditions can be assigned to one user profile. To make the user profiles switch according to the preset time condition rules, check the Automatic profile switching option in the Users – Users and Groups menu of the Basic tag. Phone Directories The Phone directories tag is located in the Users –...
Page 209 %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 . The selection ® is intended for displaying messages on Cornets and in NetStar Assistant SIP extensions – use this parameter to enable/disable resending messages to user SIP stations regardless of the user settings, or respecting the user settings (According to stations). Email extensions – use this parameter to enable/disable resending messages to user email stations regardless of the user settings, or respecting the user settings (According to stations). Mobility Extensions – use this parameter to enable/disable resending messages to user external stations regardless of the user settings, or respecting...
Page 210 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. It means that all incoming calls will be forwarded directly to the VoiceMail if this profile is active (unless there is a hierarchical exception). CFEC – Call Forwarding on Error Cause – here set the forwarding to VoiceMail in the case of busy user or another error cause detection (e.g. call rejection). Welcome note Welcome note – use this option to choose a VoiceMail progress tone from a list. Set welcome note – use this option to enable/disable recording of a VoiceMail progress tone via the VoiceMail Record welcome note (*35) service. Messages Maximum record length [s] – use this parameter to set the maximum voice message ...
Page 211 %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 setting s. The selection ® is intended for displaying messages on Cornets and in NetStar Assistant SIP extensions – use this parameter to enable/disable resending messages to user SIP stations regardless of the user settings, or respecting the user settings (According to stations). Email extensions – use this parameter to enable/disable resending messages to user email stations regardless of the user settings, or respecting the user settings (According to stations). Mobility Extensions – use this parameter to enable/disable resending messages to user external stations regardless of the user settings, or respecting...
Page 212 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. In addition, you can assign here a setting to the selected user tariff rate as defined earlier for any other user or virtual port. To change the tariff rate if necessary, use the Used tariff rate opt ion. If you do so, you will lose all data saved on free minutes with the given tariff rate via this user. To cancel the user tariff rate, push the Cancel free minutes/SMS butto...
Page 213 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. Free minutes for this month – the column shows the current count of free minutes ...
Page 214 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 means Never (Manually) and setting ...
Page 215: User Rights
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. You can choose one of the types specified below. Change login – use this option change the login type. The option cannot be used for Admin login. Each login is also assigned a type that defines the respective right assignment level. Choose one of the following options: Vice Admin ...
Page 216 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. Must change password – use this option to set automatic advice of a password change upon access to the Assistant application.
Page 217: Extension Types
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. Rename – use this option to rename a selected extension type.
Page 218: Extensions
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. If you choose an already existing name, the extension will not be created and you will be ...
Page 219 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. It should be assigned to a terminal on the SIP Proxy. E-mail – an e-mail station. Not intended for calling. External – a Mobility Extension station. Scheme – use this option to define the station identification scheme. Choose either a telephone number, or URI. Prefix – here choose one of the prefixes defined in the Global parameters menu. This prefix partly substitutes the number subtype and facilitates CallBacks. Number/URI – define the station identification. Enter a number, e-mail address, ...
Page 220 Others Virtual port – this parameter shows the port to which the extension is currently assigned. The parameter has an informative character only and cannot be changed in this menu. Protocol – this parameter defines the communication protocol to be used by the virtual port to which the extension is currently assigned. The parameter has an informative character only and cannot be changed in this menu. Terminal – this option provides a correct identification of the calling user. It is used only for the extensions that are assigned to the ISDN, SIP or Cornet ports. In other cases, you can connect one terminal only to each physical port and so the terminal identification matches the extension number. You can connect two terminals to the Cornet port – Master and Slave but the PBX can only connect one digital telephone to the physical port and so you are recommended to keep the Master setting. You can connect a bus with up to eight terminals to the ISDN BRI port. Each ...
Page 221 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...
Page 222 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 . Find the following parameters in the Basic tag of the extension profile: Active – use this option to activate an extension within a selected user profile. If it is not checked off, all calls coming to this extension are rejected. The extension can establish outgoing calls. Do not ring at call to user – use this option to enable call routing to an extension within call routing to an extension user when the user profile is active. If it is checked off, only the calls routed directly to this extension alert the extension.
Page 223: Phone Directories
8.5 Phone Directories Here is what you can find in this section: User Phone Directories Group Phone Directories Group Phone Directories (Generated) Common Phone Directories SIP Phone Directories...
Page 224 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. To edit the records use the Users – Users & Groups menu in the ...
Page 225 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 –...
Page 226 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 –...
Page 227 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.
Page 228 telephone format from the selected source and sends it. 2. Having received the tftpPhoneBook.xml downloading request, NetStar ® generates the file in the StarPoint IP T2x telephone format from the selected source and sends it. 3. Having received a downloading request for another file, NetStar searches the TFTP storage and sends the file if available.
Page 229: Setting Properties
9. Setting Properties Here is what you can find in this chapter: 9.1 Setting Properties...
Page 230: Setting Properties
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. Higher levels have higher priorities It implies from the figure above that the parameters set on the extension profile level have the highest priority and the parameters set on the virtual port type level have the lowest priority. If a parameter is set to the Default value on a level, a different setting on a lower level is searched for this parameter. If a parameter is not set on any level (Default is set on all levels), the PBX uses the value preset by the source code.
Page 231 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. Some are used only on certain levels because they have no sense on others. The text below explains all the parameters available in the subtags.
Page 232 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. If the queue is disabled, the incoming call on a busy extension is terminated with the 'User busy' cause. The default value of this parameter is NO (queue disabled).
Page 233 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 is selected, the transferred call will be identified by the CLI of the transferred user ( ) instead of that of the user who transferred it ( ). The default value of this parameter is Outgoing hold CLIP – use this parameter to display the original calling party number in the case of call transfer. It means that, if is selected, you will see the calling party number of the transferred user ( ) instead of that of the user who transferred the call ( ). The default value of this parameter is ...
Page 234 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. ...
Page 235 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 forwards its calls to extension , then extension can call to extension without being forwarded. This function ...
Page 236 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 ...
Page 237 ® OpenStage 15, 2N OpenStage 30 and OpenStage 40 key phones. ® Up to two 38-button extenders (IP key modules) can be connected to the 2N StarPoint IP phones of the types T26 and T28. With the Entry Economy Basic Standard , , , and ...
Page 238 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). STATE– use the State button to set speed dialling for the selected number and monitor the state of the selected virtual port, user or extension at the same. The user state displays all user extensions. The state is indicated by a LED at the button: INTERCOM – push the ...
Page 239 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. The parameter may range from 0 to 15. Display contrast – use this parameter to set the display contrast. The parameter may range from 0 to 7.
Page 240 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. If a match is found, the name is added to the number. The whole tag is available on the group and user levels only. The subtag helps you define the warranted count of call records to be displayed on the key phone and in the NS Assistant. Set the count of Missed Received , ...
Page 241 2. disabled for User while enabled with no timeout for user . 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 When the SMS at no answer is enabled for user too, an SMS at no answer is sent containing the text from the upper row of the configuration of user . Services The Services subtag helps you create individual service settings, thus replacing the global ones. You can modify such parameters as progress tones, timers and routers, or activate the PIN request in the service settings. To disable individual settings push the Remove individual setting button.
Page 242 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. Define the supported method of the CPN sending...
Page 243: Billing And Tariffs
10. Billing and Tariffs Here is what you can find in this chapter: 10.1 Billing and Tariffs...
Page 244: Billing And Tariffs
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. Default – reset the pre–defined provider. Credit list You can enter any number of credits for each provider and describe each credit with a different set of properties. Context menu options: Add – add a credit. Rename – rename the selected credit. Delete – remove the selected credit.
Page 245 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. Delete – remove the selected description table row. Column description: Note – for information only. Minimum charged time [s] – set the minimum call cost. If a call is answered, these seconds are charged to the calling subscriber regardless of the duration of the call. Typically, this value is set to 60s.
Page 246: Configuration Examples
11. Configuration Examples Here is what you can find in this chapter: 11.1 Other Useful Information 11.2 Mobility Extension Configuration 11.3 NetStar Installation Guide...
Page 247: Other Useful Information
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 setting Figure: View of Hyperterminal Application Settings...
Page 248 Console structure Figure: View of Console Structure for Easier Orientation...
Page 249: Mobility Extension Configuration
Mobility Extension ® Mobility Extension is an extension feature of the 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 ...
Page 250 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. Refer to Figure 2 for details.
Page 251 Routing of incoming calls of external extensions is associated with the recognition of ® these calls immediately after the call has arrived at one of the ports of the 2N Netstar PBX. Recognition is based on compliance of CLIP (caller identification) with its subtype. Further routing is then governed by the settings at the recognized external extension ("Routing" tag in the extension properties). This tag is shown in ...
Page 252 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 ...
Page 253 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. To distinguish external station rights, simply create several DISA services and routers and assign a different router to each group with identical rights in the DISA to define the possibilities of the calling subscriber. After the call has been made, it is possible to put it on hold any time. In order to access ...
Page 254 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. an external extension; 2. "No port" routing; 3. "From port" routing; 4. a bundle of ports; 5. permitted transmission for holding a call. Routing of outgoing calls to an external extension mainly depends on the configuration of this extension and on the way it is called. In principle, an external extension can be called in two ways. The first is to call a number, which is then routed to the respective external extension. The second way is to call a user of this extension. In such case, it is necessary to uncheck the option "Do not ring when calling a user" in the "Basic" tag at the extension, as shown in Figure 6...
Page 255 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 shows a suitable solution of routing of an outgoing call of the external extension. Inclusion of a bundle of GSM ports in the routing "Without port" reduces the probability of rejection of a call at the respective external extension when one GSM port is busy at the moment. One of the possible settings of the bundle is shown in Figure 7 . ...
Page 256 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. The first part represents configuration for SMS sent by the PBX to the counterparty when the call initiated by an internal or external...
Page 257 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. Here a specific port is set over which the SMS will be sent to the routing number of the external extension, but it is also possible to set a bundle of ports.
Page 258 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...
Page 259 Appendix Annex 1: Flow chart showing the processes for an incoming call from an external extension ...
Page 260 Annex 2: Flow chart showing the processes for an outgoing call to an external extension...
Page 261 Annex 3: Flow chart showing the processes for sending SMS to an external extension...
Page 262 Annex4: Flow chart showing the processes for forwarding SMS to external extension ...
Page 263: Netstar Installation Guide
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 <ENTER> to get the login screen. The default login information ...
Page 264 Having inserted correct login information, you will get the initial configuration screen. By pressing the proper digits you will get to the configuration menus. Press 1 and 1 for IP configuration. The IP configuration screen gets displayed. Press options 2 to 4 for IP setting. To escape from the menu or cancel the current operation, use the <ESC> key. Having completed all settings, push <ESC> twice to get to the default menu.
Page 265 IP address or domain name of the NTP server (one DNS at least has to be set in the IP settings). Connection of Configuration Tool to NetStar Start the NetStar configuration tool. In case no connection to NetStar has been created, create a new one. Create a new group for the customer by choosing the Folder icon and the On the same level option and name the new group. In our case it is called ...
Page 266 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.
Page 267 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.
Page 268 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).
Page 269 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. After the hardware is activated, the wizard will guide you through the basic gateway configuration settings like localisation, where you have to choose the country where NetStar will be installed,...
Page 270 and purpose of the NetStar. Here choose the GSM GW option.
Page 271 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...
Page 272 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.
Page 273 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. For correct interconnection, NetStar has to be configured as NT. To do so you have to: Set PRI card jumpers – to do so switch NetStar into the service mode. When the light goes off on the PRI card, remove the card from the rack and check the...
Page 274 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. In case you are not sure about the PBX BRI port configuration, contact a person responsible for the PBX maintenance without delay. In our case, the PBX was connected to the PSTN and so the BRI port is configured as TE and Point-to-Point. For correct interconnection, NetStar has to be configured as NT and also PTP. To do so you have to:...
Page 275 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. Our task is to enable all outgoing calls to be passed to GSM and all incoming calls to play the DISA welcome note or be passed to the PBX IVR. Outbound calls We need to take three steps for outbound calls. 1. Create a GSM bundle responsible for a correct and well-balanced use of all GSM...
Page 276 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. Configure the bundle – set the allocation strategy to Cyclic Create router Go to Routing – Routers , click on the right mouse button and choose ...
Page 277 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 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 ...
Page 278 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...
Page 279 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). In the latter case, we have to add 020123 to make the PBX receive the number in same way as ...
Page 280 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.
Page 281 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.
Page 282 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 1949v3.1.0...

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