User Manual NANO 2. Version 2.5

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1 Version 2.0

2 User Manual NANO 2 Version 2.5

3 Table of Contents 1. Introduction Typical setup of NanoPBX Getting Started With the NanoPBX Installation Accessing the GUI (Graphical User Interface) Setting up Features System status Configure hardware Trunks Analog Trunks VOIP Trunks Adding a New VoIP Account Details Editing / Deleting an Existing VoIP Account: Outgoing Calling Rule Dial plans Users (User Extensions) Create New User: Modify/Delete selected users Music on Hold (MOH) Call Queues Configuring a Queue Agent Login Settings Voice Menus Time Intervals Incoming Calling Rules Voic General settings (Accessing, Retrieving & Managing Voice Mail) Settings for Voic Conferencing Follow Me Follow-me feature for extensions Directory... 27

4 3.16. Call Features Voic Groups Configuring a Voic Group Voic Group Details Voice Menu Prompts Recording Voice Files Uploading Voice Files System Info Backup Option General Preferences Language General Settings General Settings Details Reboot Advanced options Network Settings WAN Configuration LAN Configuration Host Configuration Firmware Upgrade Options Call Detail Records Active Channels Bulk Add Call Record File Editor Asterisk CLI SIP Settings Diagnostic DID Routing Apply Changes Managing & Handling Nano2 Features Attended Transfer Blind Transfer... 52

5 5.3. Conferencing Call Forwarding Call Parking Call Hold and Retrieve BLF (Busy Lamp Field) Support Hard Reset USB Drive NPA Dialing Phones configuration required for NanoPBX Interoperability X-Lite Configurations SNOM IP Phone Configuration Linksys PAP2 ATA Glossary of Terms... 56

6 NANO 2 - Introduction 1. Introduction The NanoPBX is a compact system that puts the rich features of a high-end PBX into the reach of small businesses. Its built-in voic , multi-level auto attendants, remote extensions and sophisticated call handling features help businesses reduce communications costs, while allowing employees to stay connected worldwide. Setting up and configuring the NanoPBX is a breeze with the user-friendly GUI and this document will show you just how easy it is! User Manual v2.0 1

7 NANO 2 - Introduction 1.1. Typical setup of NanoPBX User Manual v2.0 2

8 NANO 2 - Getting Started 2. Getting Started With the NanoPBX 2.1. Installation 1. Plug one end of the RJ45 Ethernet cable into your Router 2. Plug the other end of the RJ45 Ethernet cable into the WAN port of the NanoPBX 3. Plug the Power Adapter included into an available power outlet 4. Plug the other end of the Power Adapter into the DC-IN port of the NanoPBX 5. The NanoPBX will power up, and automatically connect itself to your network via DHCP (which you can later configure in the SETTINGS > Network Settings section) Important Note: 1. Wait until the POWER and all the six PHONE LED's turn orange and remain stable on the Front Panel of your NanoPBX. 2. Use Straight through Ethernet cable to connect between the NanoPBX to Router/Switch/PC 6. Configure your NanoPBX according to the instructions below 2.2. Accessing the GUI (Graphical User Interface) Connect an Analog phone to the any of the FXS ports of the NanoPBX and dial * * to get the WAN IP address of the NanoPBX. Using the IP address obtained, open the Web browser and type: IP Address:8088>. Here is an example of how it should look: Or also you can access the GUI of the NanoPBX by connecting a PC to the LAN port of the NanoPBX. Enable the DHCP option in the Network Settings of the PC, and then enter in the Web Browser Address field. (Where :8088 is the default local LAN IP address of the NanoPBX). User Manual v2.0 3

9 NANO 2 - Getting Started On the login screen, the default username and password is admin and admin respectively. Press the Login button to enter the NanoPBX web panel. To change the password, please refer to the Settings > General section in the navigation. Important Note: Recommended to use Mozilla Firefox / Google chrome / Safari. The nano2 comes with default credentials admin/admin you have to change these as soon as possible to avoid getting hacked. Allo.com cannot be held responsible for unauthorized access to the nano2 PBX. Please make sure you secure your PBX by placing it behind a firewall and changing user names and passwords with frequency. User Manual v2.0 4

10 3. Setting up Features 3.1. System status Navigation to System status, Status of NanoPBX including Memory Status, VoIP Status, Networking Status and Client Status (to see which clients are connected to the system). After you login, you are brought to a System Status screen, which offers information about the NanoPBX, and help files to assist you in learning about all the different features of the system. 1. Under Trunks you will see the Registration Status of the VoIP Account(s) configured. If it displays Registered then it is successfully configured and connected. 2. Under Conference Rooms you will see all Conference Extensions available or unavailable. 3. Under Parking Lot you will see all Parking Extensions available or unavailable. 4. Under Extensions you will see all the users and the extensions connected to the NanoPBX. Also you can sort out the extension by clicking in the extension 5. Under Queues you will see all Queue Extensions available or unavailable. 6. Under System status > System Info > General you will have a summary of your General information, such as Hostname, Server Date & time zone, Uptime. 7. Under System status > System Info > Memory Status you will see total memory resources, including RAM usage, Compact Flash usage (to store voice files and voic ) and Inbox status (for voic inbox). 8. Under System status > System Info > Network Status you will have a summary of your Network information, such as Hostname, WAN IP Address, Subnet Mask, WAN MAC Address and Default Gateway (you may refer to Settings > Network Settings for more info). 9. Under System status > System Info > Disk you will have a summary of Disk usage and Disk free space available on the file system. User Manual v2.0 5

11 3.2. Configure hardware Analog Hardware Setup & Configuration Port Configuration: Port Configuration allows you to manually configure Ports 5 and 6 either as FXS ports or FXO ports, which are configurable ports on the NanoPBX. That means you can configure these ports to be either as FXS ports (to connect Analog Phones) or FXO ports (to connect PSTN line). Important Note: 1. After choosing the port configuration click on Update Settings Button, NanoPBX will reboot for the changes to take effect. Once the system comes up you can use the ports as per your configuration. 2. Please make sure to disconnect incoming PSTN Line connected to the Port Number 5 and 6 before you switch from 4FXS and 4FXO Mode to 6FXS and 2FXO Mode. It may damage your NanoPBX Unit Trunks Navigation: Trunks > Analog Trunks: Here you can configure the Analog Trunks Analog Trunks The Trunks are configured or created by selecting the channels i.e. checking in the check box provided next to channels which are grouped under Groups options. The Trunk Details are as follows: 1. Channels: select the channel which you want to use. 2. Trunk Name: Name of the trunk which user wishes to use. 3. CallerID: Number to be displayed when the calls are routed through trunks. User Manual v2.0 6

12 Advanced options: These options are used to configure the various options which are user friendly such as Busy Detection, Busy count, Busy pattern, Ring Timeout, Answering/Hanging based on polarity switch etc. User while creating the trunks can create without setting the advanced options as these options by default are set. After all the changes are made click on update button followed by the Apply changes button which is on the navigation bar at top right. User can even edit or delete trunks by using the Edit and Delete button VOIP Trunks Navigation: SETTINGS > VOIP Trunks: This is where you setup VoIP Trunk, or manage existing ones. In this page, fill in the Provider Name, Host name, Host Port, Username, Password and Proxy information given to you by your VoIP provider (known as SIP Credentials). Apply any Codec Settings required. You can prioritize your active codec s by using the drop down buttons. After you have entered the details, click the Save button at the bottom Adding a New VoIP Account Details 1. Provider Name: SIP service subscriber name to identify the trunk when listed in incoming and outgoing rule. 2. Host name: IP address provided by VoIP service provider. 3. Host Port: Used to set the Proxy Port numbers. 4. Username: User account information, provided by VoIP service provider (ITSP), usually has the form of digit similar to phone number or actually a phone number. 5. Password: Set a password to register to (SIP) servers provided by the ITSP. 6. Authorization User: Same as username. 7. Caller ID: Caller id given here will be displayed to the other end for outgoing call using this VoIP trunk 8. Proxy IP Address: IP address or Domain name provided by VoIP service provider. 9. From Domain: Domain name provided by VoIP service provider. User Manual v2.0 7

13 10. Insecure: select the very option (recommended) which means includes all options Editing / Deleting an Existing VoIP Account: On the right side of the page, you can see the list of VoIP trunk you have setup. To edit, or delete any of them, simply click the appropriate icon provided to the right of each trunk. Once you click on the edit button of a VoIP trunk then it will display the information of that particular VoIP trunk, here you can change the required details and click on the Save button and then click on the apply changes tab to save the changes made. To ensure successful registration of your VoIP Trunk, you must click the System Status tab on the top navigation menu (see Status section for more info) Important Note: Make sure to click the APPLY CHANGES tab in the top navigation bar, after adding any new VoIP trunk, or editing / deleting Outgoing Calling Rule Navigation: Outgoing Calling Rules > New Calling Rule: This is where you configure Dial Out Rules. Outgoing Calling Rules represent the prefix sequence used to dial when making an outgoing call either through PSTN (Analog) or VoIP. It is prefix used to configure and enable system to judge outgoing call via FXO or VOIP and also used to select a least cost Routing provider. This will allow user to configure add/delete of rule. User Manual v2.0 8

14 There are two ways to make outgoing calls for the registered extension users: VoIP / SIP trunk via ITSP gateway LINE / PSTN trunk via FXO port Outgoing Calling Rules Details Outgoing calls go through Trunk sequences. If it fails, it will go through the next selected sequence, and so on. Please go through the Dial Pattern section to know about how to choose outgoing trunks. Calling Rule Name: Provide proper rule name Dial Pattern: A unique set of digits that will select this trunk. Enter one dial pattern per line. A pattern consists of digits 0-9, digit set [digits], and wildcard characters like "." and "X". Below table explains digit set and wildcard characters. Expression [digits].(dot) Description Match any single digit listed explicitly. E.g., digit set [13579] match odd digits. One may use '-' to indicate a range of digits, e.g. [2-8]. Is a wildcard, match any digit in any length i.e. one or more digits. Usually given in the endof a pattern to include all numbers X Match any single digit from 0 to 9. Is a wildcard, matches any digit i.e. zero or more digits You can even use the Calling rule to route it to local extensions as Destination or Route the calls to Trunks created so as to make the calls successful. There as still many options like striping the number of digits from front, prepend the digits before dialing and filter. There is also another alternative way if the trunk fails to route the call i.e. Failover Trunk an alternative trunk to route calls. When you have finished adding your Outgoing Calling Rules, click the Save button. Here we will discuss about how to choose outgoing trunks in two different ways. The First way is, choose a provider or a trunk based on prefix. This type of rule will allow user to create a prefix for choosing SIP (VoIP) or LINE (PSTN / FXO) trunk to make an outgoing call. For Eg 1: If you would like to strip out the first digit from the dialed number follow the example: Add _8X. in pattern & configure 1 in strip out field to remove the first digit from the dialed number either analog or SIP users which uses the same dial plan Second way is, choose provider or trunk based on actual number dialed, User Manual v2.0 9

15 This type of rule will allow the user to choose suitable provider based on actual number. For e.g. adding _44X. in Dial Pattern and selecting VoIP Provider 2 in trunk sequence, PBX will allow number dialed from 44+ followed by (any digit from 0 to 9) like will route through VoIP Provider 2. If the ITSP / VoIP provider offer cheaper rates for the region where number starts from 44 users can make use of this rule. In the same way, user can create a prefix and select a LINE (PSTN / FXO) trunk to make an outgoing call. Navigation: Outgoing Calling Rules > Restore Default Calling Rule: This is where you configure Default Outgoing Calling Rules. These are the default calling rules where they show us an example of the patterns and which all fields can be used as default Dial plans Navigation: Dialplans: This is where you configure Dialplans for the users. User Manual v2.0 10

16 A Dialplan is a collection of outgoing rules. Dialplan are assigned to users to specify the dialing permissions they have For Example: you might one Dialplan for local calling that permits the users of that DialPlan to dial local numbers, via the local outgoing calling rule. Another user may be permitted to dial long distance numbers, and so would have a DialPlan that includes both the local and log distance outgoing calling rules. You have to create the Dialplan first before you create any user accounts to make your call successful. The Dialplan details are as follows. Dialplan Name: the name user wish to see in that field of Dialplan. Include Outgoing Calling Rules: when the outgoing Calling rules are created it displays here so that to include it in the dialplan. Include local contents: here the user can select the features which he wishes to use. After all the changes, click on the save button. And don t forget to click on apply changes button on top navigation bar immediately after save button Users (User Extensions) Extensions are the core of the NanoPBX. An extension is a number mapped to a person. So basically, every employee that is connected to the NanoPBX should have their own unique extension number, so that he/she can be reached, and be able to place calls. The NanoPBX supports 2 types of Extensions: IP Extensions and Analog Extensions. IP Extensions: IP extension are devices that have only data networking connection such as Ethernet and they communicate with the NanoPBX using IP based protocol for signaling and Voice, User Manual v2.0 11

17 examples are IP Phone, Soft Phone application. The NanoPBX can support up to 100 IP Extensions registration. Analog Extensions: An Analog Extension is used with a regular telephone system which can be connected to an available FXS port on the back of the NanoPBX. The NanoPBX can have up to 6 Analog Extensions. All the features are supported by both the Analog and IP extensions. Extensions can be part of other features, such as: Queues and Voic Groups. Also, the extension can have a voic of its own Create New User: You have to create atleast one Dialplan using Dialplan option before trying to create Analog/IP Extensions Navigation: Users > Create New user: This is where you setup your Analog/IP extensions Analog/IP Extensions: To add a new Analog/IP Extension, First select the Dialplan option by choosing from one of the available Dialplans. Then, fill in the required pieces of information, such as Extension, CallerID Name, and CallerID Number and select if you d like to make that extension available for Voic configuration. You have to select the technology (See to that Check Box next to SIP Technology in not marked) from the drop down box which displays the ports available for creating the analog User Extension. You have to checkin the Check box next to SIP Technology for creating the IP Extensions. You can also configure various Codecs and Other Options available for using users in other features. Once you are done, click the Update button. Details are given below Extension: Unique identifier (i.e. :6000) CallerID Name : Display name of the user for the called user Dialplan : Select one from the dropdown box (appears only if dialplans are created using Dialplans) CallerID Number : Is same as Extension field it s a number which displays to called users Enable Voic for the User: Enable this extension so that it can be visible when adding members to the Voic Group and Voic . The Voic PIN password will be same as the password provided to the Extension while creating an Extension. This will applies to both Analog and IP Extension. If you are trying to access or retrieve Voice Message, use that particular Extension password. Address: Mail address which the user wishes to receive mails (for Voic ) Technology: User Manual v2.0 12

18 a) For analog user Extension: select the ports from the drop down box. It disables the VOIP settings fields as they are reserved only for VOIP extensions b) For IP user Extension : mark in check box next to SIP Codec Preference: select the codecs from the drop down boxes NAT: for IP extensions enable the NAT by marking in the check box Insecure: select very (recommended) its optional Other Options: select them as they makes user extensions support for other features. The Other fields are optional they will be set to their default values when created. Important Note: The Incoming/ Outbound calls with g729 codec won t be recorded User Manual v2.0 13

19 Modify/Delete selected users Navigation: Users: This is where you can edit / delete existing Analog/IP Extensions individually. On the right side of the page, you can see the list of extensions you have setup. To edit, or delete any of them, simply click the appropriate icon provided to the right of each account/extension. Once you click on the Edit button of an Extension then it will display the information of that particular extension. Here you can change the required details and then click on the Update Extension button to save the changes made. You can delete an extension by clicking on the delete button of the extension from the list of extensions displayed. Navigation: Users > Modify Selected users: This is where you can edit / delete many existing Analog/IP Extensions simultaneously. On the left side of the page, you can see checkbox with the list of extensions you have setup. To edit, or delete any of them, simply mark the appropriate check boxes provided to the left of each account/extension. Once you check the boxes click on the Modify selected users tab. Here you can change the required details and then click on the Update button to save the changes made. User Manual v2.0 14

20 You can delete many existing extensions by clicking on the Delete Selected Users button after marking in the check boxes of extensions from the list of extensions displayed. Click ok the popup window to delete the selected users. Important Note: Make sure to click the APPLY CHANGES button in the top navigation bar, after adding / editing / deleting any Extension. The APPLY CHANGES tab appears if some changes are made and not saved Music on Hold (MOH) Navigation: Music on Hold: This is where you can upload new MOH based on the classes. This is a music file which will be played by the NanoPBX when any of the calling party is kept on hold. All the MOH files which are uploaded will be listed in the MOH selection box under selected MOH Class. Chose any of the MOH class form the drop down box and select from the User Manual v2.0 15

21 Sound files listed. click on Apply Changes tab to save the changes made. You can even delete the MOH classes and create them. Important Note: 1. The MOH files has to be in GSM or WAV format and Maximum file size is 4 MB 2. Make sure to click the APPLY CHANGES button in the top navigation bar, after Selecting / uploading the MOH Files. The APPLY CHANGES tab appears if some changes are made and not saved Call Queues Queues used to distributes incoming calls in the order of arrival to the first available extension in the queue. The system answers each call immediately and, if necessary, holds it in a queue until it can be directed to the next available extension. This feature is used to balance the workload among group of extensions. Queues will provide the following functions, Incoming calls being placed in the queue Extensions that answer the call in the queue. Option to choose ring type strategy, to handle the calls in the queue and distribute the calls in the queue. Music played while waiting in the queue. User Manual v2.0 16

22 Configuring a Queue Navigation: Call Queues: This is where you setup your Queues. To create a Queue, simply fill in all the required details including Extension, Queue Name, Strategy, Queue Length, and select the other options. When done, click the Update button. Queue Details Extension: Extension number to reach the Queue directly Queue Name: The name of the Queue Strategy Type: Ring All: Rings all available extensions Round Robin: Takes turns ringing each available extension Least Recent: Rings the extension which was least recently called by this queue Fewest Calls: Rings the extension with fewest complete calls from this queue Random: Rings a random extension Round Robin Memory: Performs a Round Robin remembering where we left off with the last ring pass Music On Hold : selecting the desired music from the available files in dropdown Queue Timeout: Select the required seconds so that the queue will get terminated after the given seconds Wrap-Up Time: After a call is finished, the time it takes an extension to become available again to become available in the queue Queue Max Len: The maximum number of callers waiting in queue for an available extension. Auto fill: The queue will complete as many calls simultaneously to the available agents. Auto Pause: Pauses an agent if they fail ti answer the call. Report Hold time: Reports the holdtime of the agents before the caller is connected to the agents. KeyPress Events: If a caller presses a key while waiting in the queue, this selects which voicemenu should process the key press. List of Available Members (Agents) This is the list of available extensions that could be part of this queue by selecting them using checkboxes. User Manual v2.0 17

23 Agent Login Settings Agent Login Extension: Extensions to be dialed for the agents to login to specific queue. Agent Callback login Extension: Extensions to be dialed for the agents to Login to the queues they are part of Voice Menus Voice Menus can also be called as IVR (Interactive Voice Response) is a pre-recorded interactive operator defined by a sequence of actions that provides a customer with a better call experience. An IVR can be chained with other IVR s creating a multi-level IVR system. Example 1: Welcome to ALLO PVT LTD! If you know your party s extension, please dial it now. Press 1 for Customer Service Press 2 for Billing Press 3 for Shipping User Manual v2.0 18

24 Navigation: Voice Menus: This is where you setup your IVR Note: Before you start to Create Voice Menu you need to have sound files in Voice Menu Prompt (Navigation: Voice Menu Prompts). As per your convenience he/she can record, upload or buy to use it in Voice menu as an announcement. To create a new IVR, enter a Name and Extension. Use the Add New Step option to list the Sequence of Actions (the sequential order of events triggered by an IVR). In Actions if listen to keypress events is listed then according to the announcement user can avail the advantage of the KeyPress events by allowing the Keypress events. When done, click the Save button. There are different types of Actions specified. Description of every actions are displayed when he/she selects them using the Add new Step dropdown option. When clicked Add new step button the action gets listed in the sequential order. They can even be moved up and down to change their sequential priority. If the user wishes to use keypress events then he/she has to enable the Allow Keypress events checkbox and then configure them according to their convenience of routing calls to required destination Time Intervals Navigation: Time Intervals: This is where you can create/edit /delete Time intervals for the scheduling of the incoming calling rules. The time intervals Details are given as: Time Interval name: the name of the time intervals according to users need. You can select either by day of week or by days of the month User Manual v2.0 19

25 You can also select time duration in a day or the whole day. After all the changes are done click on the update button to create the time intervals and click on the apply changes button to confirm the time intervals configured Incoming Calling Rules Navigation: Incoming Calling rule: This is where you can create/edit /delete Incoming calling rules. A Incoming Calling Rule is an rule which route the incoming call to phone number. Incoming Calling Rule is a feature that enables incoming calls to be routed directly to selected stations without attendant assistance. Incoming Calling Rule Details Trunk: select any trunk from the drop down box appears only if trunks are created. Time interval: scheduling of the incoming calling rules. Pattern: A unique set of digits that will route the call. Enter one dial pattern per line. A pattern consists of digits 0-9, digit set [digits], and wildcard characters like "." and "X". Below table explains digit set and wildcard characters. Expression [digits] Description Match any single digit listed explicitly. E.g., digit set [13579] match odd digits. One may use. (dot) Is a wildcard, match any digit in any length i.e. one or more digits. Usually given in the end X Match any single digit from 0 to 9. Is a wildcard, matches any digit i.e. zero or more digits User Manual v2.0 20

26 Destination: The destination where the calls must be placed when the incoming calls are made. Incoming for PSTN User Manual v2.0 21

27 Incoming for VoIP Voic Navigation: Voic This is where you can manage the configurations of the voic General settings (Accessing, Retrieving & Managing Voice Mail) The NanoPBX allows users to manage voic through voice messages in their phones. This section will summarizes how to access, retrieve and manage voic and other settings. The default feature code for accessing Voic can be set by using Extension for checking messages. After dialing this code, you will enter a basic voice menu with the option to listen or forward messages and configure voic options. When prompted, provide the appropriate Voice Mail number and the password, which is same as it was configured in the Extensions i.e. Extension & Password. While you listen to the recorded voice message you can use the following keys for navigation. 1 Read Voice mail Messages 3 Advanced options 1 Reply 3 Hear Message 5 Leave Message 4 Play previous message User Manual v2.0 22

28 5 Repeat current message 6 Play next message 7 Delete current message 8 Forward message to another mailbox 1 Use Voic number 2 Use Voic Directory 9 Save message in a folder 0 Save in new Messages 1 Save in old Messages 2 Save in Work Messages 3 Save in Family Messages 4 Save in Friends Messages * Help; during msg playback: Rewind # Exit; during msg playback: Skip forward 2 Change folders 0 Switch to new Messages 1 Switch to old Messages 2 Switch to Work Messages 3 Switch to Family Messages 4 Switch to Friends Messages 3 Advanced options 5 Send Message 1 Use Voic number 2 Use Voic Directory 0 Mailbox options 1 Record your unavailable message 2 Record your busy message 3 Record your name User Manual v2.0 23

29 4 Record your Temporary Greetings 1 Record your temporary message 2 Erase your temporary message (going back to the standard message) 5 Change your password * Return to the main menu * Help; during message playback: Rewind # Exit; There are still many options which are user friendly they can be detailed as: Direct Voic Dial: to enable direct voic dial by pressing # followed by extension number Max greeting (in seconds): Maximum number of seconds for User s voic greeting. Dial 0 for operator: enable callers to enable the voic application and connect to an operator extension. Message Options: Maximum messages per folder: Maximum number of messages that a user can have in any of their folders. Max message time: Maximum duration of the voic message in seconds. Min message time: Minimum duration of the voic message in seconds. User Manual v2.0 24

30 Settings for Voic This section will summarize how to configure and notify the user if he/she has received a new voic to their address. The caller after leaving the voic , mail will be sent to the user mail address as mentioned while configuring the user extensions. The supportive details for configuration are: Send messages by only: enabling this message will be sent to only. User cannot check it using the phone.(you need to have a SMTP server configured for this functionality) Attach recordings to enabling this voic is sent as an attachment along with the message(you need to have a SMTP server configured for this functionality) You have to fill in the valid address in the field FROM, Subject and Message can be sent as per the user convenience to alert the destined user about the voic Conferencing Navigation: Conferencing: This is where you can create/edit /delete Conference extensions. This is type of conference will allow eight parties in a call simultaneously. If anybody wants to create a conference, first he has to initiate conference with following procedure. 1. Click create New Conference Bridge 2. Provide extension number to enter into the conference 3. Select the option required Working Scenario By dialing the extension number configured in the above step will allow you to enter into Conference Bridge. User Manual v2.0 25

31 Important Note: 1.NanoPBX will initiate the conference call only with G711u-Law codec with 20ms latency. Please make sure you have enabled the same codec in all the conference IP phones. 2. Audio Conference Bridge supports Max 8 users irrespective of the Number of Conference bridge Follow Me Navigation: Follow me > Follow Me Preferences for Users Follow-me feature for extensions. This page is used to Add new follow me number which could be Local Extension or an Outside Number where the call is forwarded to the number added in Destination field after the ring timeout of the user called. The Details are as follows: 1. Status: Enable/Disable Follow-me for this user. 2. MOH class: MOH class that the caller would listen while tracking the user. 3. Dialplan: DialPlan that would be used for dialing the Follow me numbers. By default this would be the same Dialplan as that of the user. 4. Destinations: List of extensions/numbers that would be dialed to reach the user during Follow me 5. Clicking on the Add Follow Me Number will expand the window size with many options. 6. New Follow Me Number: Add new follow me number which could be Local Extension or an Outsie Number. The selected dialplan should have permissions to dial any outside numbers defined. 7. Dial Order: This is the order in which the Follow Me destinations are dialed to reach the user. User Manual v2.0 26

32 Navigation: Follow me > Follow Me Options is where you can configure the Follow Me settings for extensions. This is where the options like playing back the status message, Record the callers name for announcements and playback the unreachable status message. Important Note: If you are Selecting Dial outside Number as Follow me number follow as per the Outgoing Dialing rules. For eg: If you want add as Follow me number and if your Outgoing rule has Dialout Prefix as 8, then you need add as Follow me number in Dial Outside number Directory Navigation: Directory is where you can configure the Directory option for the extensions to search users by their First or last name. Dialing the 'Directory Extension' would present to the caller, a directory of users listed in the sytem telephone directory - from which they can search by First or Last Name. To add or remove a user from the system telephone directory, edit the 'In Directory' field of the user. The Directory Details are as follows: User Manual v2.0 27

33 Directory Extension: extension to dial for accessing the Name Directory. Read the Extension number: Read the extension number to the caller before presenting dialing options. Use first name instead of last name: Allow the caller to enter the first name of a user in the directory instead of using the last name Call Features Navigation: > Call Features is where you can configure the Call feature settings for the extensions. Feature digit time out, maximum time (ms) between digits for feature activation (default is 1000 ms) PBX Feature Prefixes: This lets the user assign the codes for different features present in the NANOPBX. Blind Transfer: This is to assign the code for performing Blind Transfer, after call establishment dial *1 followed by extension number to blind transfer the call Disconnect: This is to assign the code for performing call disconnect Attended Transfer: This is to assign the code for performing Attended transfer, after call establishment dial *3 followed by extension number to transfer the call. Call Parking: This is to assign the code for performing Call Parking Extension and the Range, which will allow you to set number of parking extensions where the call on hold will be parked. Call Pickup: This feature code is to pick up the ringing extension from another extension if the party is not available in the desk. User Manual v2.0 28

34 Call Record: This is to assign the code for performing Call Record. If the NanoPBX user extension wants to the particular active call, then he needs to press Call record feature code and that call conversation get recorded. Application Map: This section of allows you to map DTMF codes to dialplan applications. The caller will be placed on hold until the application has completed execution Voic Groups A Voic Group is a pre-programmed group of voic recipients. All the members of this group will receive the same voic message Configuring a Voic Group Navigation: Voic Group: This is where you setup your Voic Groups To create a Voic Group, select a Group Name, choose the users who will belong to that Group, and press the Save button. You can see all existing Extensions in User Mailboxes, which are checked in checkboxes to include in the group. You can also edit or delete groups using edit and delete button simultaneously. User Manual v2.0 29

35 Voic Group Details Voic s Group Extension: Label: User Mailboxes: The number to be allocated for the Voic Group Name of the Voic Group The list of available extensions that could be part of this Voic Group. The marked extensions in the list of extensions belong to this Voic Group. Important Note: Make sure to click the APPLY CHANGES tab in the top navigation bar, after adding any new Voice mail Group or editing / deleting Voice Menu Prompts Recording Voice Files Navigation: Voice Menu Prompts > Record a new voice menu prompt To record a Voice File from NANOPBX, input the file name for the voicefile, and select the extension you wish you use from the list, then press the Record button, it will ring the selected extension and allow you to record your voice through this extension. After answering the call a prompt will be played indicating you to start recording with beep sound Once you have finished recording, press the # button on your telephone. When done, you can leave it like that, or add Keypress Information (see Keypress section for more info). When you are done, click Record. The file will automatically be saved as a GSM file on your NanoPBX. Repeat the above operation as many times as the number of messages to record. User Manual v2.0 30

36 Important Note: The extension which is used for recording should have G.711 u-law codec. To play the uploaded voice file from NanoPBX Quick Time player is required. Voice file name should be unique should not coincide with IVR / Queue / trunk name. Transcoding is not available in NanoPBX; our recommendation is to record voice files through NanoPBX itself Uploading Voice Files Navigation: Voice Prompt > Uploading Voice Files To upload a Voice File, enter the Name and Description, and click the Browse button to select the pre-recorded file from your computer. The file should be in GSM format. When done, press the Upload button System Info User Manual v2.0 31

37 General Network Disk Usage Memory Usage You will have a summary of your general information, such as Firmware Version,.Uptime, Asterisk Build, Server Date & TimeZone, Hostname. Network Status you will have a summary of your Network information, such as Hostname, WAN IP Address, Subnet Mask, WAN MAC Address and Default Gateway (you may refer to Settings > Network Settings for more info). Disk you will have a summary of Disk usage and Disk free space available on the file system. Memory Status you will see total memory resources, including RAM usage, Compact Flash usage (to store voice files and voic ) and Inbox status (for voic inbox) Backup Navigation: Backup: Here you can take the backup of the configuration files or upload the configuration files. System Settings Details Upload a previous backup file: Allow you to upload a backup file to the NanoPBX, from your local PC Create New Backup: Allows you to create a backup of the existing configurations of the NanoPBX by clicking on the Create new backup List Of Previous Configuration Backups:The backup files are listed down as we take a backup, we can download to the local PC by clicking the button Download from unit, which makes the Download file to display above the Manage Configuration Backups whereby right Click on the Download File link and download using the 'Save Link As..' You can also Restore the previous Config by clicking on the Restore Previous Config. Important Note: While taking backup, the voic and CDR data should not be in the backup file. User Manual v2.0 32

38 3.21. Option General Preferences Navigation: Options > General Preferences: This is where you can configure the General Settings of the NanoPBX The Details of this page are given as : 1. Global Out Bound CID: this is used for all outgoing calls when no other CallerID is defined that has a higher priority. When making outgoing calls the following rules are used to determine which callerid will be used, if they exist. a. The first Caller ID used is a CallerID set for the user making the call defined in the User s tab. b. The second Caller ID is the one that is set in the VOIP Trunks configuration, if applicable. c. The last CallerID used for outgoing calls is the Global CID defined in the Options Tab. 2. Global Outbound CID name: This is the global Caller ID name that is used for all outgoing calls. If this value is defined, all outgoing calls will have a caller ID name set to this value. This would be usually your company name. Leave this value if you want the user Caller ID Name to appear on Out bound calls. 3. Operator Extension: this extension is dialed when a caller presses 0 to exit voice mail. It is also available as a voice Menu option. 4. Ring Timeout: Number of seconds to ring a device before sending to the user s Voic Box. 5. Enable Idle Image display: Enables the display of a graphic on a phone s LCD display when the phone is idle. 6. VoIP Phone Digit Map: This option allows the administrator to define a global digit mapping string compatible with RFC There is no default setting and this option does not sync with the dialplan assigned to an individual user. The following examples should assist in writing an acceptable digit mapping string. [2-9]11 - Where calls beginning with digits 2-9 followed by digits 11 are dialed immediately. 0T - Where calls beginning with digit 0 followed by a pause equal to the "Digit Timeout" option. +011xxx.T - Where calls beginning with the + character, followed by 011 digits and then at least three more digits before any arbitrary number is matched, dialed after Digit Timeout is reached. 0[2-9]xxxxxxxxx - Where calls beginning with 0, followed by any digit from 2-9, followed further by 9 more digits are dialed immediately. +1[2-9]xxxxxxxx - Where calls beginning with the + character, followed by 1, followed by any digit from 2-9, followed by 8 more digits are dialed immediately. [2-9]xxxxxxxxx - Where calls beginning with any digit from 2-9, followed by 9 more digits are dialed immediately. User Manual v2.0 33

39 [2-9]xxxT - Where calls beginning with any digit from 2-9, followed by three more digits are dialed after Digit Timeout is reached. [2-9]11 0T +011xxx.T 0[2-9]xxxxxxxxx +1[2-9]xxxxxxxx [2-9]xxxxxxxxx [2-9]xxxT where each entry is separated by the character. For more information, please refer to RFC VoIP Phone Digit Timeout: The timeout variable is the number of seconds the phone will wait for each segment of a digit map expressed as an integer Language Navigation: Options > Language: This is where you can configure the Languague Settings of the NanoPBX. Language settings allows the user to specify the default language prompt only for voice prompts User Manual v2.0 34

40 General Settings Navigation: Options > General settings: This is where you can configure the General Settings of the NanoPBX General Settings Details Change Admin Password: This field allows you to change the Admin password. This field is case sensitive and the maximum password length is 25 characters. Caller ID/Country Configuration of FXS and FXO Profile: FXS profile: The dropdown boxes here allows he/she to set AC profile, DC profile and Ring profile and Caller ID Setting based on the PSTN line. FXO profile: The dropdown box here allows selecting the country and call progress tone based on the PSTN line. Time Configuration: You can set the date and time of the NanoPBX either through Enabling NTP or by manual entry. If you are enabling NTP, select the time zone according to your country timing and enter the ntp server details, for eg: pool.ntp.org.if you reboot the Nano2PBX Manual entry of Date and Time will not be saved. Date and Time is used to trigger different IVR Schedules. SMTP Configuration: URI or IP address, Domain and SMTP port of the SMTP (Simple Mail Transfer Protocol) server, which will be used by Nanopbx to send Voic and FAX to s. User Manual v2.0 35

41 Reboot Navigation: Options > Reboot: This is where you can configure the General Settings of the NanoPBX The administrator of the NanoPBX can remotely reboot the NanoPBX by pressing the Reboot button at the bottom of the System management. Once done, following screen will be displayed to confirm reboot. The user can re-login to the phone after POWER LED and all the six PHONE LED's turn orange and remain stable on the Front Panel of your NanoPBX Important Note: Nano2PBX will take about 3 minutes time to reboot and during boot up time kindly don t disturb the unit Advanced options Navigation: Options > Advanced Options: This is where you can configure the General Settings of the NanoPBX The administrator of the NanoPBX can enable and make the advanced options display on the left Navigation list by clicking on the show advanced options. Where many of the advanced options can be accessed by the Administrator like Network Settings, Firmware upgrades, CDR and so on. Billing Info Navigation: Billing Info: This is where you can generate billing information. To create a new Report, select the inbound calls, outbound calls, internal calls, External calls. A list with call details will display in the Billing section. Billing info can also be filtered by selecting inbound calls, outbound calls, internal calls and External calls. You can select the number of list to shown in in the listings by selecting the right dropdown box. By clicking on the Previous and Next button you can see the list pages in next and the previous pages. User Manual v2.0 36

42 3.22. Network Settings Navigation: Network Settings: This is where you setup your Networking Configuration WAN Configuration WAN Configuration is the Internet settings of your NanoPBX. DHCP: Select Yes if DHCP server is available in the local network where Nano2PBX is installed and DHCP Server offers IP address to NanoPBX. Select No If DHCP Server is not available and set the Network Configuration manually IP Address: the IP address corresponding to your local network IP segment* Subnet Mask: the Netmask corresponding to your local network segment* Network ID: your local network segment (Ex: )* Broadcast: broadcast address (Ex: )* Gateway: the IP address corresponding to your Gateway/Router* DNS 1: the IP address corresponding to a DNS server* DNS 2: the IP address corresponding to a DNS server* Important Note: DHCP mode isn't recommended. Or trouble may arise when SIP client need to change registration server address caused by revised IP. User Manual v2.0 37

43 LAN Configuration Use this setting in the event that you want to use the NanoPBX as your network router and act as a DHCP server. IP Address: It is a Base IP Address of a LAN Port, which functions as a gateway for its LAN. Default ipaddress is Netmask: The Net mask corresponding to your LAN configuration* DHCP IP Start: the first IP in the lease range DHCP IP End: the last IP in the lease range DHCP Leases: the number of leases allowed Important Note: WAN port IP and LAN port IP Address shouldn't be in the same net segment. And we recommend not to change LAN address Host Configuration Host Configuration is used to manage your NanoPBX s Host Name. Host Name: Used to name the device to identify inside the LAN network. This field is optional but may be required by some Internet Service Providers or system administrators. When you are finished applying Network Settings, click the Update button. User Manual v2.0 38

44 3.23. Firmware Upgrade The Firmware Upgrade page allows you to update the NanoPBX with the latest release available, which can contain key updates, added functionalities and bug fixes. When a new firmware release is available, download it and save to your local PC. Then, browse for the file, and click the Upload button. Now your NanoPBX will display a Progress Screen and will prompt you when your NanoPBX is about to reboot. Let your NanoPBX reboot, and wait for the orange LED s to come back on. You can always find the latest firmware from the web: Important Note: While upgrading the firmware, please make sure that there won t be power or network disturbances & also make sure to take back-up of configuration if any Options Call Detail Records Navigation: Call Detail Records: This is where you can create Call Reports To create a new Report, select the inbound calls, outbound calls, internal calls, and External calls. A list with call details will display in the Call Reports section. By clicking on the delete button at the bottom Entire call details records (CDR) of NanoPBX are cleared CDR can also be filtered by selecting inbound calls, outbound calls, internal calls and External calls. You can select the number of list to shown in the listings by selecting the right dropdown box. User Manual v2.0 39

45 By clicking on the Previous and Next button you can see the list pages in next and the previous pages. You can even download the CDR in CSV format by clicking Download button, which makes the Download file to display top of the GUI page. whereby right Click on the Download File link and download the CDR using the 'Save Link As..' Active Channels Displays current Active Channels on the PBX, with the options to Hangup or Transfer. When calls are in progress since there is always Refreshing Active Channels. The Current Active Channels on the PBX are displayed. User Manual v2.0 40

46 Bulk Add Navigation: Bulk Add > Create range of new users: This is where you can create many user extensions all at a time. This is the page where we can create many bulk user extensions at a time. The details of the page is as follows: Create: select the number of users to be created. Users starting from Extension: select the number range from where the user extension has to start creating. Click on the Create users followed by apply changes button to update in configuration Tip: Use the 'Modify Selected Users' button from the Users page to edit any options for the created users. User Manual v2.0 41

47 Call Record Navigation: Call Record: This is where you can Download/ Play/ Delete the call record files. Call Record Setting: You can select either GSM or Wav format in the call record settings Upload to Server: You can upload the call records to the remote server through ftp and you can schedule the uploading either by time period basis or Call record File size. Once you made the configuration, Save the changes. Important Note: Call recording supported by using only G711 codec. User Manual v2.0 42

48 File Editor Navigation: File Editor: This is where you can edit the configuration files or verify whether the configuration files are updated. In the File Editor filed mention the configuration file you want to view. It displays the contents of the file. Here you can view modify the contents of the file. If the user wishes to create new configuration file it can be done using New file Asterisk CLI Navigation: >ASTERISK CLI: This page gives an easier access to the user to execute the commands of the CLI. An Example of the sip show peers displays all the peers in NanoPBX. User Manual v2.0 43

49 SIP Settings General Navigation: SIP settings > General: This is where you can configure the General sip settings. The Details are given as Context: Default context for incoming calls Realm for digest authentication: Realm for digest authentication.defaults to 'asterisk'. If you set a system name in asterisk.conf, it defaults to that system name. Realms MUST be globally unique according to RFC Set this to your host name or domain name. UDP Port to bind to: SIP standard port is IP address to bind to: binds to all Domain: Comma separated list of domains which Asterisk is responsible for. Allow guest calls: Enable guest calls. Overlap dialing support: Enable dialing support. Allow Transfers: Enable Transfers Enable DNS SRV lookups (on outbound calls): Enable DNS SRV lookups on calls Pedantic: Enable slow, pedantic checking of Call-ID:s, multiline SIP headers and URI-encoded headers. SIP Domain Support details are: 1. From Domain: When making outbound SIP INVITEs to non-peers, use your primary domain 'identity' for From: headers instead of just your IP address. This is to be polite and it may be a mandatory requirement for some destinations which do not have a prior account relationship with your server. 2. Auto Domain: Turn this on to have Asterisk add local host name and local IP to domain list. 3. Allow External Domains: Allow requests for domains not serviced by this server User Manual v2.0 44

50 TOS Navigation: SIP Settings > TOS: This is where you can configure the TOS sip settings. The Details to be filled are given as: TOS for Signalling packets: Sets Type of Service for SIP packets TOS for RTP audio packets: Sets Type of Service for RTP audio packets TOS for RTP video packets: Sets Type of Service for RTP video packets Max Registration/Subscription Time: Maximum duration (in seconds) of incoming registration/subscriptions we allow. Default 3600 seconds. Min Registration/Subscription Time: Minimum duration (in seconds) of registrations/subscriptions. Default 60 seconds Default Incoming/Outgoing Registration Time: Default duration (in seconds) of incoming/outoing registration Min RoundtripTime (T1 Time): Minimum roundtrip time for messages to monitored hosts, Defaults to 100 ms Override Notify MIME Type: Allow overriding of mime type in MWI NOTIFY Time between MWI Checks: Default Time between Mailbox checks for peers Music On Hold Interpret: This option specifies a preference for which music on hold class this channel should listen to when put on hold if the music class has not been set on the channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer channel putting this one on hold did not suggest a music class Music On Hold Suggest: This option specifies which music on hold class to suggest to the peer channel when this channel places the peer on hold. It may be specified globally or on a per-user or per-peer basis. Language: Default language setting for all users/peers Enable Relaxed DTMF: Relax dtmf handling RTP TimeOut: Terminate call if 60 seconds of no RTP activity when we are not on hold User Manual v2.0 45

51 RTP HoldTimeOut: Terminate call if 300 seconds of no RTP activity when we are on hold (must be > rtptimeout) Trust Remote Party ID: If Remote-Party-ID should be trusted Send Remote Party ID:If Remote-Party-ID should be sent Generate In-Band Ringing: If we should generate in-band ringing always use 'never' to never use in-band signalling, even in cases where some buggy devices might not render it. Default: never Server UserAgent: Allows you to change the user agent string Allow Nonlocal Redirect:If checked, allows 302 or REDIR to non-local SIP address Note that promiscredir when redirects are made to the local system will cause loops since Asterisk is incapable of performing a 'hairpin' call Add 'user=phone' to URI: If checked, 'user=phone' is added to uri that contains a valid phone number DTMF Mode: Set default dtmfmode for sending DTMF. Default: rfc2833h Send Compact SIP Headers: send compact sip headers Max Registration/Subscription Time: Maximum duration (in seconds) of incoming registration/subscriptions we allow. Default 3600 seconds. Min Registration/Subscription Time: Minimum duration (in seconds) of registrations/subscriptions. Default 60 seconds Default Incoming/Outgoing Registration Time: Default duration (in seconds) of incoming/outoing registration Min RoundtripTime (T1 Time): Minimum roundtrip time for messages to monitored hosts, Defaults to 100 ms Override Notify MIME Type: Allow overriding of mime type in MWI NOTIFY Time between MWI Checks: Default Time between Mailbox checks for peers User Manual v2.0 46

52 NAT Navigation: SIP settings > NAT: This is where you can configure the NAT sip settings. Enter the NAT Traversal IP address i.e. Public IP Address, to communicate with Public Network when NanoPBX is behind the NAT. This IP address will substitute in all outgoing SIP messages instead of Local IP address. The Details are given as Extern ip: Address that we're going to put in outbound SIP messages if we're behind a NAT Extern Host: >Alternatively you can specify an external host, and Asterisk will perform DNS queries periodically. Not recommended for production environments! Use externip instead. Extern Refresh: How often to refresh externhost if used. You may specify a local network in the field below. Local Network Address: ' / ' : All RFC 1918 addresses are local networks, ' / ' : Also RFC1918, ' /12' : Another RFC1918 with CIDR notation, ' / ' : Zero conf local network. NAT mode: Global NAT settings (Affects all peers and users); yes = Always ignore info and assume NAT; no = Use NAT mode only according to RFC3581; never = Never attempt NAT mode or RFC3581 support; route = Assume NAT, don't send report. Allow RTP Reinvite: Asterisk by default tries to redirect the RTP media stream (audio) to go directly from the caller to the callee. Some devices do not support this (especially if one of them is behind a NAT. User Manual v2.0 47

53 Misc Navigation: SIP settings > Misc: This is where you can configure the Miscellaneous sip settings. The details to be filled are given as: FAX Pass-through : T.38 fax (UDPTL) Pass-through: Enables T.38 fax (UDPTL) passthrough on SIP to SIP calls Fax to attachment : To save a fax attachment to send user. Outbound SIP Registrations: Register: Register as a SIP user agent to a SIP proxy (provider) Register TimeOut: Retry registration calls at every 'x' seconds (default 20) Register Attempts: Number of registration attempts before we give up; 0 = continue forever Video: Max Bitrate (kb/s):maximum bitrate for video calls (default 384 kb/s). Support for SIP Video:Turn on support for SIP video. Generate Manager Events: Generate manager events when sip ua performs events (e.g. hold). Reject NonMatching Invites: When an incoming INVITE or REGISTER is to be rejected, for any reason, always reject with '401 Unauthorized' instead of letting the requester know whether there was a matching user or peer for their request. NonStandard G.726 Support: If the peer negotiates G audio, use AAL2 packing order instead of RFC3551 packing order (this is required for Sipura and Grandstream ATAs, among others). This is contrary to the RFC3551 specification, the peer _should_ be negotiating AAL2-G instead. User Manual v2.0 48

54 Codecs Navigation: SIP settings > Codecs: This is where you can select the codecs. Important Note: VIDEO pass-through supported in H264 and H263, only DTMF will work Diagnostic Navigation: Diagnostic: This is where you can ping and trace route of the nanopbx. If you enter the IP Address in the blank field of ping and clicked go. You can see the ping status of the ip address of the ip address u entered. If You enter the IP Address in the blank field of traceroute and clicked go. You can trace the route which you mentioned in the field. User Manual v2.0 49

55 DID Routing. Navigation: DID Routing: Direct Inward Dial. A specially configured phone line from the telephone company that allows for dialing inside a company directly without having to go through an attendant. A DID line cannot be used for outdial operation since there is no dial tone offered. However, it can be configured so an outside caller can reach an inside extension with a 7-digit number through the phone company's central office. User Manual v2.0 50

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