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User Manual - CEM Solutions Pvt. Ltd.

1. uuuuuuunnanunnanunnanunnanenn 60 10 1 Attended Transfer nenne nnennnennnnnn nenne nnennnennennen nennen nennen 60 102 Bing RRE vers E EA 60 Ee o aE Eege 60 10 4 Gall OWN A CNG an a E EEEa RDE A TE A ERER 61 TOC E Panne ee ee E ee ee 61 10 6 Cal Hold and RENOVE reset ee ee 61 10 7 BLF Busy Lamp Field Gupport nennen 61 10 12 BIO Re een ee 62 10 12 NPA BANG seen nee pe ent 62 Appendix A Glossary of Terms uu0 u0002a00nnan0nnanunnanunnanunnnnunnnnunnnnunnnnennnnn 64 User Manual v2 8 www cem solution net PSTN VolP PBX Introduction 1 Product introduction 1 1 Overview The PSTN VoIP PBX is a compact system that puts the rich features of a high end PBX into the reach of small businesses Its built in voicemail 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 PSTN VoIP PBX is a breeze with the user friendly GUI and this document will show you just how easy it is A typical network diagram shows the function of PSTN VoIP PBX as below Internet Service Provider Line 1 Link 7 CO Office i Pol N Upto 75 IF Phones Pregranwrnabls Programmable Phars 5 Bhora E User Manual v2 8 www cem solution net NANO 2 Getting Started 2 Getting Started With the PSTN VoIP PBX 2 1 Installation Factory
2. Insecure Port Allow peers matching by IP address without matching port number e Very Allow peers matching by IP address without matching port number Also authentication of incoming INVITE messages is not required e No Normal IP based peers matching and authentication of incoming INVITE The default setting is No Advanced Options In Directory Check this option if the user is to be listed in the system telephone directory Is Agent Check this option if this user is a call queue member Incoming Call record Record the incoming calls Note calls with G722 and G729 codecs don t record Outgoing Callrecord Record the outgoing calls Note calls with G722 and G729 codecs don t record Pickup Group Select a Pickup Group where a phone picksup the incoming call if it matches one of the call s call groups Important Note The Incoming Outbound calls with G722 and 9729 codec won t be recorded 4 2 2 Modify Delete selected users Navigation Users This is where you can edit delete existing Analog IP Extensions individually User Manual v2 8 www cem solution net NANO 2 Setting up Features 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 H
3. 44 in Prepend field amp selecting VoIP Trunk Analog Trunk in trunk sequence PBX will allow number dialed from 44 followed by any digit from O to 9 like 449872837532 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 User Manual v2 8 www cem solution net NANO 2 Setting up Features Use failover trunk Failover trunks Check this option to use Failover trunks where a call goes through an alternate route when the primary trunk is busy or down Trunks List Gives the list of trunks configured Strip Allows the user to specify the number of digits that will be stripped from the beginning of the dialed string before the call is placed via the selected trunk Prepend Specify the digits to be prepended before the call is placed via the trunk Those digits will be prepended after the dialing number is stripped Filter This option is used to filter out certain characters The characters listed in the field will be permitted while all others will be filtered out 4 4 2 Restore Default Calling Rule Navigation Outgoing Calling Rules gt 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 an
4. CEM Solutions PSTN VolP PBX User Manual User Manual PSIN VoIP PBX Version 2 8 PSTN VolP PBX Introduction Table of Contents 1 PFOGUCL le it ee Te de E 4 0 1 Overview cece 4 2 Getting Started With the PSTN VoIP PBX u22002220000200000ann0nnnnnunnnnnunnnnn0nn 5 2 1 IMSTANAUON ee ee nee nun ae 5 2 2 Notification LEDs On the Front Panel 6 2 oc QUICK IN Stal AHON W VIZ AN Cs nee ats 7 2 4 Accessing the Main GUI Graphical User Interface cccceecccesesseeeeeaeeeeeesseeeeeesaaees 7 Bu IV SL E TC 9 4 Setting up the Features zu cece stece ee sestteas nennen nennen nenne nee seavewssseaseecuseeesiuceares 10 4 1 Bontigure hardware seen 10 4 2 EXIENSIONE AAO amp GIP E 10 4 231 Ae ANS NOW E 11 4 2 2 Modify Delete selected user 13 e SS Ehe loi 16 4a 1 ANalog Re 16 GE VO T O ee ee EE 17 4 3 3 Adding a New VoIP Account Deals nn nnnnnnnnnnnnnnnnnnnnnnnennnnnnenn 17 4 3 4 Editing Deleting an Existing VoIP Account 18 4 4 2 Restore Default Calling PHuie 21 429 DIA DIAS see ee ee nee 21 AG III et 22 4 7 Voicemail Groupe 23 4 7 1 Configuring a Voicemail Group 23 4 8 le elef Hold MOHI WE 24 RE 25 29 1 CONOUG UCC sen eie ee nen 25 49 2 Agent EE rennen een ee ne ee nee endende ee une 27 SEENEN 28 4 10 Greeting WICSSAQ CS nase ee ee 28 4 10 1 Recording Voice tes 28 4 10 2 Uploading Voice Files AAA 29 User Manual v2 8 www cem solution net PST
5. Configure the maximum duration in seconds of incoming registration and subscription allowed by the PSTN VoIP PBX The default setting is 3600 Configure the minimum duration in seconds of incoming registration and subscription allowed by the PSTN VoIP PBX The default setting is 60 Configure the default duration in seconds of incoming outgoing registration The default setting is 120 NANO 2 Additional Features 5 5 3 NAT Navigation General Setting gt SIP settings gt NAT This is where you can configure the NAT sip settings Email Settings Extern IP Configure a static address and port optional that will be used in outbound SIP messages if the PSTN VoIP PBX is behind NAT Extern Host you can specify an external host name and Asterisk will perform DNS queries periodically based on the External Refresh Interval Extern Refresh Configure the refresh interval for the external host if used The default setting is 10 Local Network Specify a list of network addresses that are considered inside of the Address NAT network Multiple entries are allowed If not configured the external IP address will not be set correctly A sample configuration could be as follows 192 168 0 0 255 255 0 0 NAT Mode This is a global NAT setting that will affects all peers and users The default setting is YES e YES Always ignore info and assume NAT e NO Use NAT mode only according to RFC3581 e NEVER Never attem
6. If Name configured all outbound calls will have the CallerlD Name set to this name If not the extension s CallerID Name will be used Operator Extension Specify the operator extension which will be dialed when users presses 0 to exit voicemail application The operator extension can also be used in IVR option Ring Timeout Configure the number of seconds to ring an extension before the call goes to the user s voicemail box The default setting is 20 User Manual v2 8 n www cem solution net NANO 2 Additional Features VolP Phone Digit Map This option allows the administrator to define a global digit mapping string compatible with RFC 3435 There is no default setting and this option does not sync with the dial plan assigned to an individual user The following examples should assist in writing an acceptable digit mapping string e 2 9 11 Where calls beginning with digits 2 9 followed by digits 11 are dialed immediately e OT Where calls beginning with digit O followed by a pause equal to the Digit Timeout option e 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 e 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 e 1 2 9 xxxxxxxx Where calls beginning with the cha
7. NANO 2 Managing amp Handling Nano2PBX 10 8 Pickup Group If a phone belongs in a Pickup Group that matches one of the call s call groups that phone may pickup the incoming call by calling 8 on his phone 10 9 Pickup extension If you want to pick up a particular Extension in the same Pickup Group then dial 8 followed by the Extension number 10 10 Call Duration It is the maximum timeout for which an outgoing FXO call is established In Feature Settings the Call Duration Timeout is given 10 11 IP Trunking IP Trunking is also know as Trunking Without Registration With IP Trunking it is possible to make Voip Trunk calls from PSTN VoIP PBX without Registration 10 12 Hard Reset A factoryreset button is provided in the back panel of PSTN VoIP PBX This will hard reset the PSTN VoIP PBX Press and hold the red button for 5 6 seconds so that the PSTN VoIP PBX will reset to factory settings Hard reset is for factory default you will lose your configuration 10 13 USB Drive A USB flash drive is a data storage device that includes flash memory with an integrated Universal Serial Bus USB interface USB flash drives are typically removable and rewritable 8 GB USB drive available along with PSTN VoIP PBX v2 for storing MoH Voicemail and Call Recording 10 14 NPA Dialing The North American Numbering Plan NANP is an integrated telephone numbering plan of 24 countries and territories the United States and its
8. e G 722 e H 263 e H 263p Select the codecs from the list by enabling it User Manual v2 8 www cem solution net NANO 2 Status 6 Additional Features 6 1 Conferencing The conference bridge configurations can be accessed under Advance Features gt Conference In this page users could create edit and delete conference bridges PSTN VoIP PBX supports up to 8 simultaneous PSTN or IP participants irrespective of the number of conference bridges Conference Bridge Settings Extension Play Hold Music For First Caller Announce Callers Count on joining the Conference Quiet Mode Working Scenario Configure the conference number for the users to dial into the conference If enabled the PSTN VoIP PBX will play Hold music to the first participant in the conference until another user joins in The default setting is No f enabled the caller will be announced to all conference participants when there the caller joins the conference The default setting is No If enabled if there are users joining or leaving the conference voice prompt or notification tone won t be played The default setting is No By dialing the extension number configured in the above step will allow you to enter into Conference Bridge During the conference call users can manage the conference from web GUI Log in PSTN VoIP PBX web GUI during the conference call the participants in each conference bridge will be listed in
9. provider Register Time Out Configure the register retry timeout in seconds The default setting is 20 Register Attempts Configure the number of registration attempts before the PSTN VoIP PBX gives up The default setting is 0 which means the PSTN VoIP PBX will keep trying until the server side accepts the registration request Video Settings NANO 2 Additional Features Max Bit Rate Configure the maximum bit rate in kb s for video calls The default setting is 384 Support for SIP Video Select to enable video support in SIP calls The default setting is Yes Generate Manager If enabled the PSTN VoIP PBX will generate manager events Events when SIP UA performs events e g Hold The default setting is No Reject Non Matching If enabled when rejecting an incoming INVITE or REGISTER Invites request the PSTN VoIP PBX will always reject with 401 Unauthorized instead of notifying the requester whether there is a matching user or peer for the request This reduces the ability of an attacker to scan for valid SIP lf enabled when the peer negotiates G726 32 audio the PSTN VoIP PBX will use AAL2 packing order instead of RFC3551 packing order AAL2 G726 32 The default setting is No Nonstandard G 726 Support 5 5 5 Codecs Navigation SIP settings gt Codec s The following Audio amp Video codec s are supported in PSTN VoIP PBX e G7 11 u law e G7 11 a law e G 726 e G 729
10. 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 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 PSTN VoIP PBX 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 gt System Info gt General you will have a summary of your General information such as Hostname Server Date amp time zone Uptime 7 Under System status gt System Info gt Memory Status you will see total memory resources including RAM usage Compact Flash usage to store voice files and voicemail and Inbox status for voicemail inbox 8 Under System status gt System Info gt 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 gt Network Settings for more info 9 Under System status gt System Info gt Disk you will have a summary of Disk usage and Disk free space available on the file system ad NanoPBX System Status System Status Please click on a panel to manage related features Setup Conference Rooms d Trunks v 1 1 Status Trunk Type Use
11. 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 Voicemail number 2 Use Voicemail Directories 0 Mailbox options 1 Record your unavailable message 2 Record your busy message 3 Record your name 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 User Manual v2 8 www cem solution net NANO 2 Additional Features There are still many options which are user friendly they can be detailed as Direct Voicemail Dial to enable direct voicemail dial by pressing followed by extension number Max greeting in seconds Maximum number of seconds for User s voicemail greeting Dial 0 for operator enable callers to enable the voicemail 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 voicemail message in seconds Min message time Minimum duration of the voicemail message in seconds CH General VoiceMail Settings General Voicemail Settings Email Settings for VoiceMails General VoiceMail Settings Extension to check me
12. VoIP PBX lf users have other backup files on PC to restore on the PSTN VoIP PBX click on Browse first and select it from local PC to upload on the PSTN VoIP PBX and then click Upload 7 2 Firmware Upgrade The Firmware Upgrade page allows you to update the PSTN VoIP PBX 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 PSTN VoIP PBX will display a Progress Screen and will prompt you when your PSTN VoIP PBX is about to reboot Let your PSTN VoIP PBX reboot and wait for the orange LED s to come back on You can always find the latest firmware from the web hitp www cem solutions net pstn voip pbx html e Important Note While upgrading the firmware please make sure that there won t be power or network disturbances amp also make sure to take back up of configuration if any User Manual v2 8 www cem solution net NANO 2 Status o During the firmware upgrade All the Front LEDs start blinking and LEDs will turn orange after the completion of Firmware upgrade 7 3 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 th
13. and time when the message is left Voicemail Email alert preferences General Voicemail Settings Email Settings for VoiceMails Send messages by e mail W Attach recordings to e mail U Subject hi voicemail hi wvoicemail Message D E Save Cancel User Manual v2 8 www cem solution net NANO 2 Additional Features 5 2 Network Settings Navigation Network Settings This is where you setup your Networking Configuration 5 2 1 WAN Configuration Please refer to the following tables for basic network configuration parameters WAN Configurations IP Address Mode Select DHCP Static IP The default setting is DHCP IP Address Enter the IP address for static IP settings Subnet mask Enter the subnet mask address for static IP settings The default setting is 255 255 255 0 Network Id your local network segment Ex 192 168 0 0 Broadcast Enter the broadcast address Gateway Enter the gateway IP address for static IP settings DNS Primary Enter the DNS server 1 address for static IP settings DNS Secondary Enter the DNS server 2 address for static IP settings Important Note DHCP mode isn t recommended Or trouble may arise when SIP client need to change registration server address caused by revised IP 5 2 2 LAN Configuration Use this setting in the event that you want to use the PSTN VoIP PBX as your network router and act as a DHCP server By d
14. resources including RAM usage Compact Flash usage to store voice files and voicemail and Inbox status for voicemail inbox 9 APPLY Changes Navigation APPLY CHANGES On the right top of the web GUI This is the button which you must press after adding editing deleting such things as Extensions Voice Files IVR s and modifying settings such as General Settings VolP Accounts Network Settings DID Routing Firmware Upgrades and other System Settings User Manual v2 8 www cem solution net NANO 2 Managing amp Handling Nano2PBX 10 Managing amp Handling PSTN VoIP PBX Features 10 1 Attended Transfer This type call transfer occurs when before making the transfer a user first call to the third party to inform that a transferred call is coming their way Follow the steps below to perform an attended or supervised call transfer With an active call in progress press the 9 button This puts the original caller on hold and gives you a dial tone on a second line Dial the party that you wish to transfer to Inform the third party that they are about to receive a call Once transferor hangs up the call the original caller and the party you transferred to are now connected 10 2 Blind Transfer This type of call transfer occurs when the person receives a call and transfers the caller to another person or call without any consultation or announcement from the transferor party Follow the steps be
15. territories Canada Bermuda and 16 of the Caribbean countries For E g If this is your phone number 001 2127773456 then you can configure as NPA in the Dial out Rules as a 001212 XXXXXXX where 001 is a Country code and the 212 as a Area code and X is a number of digits to dial out In this above case you have to dial only 7773456 from any of the PSTN VoIP PBX Extensions Once you dial it will compare the number of digits which is configured in the Dial out Rules configured for NPA User Manual v2 8 www cem solution net NANO 2 Managing amp Handling Nano2PBX User Manual v2 8 www cem solution net NANO 2 Glossary of Terms Appendix A Glossary of Terms ATA Analog Telephone Adapter Used to connect a standard telephone to a high speed modem to facilitate VoIP and or fax calls over the Internet DHCP Short for Dynamic Host Configuration Protocol a protocol for assigning dynamic IP addresses to devices on a network With dynamic addressing a device can have a different IP address every time it connects to the network DHCP also supports a mix of static and dynamic IP addresses DID 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 ext
16. the System Status Page you can kick out the participant by Click on to kick a participant from the conference Important Note 1 PSTN VoIP PBX 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 User Manual v2 8 www cem solution net NANO 2 Status 6 2 Follow Me Navigation Additional Features gt Follow me Where you can enable the Call Forward forthe Users Extensions 6 2 1 Follow me Preference for Users 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 2 3 Status Enable Disable Follow me for this user MOH class MOH class that the caller would listen while tracking the user 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 Destinations List of extensions Numbers that would be dialed to reach the user during Follow me Clicking on the Add Follow Me Number will expand the window size with many options New Follow Me Number Add new follow me number which could be Local Extension or an Outside
17. the format GSM WAV g729 for the IVR prompt file to be recorded User Manual v2 8 www cem solution net NANO 2 Setting up Features e Select the extension to receive the call from the PSTN VoIP PBX to record the Greeting message e Click the Record button A request will be sent to the PSTN VoIP PBX The PSTN VoIP PBX will then call the extension for recording the IVR prompt from the phone e Pick up the call from the extension and start the recording following the voice prompt e The recorded file will be listed in the Greeting Message web page Users could select to re record play or delete the recording 4 10 2 Uploading Voice Files Navigation Greeting Messages gt Upload a custom Voice Menu Prompt If the user has a pre recorded Voice file click on Upload IVR Prompt in Web GUI gt Greeting Messages gt Upload a custom Voice Menu Prompt page to upload the file to the PSTN VoIP PBX The following are required for the IVR prompt file to be successfully uploaded and used by the PSTN VoIP PBX e PCM encoded e 16 bits 8000Hz mono e In GSM WAV g729 format Important Note File Size Should be under 5 MB Convert Sound File Usually the recorded sound file cannot be directly used by the PSTN VoIP PBX Users would need convert the sound file to make it compatible with the PBX system before uploading using audio processing software tools The recommended format for the compatible sound file format is GSM WAV G
18. 729 For example the following online converter provided by Digium Inc can be used to convert the sound file http my digium com en products ivr audio converter User Manual v2 8 www cem solution net NANO 2 Setting up Features Converter Voice Prompt File Choose File No file chosen ty Select Codec Oo KHZ vay 0 KHz Wan GSM signed Linear a 6 729 4 11 IVR 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 Navigation Setup gt IVR User could Setup Edit and Delete an IVR IVR Configuration Parameters Name Configure the name of the IVR Extension Enter the extension number for users to access the IVR Allow Dialing Other If enabled all callers to the IVR can dial other extensions The Extension default setting is No User Manual v2 8 www cem solution net NANO 2 Setting up Features Actions Add new Step Allow Key press Events Edit VoiceMenu voicemenu custom 1 A sequence of actions performed when call enters the IVR Select the IVR steps as per the requirement Select the event for each key pressing for 0 9 Thee event options are Extension e Voicemail e Conference Rooms e Voicemail Group e IVR e Queues e VR e Hang u
19. 88 gt Here is an example of how it should look http 192 168 1 100 8088 NanoPBX Home Welcome to NanoPBX Please login Login to NanoPBX Web Panel Username admin Password NOTE Please use Mozilla Firefox Only User Manual v2 8 www cem solution net NANO 2 Getting Started Or also you can access the GUI of the PSTN VoIP PBX by connecting a PC to the LAN port of the PSTN VoIP PBX Enable the DHCP option in the Network Settings of the PC and then enter http 192 168 113 1 3088 in the Web Browser Address field Where 192 168 113 1 8088 is the default local LAN IP address of the PSTN VoIP PBX On the login screen the default username and password is admin and admin respectively Press the Login button to enter the PSTN VoIP PBX web panel To change the password please refer to the General Settings gt Admin Settings gt PBX Settings section in the navigation User Manual v2 8 www cem solution net NANO 2 System Status 3 system Status Navigation to System status Status of PSTN VoIP PBX 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 PSTN VoIP PBX 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
20. 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 Directory Extension Specify the Extension number 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 6 4 Call Record Navigation Additional Features gt Call Record The PSTN VoIP PBX allows users to record audio during the call Please follow the instructions below to record the call 6 4 1 Call Record Files User Manual v2 8 www cem solution net NANO 2 Status Ch Callrecord Files CallRecord Files CallRecard Settings From User To User Sa TT auto 1386863436 5002 102 way Download Ee Delete 2 auto 1 386862206 5002 102 way Download Play Delete Here the entire Call recordings are listed You can Download Play or Delete the recordings You can play the Call recordings by click on Play and you can select the user extension to listen 6 4 2 Call record Settings Ch Callrecord Files CallRecord File CallRecord Settings CallRecord to Email Di Cl For
21. N VolP PBX Introduction ie Wie NOW E 30 AA2 INCOMING Gallng RUGS scires inserieren 32 e DID ROUINO ME 33 5 ENEE dl e E 34 Oe es e en UE 34 5 1 1 General settings Accessing Retrieving amp Managing Voice Maul 34 Emall Sellings tor ee EE 37 9 2 ING WV OIA SE Te ee ee een 38 92 1 WAN COM JU AU EE 38 922 LAN COMO SON E 38 92 3 HOS COM AON une en ee ans ee 39 5 3 Feature Gettnges 39 9 AONE 11 1 GE 41 5 4 1 General PLeierences E 41 EE ee e 42 EE BR SONGS EE 42 OP 2 210 00 ee 44 Ende EE 44 79 1 GEHEN eege 44 Ee 1 eege 45 o NA EE 46 Eeer 4 EE EE 48 6 e e Diels CIR TE 49 6 1 CGonterencimng nennen nnnnnnnnnnnnsnnnnnnennnnnnnnnnnnn 49 92 F OOW E 50 6 2 1 Follow me Preference for Users ccsssccccecccccscnsseeeeecesessaseeeseeecessssaueeeseeessssauaneeseeeessnaas 50 622 70 OW ME e EE 50 DIDI ECO E EE 51 0A alll RO ON BE 51 54 e ee 51 G42 Gall TOCOM SCUINOS E 52 0 9 I e e WEE 53 Or FOS acess cette ee ee ee 53 0 7 AON OCON EE 55 User Manual v2 8 www cem solution net PSTN VolP PBX Introduction Di Kee E 56 Bl BACK Do EE 56 f2 FMW E DI EE 56 7 3 FO ONO ee en ee a see 57 Ze ASIOHISK Gel ee E ae era ee 57 EE EE HE enge 57 CT E 58 E e UR DEI A eet ee 58 8 2 Active Channels u222u0sssssseennnnnennnnnennnnnnennnnnnnnnnnnnnnnnnennnnnnnnnnnnnnnnne nassen nenne snsnnnesenn 58 eV SL O ET 59 9 APPLY el Le CC 59 10 Managing amp Handling PSTN VoIP PBX Features
22. Number The selected dialplan should have permissions to dial any outside numbers defined Dial Order This is the order in which the Follow Me destinations are dialed to reach the user iO Follow Me FollowMe Preferences for Users Followhle Options Extension Follow Me Follow Order 10 Enabled 13 Disabled Not Configured 111 Disabled Not Configured 112 Disabled Not Configured 113 Disabled Not Configured 4496633 Disabled Not Configured 6 2 2 Follow me Options Navigation Additional Features gt Follow me gt 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 caller s name for announcements and playback the unreachable status message User Manual v2 8 www cem solution net NANO 2 Status 4 Follow Me Followhe Preferences for Users FollowMe Options FollowMe Options Follawhle Ring Timeout 20 C Playback the incoming status message prior to starting the follow me stepis Record the caller s name so it can be announced to the callee on each step and Announce weather to recieve the call or reject it C Playback the unreachable status message if we ve run out of steps to reach the or the callee has elected not to be reachable 6 3 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
23. Reset Button LE Power Reset WAN Memory Card Phone 4 3 2 I Une o il I o a ae ee eee L Console FALL S for troubleshooting Power on Reset Power Button Supply A PSTN PSTN Cloud Computer 1 Unpack your PSTN VOIP and make sure you received everything Each box contains 1 x PSTN VOIP PBX 1 x AC adapter 120V 220V 2 wall plugs EU amp US 1 x Ethernet cable 2 Plug in your PSTN VOIP PBX LAN port to your PC using the network cable provided 3 Plug the power cable to the PSTN VoIP PBX After 10 secs All the 6 LEDs start blinking and LEDs turn to orange after the completion of boot up 4 Check your PC network settings and make sure that you have enabled the DHCP 5 Use following IP address in web browser to start the configuration process 192 168 113 1 8088 6 Factory default login credentials are Username admin Password admin 7 Follow the quick setup wizard steps User Manual v2 8 www cem solution net NANO 2 Getting Started 2 2 Notification LEDs On the Front Panel LED Indicator LED Status FXS ports Phone FAX i Off Hook Port No 1 6 On Hook FXO Telco Lines i Channels idie Port No Channels Busy Warning amp Error Notifications LED Status USB is Not detected LED Blinks Flash Memory is Full LED Blinks Reboot Process After 10 secs All the Front panel LEDS start blinking and LEDs will turn orange after the compl
24. Trunk an alternative trunk to route calls Send this call through trunk Use Trunk Select the trunk for this outbound rule FXO VOIP Trunk Strip Allows the user to specify the number of digits that will be stripped from the beginning of the dialed string before the call is placed via the selected trunk Prepend Specify the digits to be prep ended before the call is placed via the trunk Those digits will be prepended after the dialing number is stripped Filter This option is used to filter out certain characters The characters listed in the field will be permitted while all others will be filtered out 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 amp 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 This type of rule will allow the user to choose suitable provider based on Country code For e g If the user want to Dial 44 9872837532 then adding _X in Dial Pattern and configure 0 in Strip Out field and add
25. a ITSP gateway e LINE PSTN trunk via FXO port User Manual v2 8 www cem solution net NANO 2 Setting up Features New CallingRule x Calling Rule Mame D SE emm Caller IDO g Send ta Local Destination D Destination e K Send this call through trunk lise Trunk Strip D digits from front and Prepend these digits betore dialing using this filter D CI use Failover Trunk D Failover Trunks Trunks List NEBR Strip D digits from front and Frepend these digits D before dialing using this filter D Esme Seanca Calling Rule Name Configure the name of the calling rule e g local long distance and etc Letters digits etc Pattern All patterns are prefixed with the _ X Any Digit from 0 9 Z Any Digit from 1 9 12345 9 Any digit from 1 to 9 N Any Digit from 2 9 Wildcard Match one or more characters I Wildcard Match zero or more characters immediately Caller ID Name Configure the CallerID Name associated with the FXO trunk User Manual v2 8 www cem solution net NANO 2 Setting up Features 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
26. ark a call dial the Blind Transfer prefix code followed by Call Parking feature code i e 8 7000 The call is parked and the caller is held And it will announce the parked extension i e 7005 To retrieve the call the user can go to any phone in the group and dial the parked extension i e 7005 Then call will be retrieved and connects to the retrieved user Use the following steps to park and retrieve the call While you are on a call press the transfer button on your phone Dial extension 7000 default PBX announces the parked extension e g 7001 7002 etc Hang up the phone To retrieve the parked call dial the previously announced parked extension 7001 7002 etc 10 6 Call Hold and Retrieve Enables users to automatically hold and retrieve incoming calls without requiring the use of feature access codes This feature is especially useful for attendants managing a large volume of incoming calls Nee Ee Any user can put a call on hold by pressing the Flash key The extension on hold will start getting Music on hold The same call can be retrieved back by pressing the flash key again 10 7 BLF Busy Lamp Field Support PSTN VoIP PBX supports BLF which sends information about other extensions to a phone connected to the same PBX to inform the status By default BLF Support is enabled This feature is used by a receptionist or secretary for routing incoming calls User Manual v2 8 www cem solution net
27. as to start creating Click on the Create users followed by apply changes button to update in configuration 6 6 Ring Groups Navigation Additional Features gt Ring Groups gt Create New RingGroup Using this option you can dial more than one extension simultaneously or to ring more than one phone sequentially The Ring Groups details are as follows User Manual v2 8 www cem solution net NANO 2 Status Ring Group Name Name of the Ring Group Extension for this Ring Group This is extension dialed to make all the phones ring in a group Click on the save button followed by selecting the ring Group members from the available users and configuring the ring Group Options New RingGroup RingGroup Name D ringgroup Extension for this ring group D 779 Ring Group Members Available Users Ring Group Options Strategy D Ring in Order ei Seconds to ring each member D 20 f not answered Goto D Ignore redirections D User Manual v2 8 www cem solution net NANO 2 Status 6 7 Paging Intercom Navigation Additional Features gt Paging Intercom Create new Page Intercom Group Using a Page Intercom Group an announcement can be made over the speakerphone or a group of phones Create a Page Intercom Group by giving an Extension for this Group and selecting the type of Group i e either 2 Way Intercom or 1 Way Paging for calling individual or group of Extensions Once made the co
28. ations line or wireless link and typically share the resources of a single processor or server within a small geographic area for example within an office building NETMASK Used by the TCP IP protocol to decide how the network is broken up into sub networks ex 255 255 255 0 PBX Private Branch Exchange An in house telephone switching system that interconnects telephone extensions to each other as well as to the outside telephone network PROXY A server that receives requests intended for another server and that acts on the behalf of the client behalf as the client proxy to obtain the requested service A proxy server is often used when the client and the server are incompatible for direct connection For example the client is unable to meet the security authentication requirements of the server but should be permitted some SIP Session Initiation Protocol An application layer control protocol a Signaling protocol for Internet Telephony SIP can establish sessions for features such as audio videoconferencing interactive gaming and call forwarding to be deployed over IP networks thus enabling service providers to integrate basic IP telephony services with Web e mail and chat services In addition to user authentication redirect and registration services SIP Server supports traditional telephony features such as personal mobility time of day routing and call forwarding based on the geographical location of the person being call
29. call back between your phone and remote party 2002 dial 72002 from your phone then it will hang up automatically The remote party side he will get the call and once he answers the call you will get the call back Suppose If remote party didn t answer the call it will retry the call with retry count Default code 45 To deactivate Day Night manual switching Default code 46 To enable the Day mode for a specific Time interval Default code 47 To enable Night Mode for a specific Time Interval NANO 2 Additional Features Enable DND Default code 78 This is the code to enable DND for a particular Extension so that it will not accept any calls Disable DND Default code 79 This is the code to disable DND for a particular Extension so that it will accept calls Enable Call Duration Call Duration Timeout lt is the maximum timeout for an outgoing FXO call for which the call is established 5 4 Admin Settings 5 4 1 General Preferences Navigation Admin Settings gt General Preferences This is where you can configure the General Settings of the PSTN VoIP PBX General Preferences Global Outbound CID Configure the global CallerlD used for all outbound calls when no other CallerlD is defined with higher priority If no CallerlD is defined for extension or trunk the global outbound CID will be used as CallerlD Global Outbound CID Configure the global CallerlD Name used for all outbound calls
30. d which all fields can be used as default 4 5 Dial plans Navigation SETUP Dial plans This is where you configure Dial plans for the users A Dial plan is a collection of outgoing rules Dial plan 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 User Manual v2 8 www cem solution net NANO 2 Setting up Features 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 Manage DialPlans A Dial Plan is a collection of Outgoing Call Rules Dial Plans are assigned to Users to specify the dialing permissions they have For example you might have one Dial Plan for l
31. e 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 7 4 Asterisk CLI Navigation Tools gt 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 PSTN VolP user extensions with the registration status 7 5 Diagnostics On the PSTN VoIP PBX users could capture traces ping remote host and traceroute remote host for troubleshooting purpose under Web GUI gt Tools gt Diagnostics Ping Enter the target host in host name or IP address Then press Start button The output result will dynamically display in the window below Ping Enter IP 192 169 0 152 Go PING 192 168 0 152 192 168 0 152 56 data bytes 64 bytes from 192 1668 0 152 icmp seq 0 ttl 125 time l 64 bytes from 192 165 0 152 icmp seg l trl 125 time dO 64 bytes from 194 168 0 152 icmp seqre ttl 1l2o timed 64 bytes from 192 165 0 152 icmp seq s trl 125 time l 64 bytes from 194 168 0 152 icmp seq 4 ttl 1z28 time m IS IS GO M om e us gt lez das Dade pang statistics A packets transmitted 5 packets received OF packet loss round trip win argmax O 3 0 7 1 2 ms User Manual v2 8 www cem solution net NANO 2 Managing amp Handling PSTN VoIP PBX 8 Status 8 1 Call Detail Records Navigation Call Detail Records This is where you ca
32. ed PSTN Public Switched Telephone Network This is defined as the regular telephone network services VoIP Voice over Internet Protocol The technology used to transmit voice conversations over a data network using the Internet Protocol Such data network may be the Internet or a corporate Intranet WAN Wide Area Network A computer network that spans a relatively large geographical area Typically a WAN consists of two or more local area networks LANs User Manual v2 8 www cem solution net
33. efault LAN IP address is 192 168 113 1 amp you cannot change the ipaddress of LAN interface 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 User Manual v2 8 www cem solution net NANO 2 Additional Features 5 2 3 Host Configuration Host Configuration is used to manage your PSTN VoIP PBX 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 5 3 Feature Settings Navigation gt Call Features extensions is where you can configure the Call feature settings for the Feature digit time out maximum time ms between digits for feature activation default is 1000 ms Call Parking Settings Extension to Dial to Park a Call Default Extension 700 Extensions for Parked Calls Parked Call Timeout in secs eDuring an active call initiate blind transfer and then enter this code to park the call Default Extension 701 720 e These are the extensions where the calls will be parked i e parking lots that the parked calls can be retrieved Default setting 1000 e This is the timeout allowed for a call to be parked After the timeout if the call is not picked up the extension who parks the call wil
34. ension with a 7 digit number through the phone company s central office DNS The Domain Name System is the system that translates Internet domain names into IP numbers A DNS Server is a server that performs this kind of translation FXO In telecommunications a Foreign Exchange Office or FXO is a telephone signaling interface that receives POTS or plain old telephone service FXS Foreign Exchange Station is the interface on a VoIP device for connecting directly to phones faxes and CO ports on PBXs or key telephone systems GATEWAY A network point that acts as an entrance to another network IP ADDRESS Every machine that is on a network a local network or the network of the Internet has a unique IP number four sets of numbers divided by period with up to three numbers in each set i e 192 168 0 100 If a machine does not have an IP address it cannot be on a network IVR Interactive Voice Response A system to automatically manage incoming calls IVR can link phone callers voice and or touchtone with a computer database It can accept a question access the company s database and provide a caller with the information they are seeking It can also take information from the caller convert it to data and input that data to the database User Manual v2 8 www cem solution net NANO 2 Glossary of Terms LAN Local Area Network A LAN is a group of computers and associated devices that share a common communic
35. ere 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 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 4 2 3 Busy Lamp Field BLF Shows mainly three status 1 Ready 2 Ringing 3 On the phone Busy Lamp Field BLF is a light on an IP phone which tells you whether another extension connected to the same PBX is busy or not This has to be configured manually from the phone user and it is usually done by making use of the web interface When configured the phone subscribes to a resource list available on an IP PBX to be notified with such information about other extensions Here is the configuration in eyebeam softphone User Manual v2 8 www cem solution net NANO 2 Setting up Features Settings Presence Choose Setting Category Presence mode Presence Agent i SIP Accounts 2 197 168 0 87 Peer to Peer Settings o Jerver Foll time 00 seconds gt Firewall MAT SR Server side Storage Presence Tunnels Presence Agent Advanced GH Add a New SIP Account E Media Update authorization policy via CAP E System Subscribe to contact list Di User Interface Diagnostic
36. ess Events If a caller presses a key while waiting in the queue this selects which voice menu should process the key press Max Len The maximum number of callers waiting in queue for an available extension Retry to retry the call after given seconds 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 8 www cem solution net NANO 2 Setting up Features Edit Queue 333 Extension 333 Name 333 strategy ringan Music On Hold detent LeavevVhenEmpty ve JonEmpty ves Format GM Enable Recording C Gueue Options Member Gueue 15 Wrapup D TimeOut TimeOut Time S C Auto Fill C Auto Pause a TEPON mal Time KeyPress None Events Max Len 5 Retry D Agents Ir bob jones Analogi11 M 13 Analog 13 D 14 Analogi F 5600 SIP 5600 M 999 Agent 999 update cancel 4 9 2 Agent Login Settings Navigation Call Queues gt Agent login settings Enter an agent login extension that can be dialed to login to a specific queue If want to remain online and dial the extension then give agent callback extension Save the changes made to logout of agent login Hang up your phone To logout of agent callback Login dial the same extension used to login when prompted dial the extension and password with Agent Login S
37. etion of Bootup process Firmware upgrade All the Front panel LEDS start blinking and LEDs will turn orange after the completion of Firmware upgrade Hold the reset button till all the LEDs turn Blue About 5 6 Secs User Manual v2 8 www cem solution net NANO 2 Getting Started 2 3 Quick Installation Wizard By following the Wizard you will create a very basic configuration file allowing you to make and receive calls thus confirming your PSTN VOIP PBX is fully functional Most of the features wont be visible until you complete the wizard and access the full fledge user interface After the completion of Quick setup wizard PSTN VOIP PBX will reboot and launch the basic configuration file you ve created You are ready to test You will also have access to the full fledge user interface and all the features offered Once the reboot is done Connect the WAN port of the PSTN VoIP PBX to your network switch also you can remove the LAN connection and use the WAN IP to access the Full Fledge User Interface of the PSTN VOIP PBX The WAN IP address can be obtained by dialing from the Analog phone which is connected to the PSTN VoIP PBX 2 4 Accessing the Main GUI Graphical User Interface Connect an Analog phone to the any of the FXS ports of the PSTN VoIP PBX and dial to get the WAN IP address of the unit Using the IP address obtained open the Web browser and type http lt WAN IP Address 80
38. ettings Agent Login Extension 5421 D Agent Callback Login Extension 522 D To logout of Agent Login Hangup your phone To Logout of Agent Callback Login Dial the same extension used to login specify your extension and password when prompted and hit when asked for your callback extension This will successfully log you out of all queues you are a part of Agent Logout User Manual v2 8 www cem solution net NANO 2 Setting up Features e Agent Login Extension Extension to be dialed for the agents to login to the specific queue e Agent callback login extension Extension to be dialed for the agents to login to the queue they are part of except there is no need to remain online 4 9 3 Agent Settings Navigation Call Queues gt Agent Settings To create an Agent Login simply fill in all the required details including Agent User ID Agent Password and Agent Username When done click the Update button New Agent Agent User IO Io Dee e Agent User ID Unique ID of the Agent Login e Agent Password Password of the Agent Login e Agent Username Username for the Agent 4 10 Greeting Messages 4 10 1 Recording Voice Files Navigation Setup gt Greeting Message gt Record a new voice menu prompt Record a new Voice Menu prompt File Mame Format GSM dial this User Extension to record a new voice prompt 201 ze Recora cancet e Specify the Voice file name e Select
39. git from 2 9 Wildcard Match one or more characters I Wildcard Match zero or more characters immediately Destination Select the default destination for the inbound call e Extension e User s Voicemail e Conference Rooms e Voicemail Group e VR e Queues e Hang up Edit Incoming Calling Rule Tunk trunk_1 Time Interwal none Trunk 27 analogi Sal Time Interval Mone no Time Intervals matched e Pattern 5 Destination User Extension 1111 User Manual v2 8 www cem solution net NANO 2 Setting up Features 4 13 DID Routing Navigation SETUP gt 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 out dial 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 New DID DID Number 1 Destinations D Hangup be Elsave Cancel DID Configuration parameters DID Number Enter the DID numbers provided by the VOIP Service provider Destinations Select the DID destination Only the selected category can be reached by DID e User Extension e Conference e Call Queue e User s Voice Mail e Voicemail Group e Ring Group e Page Group User Manual v2 8 www cem solutio
40. ial ARG2 RINGTIME DIALOPTIONS Call Queues Voice Menus Time Intervals Incoming Calling Rules User Manual v2 8 www cem solution net NANO 2 Managing amp Handling PSTN VoIP PBX 8 3 System Info The PSTN VoIP PBX status can be accessed via Web GUI gt Status gt System Info which displays the following system information e General e Network e Disk Usage e Memory Usage NanoPBX System Status System Information Configure Hardware Trunks Outgoing Calling Rules Dial Plans inmware Version F Ve CPxX 003 R308 05032013 users Music On Hold Uptime Re E E 202397 Load Average 1 60 1 22 1 12 Time Intervals Asterisk Build A isk Incoming Calling Rules Voicemail Asterisk GUI version S N branch 2 1 0 rel Server Date amp TimeZone Thu Mar 7 13 54 28 IST 2013 Follow Me Hostname Directory uclibc Call Features WVoiceMail Groups Voice Menu Prompts You will have a summary of your general information such as Firmware Version Uptime Asterisk Build Server Date amp 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 gt Network Settings for more info Disk Usage Disk you will have a summary of Disk usage and Disk free space available on the file system Memory Usage Memory Status you will see total memory
41. icehdal Access DN code D Email Address T VO Settings SIP Password MAT ke ON Lan Reinvite Ei ON LTMF Mode RFC2633 S D ISeCUFe no l ON Advanced Options m In Directory D v ls Agent D 1 Incoming Callrecord LL E Outgoing Callrecard A Dickup Group i A Enable Voicemail for the User Enable this extension so that it can be visible when adding members to the Voicemail Group and Voicemail The Voicemail 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 Email Address Mail address which the user wishes to receive mails for Voicemail VoIP Settings SIP Password Configure the password for the user NAT Use NAT when the PSTN VoIP PBX is on a public IP communicating with devices hidden behind NAT e g broadband router If there is one way audio issue usually it s related to NAT configuration or Firewall s support of SIP and RTP ports Can Reinvite By default the PSTN VoIP PBX will route the media steams User Manual v2 8 www cem solution net NANO 2 Setting up Features from SIP endpoints through itself If enabled the PBX will attempt to negotiate with the endpoints to route the media stream directly It is not always possible for the PSTN VoIP to negotiate endpoint to endpoint media routing The default setting is No
42. l be called back PBX Feature Prefixes This lets the user assign the codes for different features present in the PSTN VOIP PBX Feature Map Blind Transfer User Manual v2 8 www cem solution net Default code 1 e Enter the code during active call After hearing Transfer you will hear dial tone Enter the number to transfer to Then the user will be disconnected and transfer is completed NANO 2 Additional Features Disconnect Attended Transfer Call Parking Call Pickup Call Record Call Back Normal Time Mode Day Time Mode Night Time Mode User Manual v2 8 www cem solution net Default code 2 This is to assign the code for performing call disconnect Default code 3 Enter the code during active call After hearing Transfer you will hear the dial tone Enter the number to transfer to and the user will be connected to this number Hang up the call to complete the attended transfer Default code 4 Enter the code during active call to park the call Default code 8 This feature code is to pick up the ringing extension from another extension if the party is not available in the desk Default code 9 This is to assign the code for performing Call Record If the PSTN VoIP PBX user extension wants to record the particular active call then he needs to press Call record feature code and that call conversation get recorded Default code 7 E g If you want to make
43. low to perform a Blind or unattended call transfer With an active call in progress press the 8 button This puts the original caller on hold and gives you a dial tone on a second line Dial the party that you wish to transfer to Once you transferred the call the transferor call will get hang up Only the original caller and the party you transferred to are now connected 10 3 Conferencing This is type of conference will allow eight parties in the bridge It s a conference bridge where all the callers will enter into the room User Manual v2 8 www cem solution net NANO 2 Managing amp Handling Nano2PBX Important Note PSTN VoIP PBX 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 10 4 Call Forwarding Call forwarding Diverting is a telephony feature If your phone is unreachable out of service area or you do not wish to receive calls the incoming calls can be diverted to other phone numbers The Call Forward feature enables incoming calls to be redirected to any other telephone in the country Outgoing calls can still be made even after you have activated Call Forward Only incoming calls are redirected to another number Please refer Section 2 1 to activate and operation of this feature 10 5 Call Parking Enables a user to hold a call and to retrieve it from another station within the group To p
44. mat osm K Upload to Server Username Password FTP Server Bil PotD 21 default is 21 Path to Save File ve a nT Uploading Based on E SCHEDULE For Ewery hrs save Hisave Cancel cancei You can select either GSM or Wav format in the call record settings Call Record to E mail Check this option to send Call records via E mail If this option is selected call recording will be sent to the admin email id which is configured in the SMTP settings Format Select the format in which calls has to be recorded 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 User Manual v2 8 www cem solution net NANO 2 Status 6 5 Bulk Add Navigation Additional Features gt Bulk Add gt Create range of new users This is where you can create many user extensions all at a time Bulk Add Bulk Add Create a Range of new users Create Users Starting from Extension Tip Use the Modify Selected Users button from the Users page to edit any options for the created users 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 h
45. n 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 PSTN VoIP PBX 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 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 Ae 8 2 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 NanoPBX System Status Channel Management Configure Hardware Toma Refresh Now Outgoing Calling Rules f i Refreshing Active Channels in 8 Seconds Dial Plans Channel State Seconds Application AC 2 00000003 Ringing d AC 4 00000002 Ring 7 Dial ARG2 RINGTIME DIALOPTIONS Users Ring Groups Music On Hold AC O 00000001 Up 25 AC 1 00000000 Up 27 D
46. n net NANO 2 Additional Features 5 4 4 Reboot Navigation Admin Settings gt Reboot This is where you can configure the General Settings of the PSTN VoIP PBX The administrator of the PSTN VoIP PBX can remotely reboot the PSTN VoIP PBX 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 PSTN VoIP PBX Important Note 1 PSTN VoIP PBX will take about 3 minutes time to reboot and keep patience Rebooting the PSTN VoIP PBX results in termination of active calls 2 IP address of the PSTN VoIP PBX after reboot Please check the IP Address by dialing from the Analog Phone connected to FXS port of the PSTN VoIP 3 While rebooting the PSTN VoIP PBX the front panel LED s will blink in blue orange once PBX booted the LED s will be orange 5 5 SIP Settings 5 5 1 General Navigation SIP settings gt General This is where you can configure the General sip settings Bind UDP Port Configure the UDP port used for SIP The default setting is 5060 Bind IP Address Configure the IP address to bind to The default setting is 0 0 0 0 which means binding to all addresses Enable DNS SRV Select to enables DNS SRV lookups on outbound calls from the lookups PSTN VoIP PBX The default setti
47. n net NANO 2 Setting up Features 5 General Settings 5 1 Voicemail Navigation Voicemail This is where you can manage the configurations of the voicemail 5 1 1 General settings Accessing Retrieving amp Managing Voice Mail The PSTN VoIP PBX allows users to manage voicemail through voice messages in their phones This section will Summarizes how to access retrieve and manage voicemail and other settings The default feature code for accessing Voicemail 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 voicemail 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 amp 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 5D Leave Message 4 Play previous message D Repeat current message 6 Play next message 7 Delete current message 8 Forward message to another mailbox User Manual v2 8 www cem solution net NANO 2 Additional Features 1 Use Voicemail number 2 Use Voicemail 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
48. nfigurations save the changes Page an Extension This option is used to make the Settings for paging individual Extension Settings This option is used to add Alert Info Header value that is sent to the phone for an Intercom call Edit Pageintercom Group 778 Extension for this FPagellntercom Group 778 Type 2 Way Intercom w Play a beep D Fayeintercom Group Members Available Users SAC User Manual v2 8 www cem solution net NANO 2 Status 7 Tools 7 1 Back Up The PSTN VoIP PBX configuration can be backed up locally or via network The backup file will be used to restore the configuration on PSTN VoIP PBX when necessary Users could backup the PSTN VoIP PBX configurations for restore purpose by clicking on Create New Backup under Tools gt Back Up Once the backup is done the list of the backups will be displayed with date and time in the web page Users can download restore or delete it from the PSTN VoIP PBX internal storage or the external device Besides local backup users could back up the configuration a remote server via TFIP HTTP protocol under Web GUI gt Tools gt Back Up RESTORE CONFIGURATION FROM BACKUP FILE To restore the configuration on the PSTN VoIP PBX from a backup file users could go to Web GUI gt Tools gt Backup A list of previous configuration backups is displayed on the web page Users could click on of the desired backup file and it will be restored to the PSTN
49. ng is No User Manual v2 8 www cem solution net NANO 2 Additional Features 5 5 2 TOS Navigation General Settings gt SIP Settings gt TOS This is where you can configure the TOS sip settings The Details to be filled are given as TOS Generate In band Ringing Server User Agent DTMF Mode Maximum Registration Subscri ption Time Min Registration Subscri ption Time Default Incoming Outgoing Registration Time User Manual v2 8 www cem solution net Configure whether the PSTN VoIP PBX should generate inband ringing or not The default setting is Never e Yes The PSTN VoIP PBX will send 180 Ringing followed by 183 Session Progress and in band audio e No The PSTN VoIP PBX will send 180 Ringing if 183 Session Progress has not been sent yet If audio path is established already with 183 then send in band ringing e Never Whenever ringing occurs the PSTN VoIP PBX will send 180 Ringing as long as 2000K has not been set yet Inband ringing will not be generated even the end point device is not working properly Allows you to configure the user agent string for the PSTN VoIP PBX Select DTMF mode to send DTMF The default setting is RFC2833 If Info is selected SIP INFO message will be used If Inband is selected 64 kbit codec PCMU and PCMA are required When Auto is selected RFC2833 will be used if offered otherwise Inband will be used The default setting is RFC2833
50. ocal calling that only permits users of that Dial Plan 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 Dial Plan that includes both the local and longdistance outgoing calling rules 4 6 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 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 Day Night Mode You can select either day or night mode for a specific time interval All other time intervals will be in normal Mode User Manual v2 8 www cem solution net NANO 2 Setting up Features Edit Time Interval rule Time Interval Name rulet K Osy day of week d Ogy Days of a Month Date 19 1171 7171 Month November Time T Entire Day Start Time 14 00 4M End Time 12 50 PM Day Night Mode Day Mode Night Mode 4 7 Voicemail Groups A Voicemail Group is a pre programmed group of voicemail recipients All the members of this group will receive the same
51. our 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 4 3 3 Adding a New VoIP Account Details Create New SIP trunk Context Naming You can select the context naming based on username Provider name Assigned by asterisk GUI By Default Based on Username Provider Name Configure a unique label to identify this trunk when listed in outbound rules inbound rules and etc Host Name Configure the IP address or URL for the VoIP provider s server of the trunk Host Port Enter the port number given by the VOIP provider 5060 is the default SIP port If configuring any roaming extension or if connecting PSTN VoIP to public network then consider changing the SIP port for better security Once port changes are done for registering IP phones sip phone use lt domain name gt lt sip port number gt Username Enter the username to register to the trunk from the provider Password Enter the password to register to the trunk from the provider Qualify lf you enable Qualify asterisk will send the SIP Options command regularly to check that the device is still online Fax Detect Configure a Fax destination where incoming Fax will be detected and received User Manual v2 8 www cem solution net NANO 2 Setting up Features DID Routing DID Routing Routes calls
52. p e Ring Group e Page Group Mame IvVRiGiorna Extension 301 D Allow Dialing Other Extensions Actions D Play recordiAperntura_CoNSsS_ nano amp Listen for KeyPress events Wail OI sec for the user to enter an extension GoSUub Queue 9007 Add new Step Select an Option D Allow keyPress Events 0 GoSub Queue 9001 1 Goto User 201 2 Sota User 202 3 Sota User 203 Important Note Advanced Edit 1 Before you start to Create Voice Menu you need to have Greeting Messages Navigation Setup gt Greeting Messages User Manual v2 8 www cem solution net NANO 2 Setting up Features 2 If SIP trunk is only supporting G729 then IVR greeting recordings should be in G729 format 4 12 Incoming Calling Rules Navigation SETUP gt 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 Incoming Calling Rule Configuration Parameters Trunk Select the trunk to configure the inbound rule Time Interval Select the time interval from the list specify the time for the trunk to use the inbound rule Pattern All patterns are prefixed with the _ X Any Digit from 0 9 Z Any Digit from 1 9 N Any Di
53. pt NAT mode or RFC3581 support e ROUTE Assume NAT don t send report Allow RTP Reinvite If enabled the PSTN VoIP PBX will try to redirect the RTP media stream audio to go directly from the caller to the callee The default setting is No NAT e Yes e No NAT Allow media path redirection Reinvite but only when the peer is not be behind NAT The RIP core can detect if the peer is behind NAT or not based on the IP address where the media comes from e Update Use UPDATE for media path redirection instead of INVITE Note Some devices do not support this especially if one of them is User Manual v2 8 www cem solution net NANO 2 Additional Features behind NAT 5 5 4 Misc Navigation General Settings gt SIP settings gt Misc This is where you can configure the miscellaneous sip settings The details to be filled are given as FAX Settings FAX Format Select the FAX format either in TIFF or PDF T 38 fax UDPTL Enables T 38 Fax Mode otherwise pass through mode is enabled Pass through Make sure that baud rate fax speed must be set 9600 in fax machines when pass through mode is enabled Fax Email ID Assign the Email Address to which the FAX attachment has to be sent Fax Destination Select the Destination where fax must be received Fax Detect TimeOut Specify the time for which the FAX has to be detected Out Bound SIP Registration Settings Register Register as a SIP user agent to a SIP proxy
54. racter followed by 1 followed by any digit from 2 9 followed by 8 more digits are dialed immediately e 2 9 xxxxxxxxx Where calls beginning with any digit from 2 9 followed by 9 more digits are dialed immediately e 2 9 xxxT Where calls beginning with any digit from 2 9 followed by three more digits are dialed after Digit Timeout is reached e 2 9 11 OT 011xxx T O 2 9 xxxxxxxxx 1 2 9 xXxxxxxxx 2 9 Xxxxxxxxx 2 Q xxxT where each entry is separated by the character For more information please refer to RFC 3435 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 5 4 2 Language Navigation Admin Settings gt Language This is where you can configure the Language Settings of the PSTN VoIP PBX Language settings allow the user to specify the default language voice prompt The currently available voice prompts are English Spanish Turkey French and Italy By default selected voice prompt language is English 5 4 3 PBX Settings Navigation Admin Settings gt PBX Settings This is where you can configure the General Settings of the PSTN VoIP PBX Change Admin Password e Enter the new password and retype the new password to confirm The new password field has to be at least 5 characters e Click on Update and the user will be logged out User Manual v2 8 www cem solution net NANO 2 Additional Feature
55. ration 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 PSTN VoIP PBX That means you can configure these ports to be either as FXS ports to connect Analog Phones or FXO ports to connect PSTN line 4 2 Extensions Analog amp SIP Extensions are the core of the PSTN VoIP PBX An extension is a number mapped to a person So basically every employee that is connected to the PSTN VoIP PBX should have their own unique extension number so that he she can be reached and be able to place calls The PSTN VoIP PBX supports 2 types of Extensions IP Extensions and Analog Extensions User Manual v2 8 www cem solution net NANO 2 Setting up Features IP Extensions IP extension are devices that have only data networking connection such as Ethernet and they communicate with the PSTN VoIP PBX using IP based protocol for signaling and Voice examples are IP Phone Soft Phone application The PSTN VoIP PBX can support up to 75 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 PSTN VoIP PBX The PSTN VoIP PBX 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 Voicemail Groups Also the exten
56. rname Port Hostname IP Office Office2 General Settings Additional Features Ports 1 Ports 2 Analog System Info Firmware Version Analog Tools Status Extensions P All Extension oa 212 213 EY CES CES e 1010 1011 om 502 501 500 200 No Extension assigned Queues amp lt User Manual v2 8 www cem solution net Name Label Reception Develapment Technical Management Plunkett Mobile Plunkett PC 1010 1011 1111 Closed Lunch VR Check Voicemails Dial by Names Status essages 1 0 Ure Free Ringing v Busy UnAvailable Type Analog User Port 1 Analog User Port 2 Analog User Port 3 Analog User Port 4 SIP User SIP User SIP User SIP User SIP User Voice Menu Voice Menu Voice Menu VoiceMailMain Directory CPZ 003 R39609072 013 Uptime Phi 58221 Un 21239 Load Average 1 26 1 29 1 16 Asterisk Build Asterisk 1 6 2 18 Asterisk GUI version SVN branch 2 1 0 rel Server Date amp Timezone Sue tee te 27 2023 Hostname uclibc NANO 2 Setting up Features 4 Setting up the Features 4 1 Configure hardware Analog Hardware Port Configuration User Trunk FXS Port 211 Reception FXS Port 212 Development FXS Port 213 Technical FXS Port 220 Management FXS Port unassigned FXS Port unassigned FXO Port Office FXO Port i Office Analog Hardware Setup amp Configu
57. s e Once the web page comes back to the login page again enter the new password to login Date amp Time Settings You can set the date and time of the PSTN VoIP PBX 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 Enable NTP Enabling NTP allows you to set the Time zone and Date based on the NTP server provided Time Zone Select the proper time zone option so the PSTN VoIP PBX can display the correct time accordingly NTP Server Specify the URL or IP Address of the NTP server for the PSTN VoIP PBX to synchronize the date and Time For Eg pool ntp org Email Settings The Email application on the PSTN VoIP PBX can be used to send out Emails to users with Fax eg Fax To Email Voicemail Voicemail To Email and other information as attachment Email Settings SMTP The IP Address or domain name of the SMTP server SMTP PORT The port number at which SMTP server is running Generally SMTP port No is 25 AUTH User Authorized username of the Admin Email ID AUTH Password Authorized password of the Admin Email ID Admin Emailed Specify the Administrator Email ID TLS Support TLS setting to require mail to be transmitted via a secure connection when users correspond with specific domains and email addresses Auth Login Enable this option act as Auth login for SMTP Authentication User Manual v2 8 www cem solutio
58. s Address tusername t ist tdomaint Refresh interwal 300 BE Refresh interval 3600 seconds License key Contact Properties Display Name Address 301 Calls amp Contacts e Detach e Available Group Received Calls Dialed Calls 6 Extension XFER HOLD PARK Aa AC OND CONE m m E e e Powered by in Ay e OU User Manual v2 8 www cem solution net NANO 2 Setting up Features Calls amp Contacts eg Bl Detach Yi Available Received Calls Dialed Calls gt Extension 801 Ready amp 501 Ringing bam sf TI A 803 On the phone Powered by Aten ge 4 3 Trunks Navigation Setup gt Trunks gt Analog Trunks Here you can configure the Analog Trunks 4 3 1 Analog Trunks The analog trunk options are listed in the table below Analog Trunks Channels Select the channel for the analog trunk Trunk Name Specify a unique label to identify the trunk when listed in outbound rules incoming rules and etc Fax Detect Configure a Fax extension where incoming Fax will be detected and received User Manual v2 8 www cem solution net NANO 2 Setting up Features 4 3 2 VOIP Trunks Navigation Setup gt Trunks gt 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 y
59. sion can have a voicemail of its own 4 2 1 Create New User You have to create atleast one Dialplan using Dialplan option before trying to create Analog IP Extensions Navigation Users gt Create New user This is where you setup your Analog IP extensions Technology SIP Select SIP if the user is using SIP or a SIP device Analog Station select the FXS port need to be configured from the drop down menu Codec Preference Select audio and video codec for the extension The available codecs are G711U law G711A law G 726 G 722 G 729 H 263 and H 263p Extension The extension number associated with the user CallerlID Name Configure the CallerID Name associated with the user Dial Plan Select one from the dropdown box appears only if dialplans are created using Dialplans By Default Dialplan1 CallerID Number Configure the CallerlD Number that would be applied for User Manual v2 8 www cem solution net NANO 2 Setting up Features outbound calls from this user Note The ability to manipulate your outbound Caller ID may be limited by your VoIP provider Edit User Extension 33 Advanced Edit Technology C sp O Analog Station port D D Codec Preference First u taw Second Jan a Third G 729 ed Fourth H 263 Fifth Hone General Extension 32 D Caller Name 33 D DialPlar DialPlant D Internal Caller 33 D Caller Number 33 D E Enable Yoicemail tor this User D Vo
60. ssages D Direct voicemail Dial M Max greeting in seconds Dial 0 for Operator On Message Options Maximum messages per folder OI 100 d Max message time i minute sl Win message time E second Playback Options Say message Caller ID Say message duration Play envelope I I I 0 Allow users to review User Manual v2 8 www cem solution net NANO 2 Additional Features Email Settings for Voicemail This section will summarize how to configure and notify the user if he she has received a new voicemail to their email address The caller after leaving the voicemail mail will be sent to the user mail address as mentioned while configuring the user extensions Voicemail Email Alert Preference Send message by Email lf enabled Customized message can be sent to users email address Attach Recordings to Email lf enabled voicemails will be sent to user s Email address as voice recording The default setting is Yes Template for Voice mail Fill in the Subject and Message content to be used in the Emails Email when sending to the users The template variables are e V TAB e VM_NAME Recipient s first name and last name e VM_DUR The duration of the voicemail message e VM_MAILBOXt The recipient s extension e VM_CALLERID The caller ID of the person who has left the message e VM_MSGNUM The number of messages in the mailbox e VM_DATE The date
61. te the MOH classes and create them User Manual v2 8 www cem solution net NANO 2 Setting up Features Manage Music on Hokdl Classes Manage Music On Hold Classes select MOH class default se New MOH class manage MOH class default List of Sound Files Note Changing MOH Class requires REBOOT Upload an 8 KHz Mono Music ie Du Choose file to Upload No file selected q ur 4 9 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 4 9 1 Configuring a Queue Navigation SETUP gt 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 User Manual v2 8 www cem solution net NANO 2 Setting up Features Queue Details Extension Extension number
62. to a single specific extension TrunkingWithout Trunking Without Registration is to make voip trunk calls Registration from PSTN VoIP Without Registration Outbound Proxy Configure Outbound proxy to send Outbound signaling to that proxy 4 3 4 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 4 4 Outgoing Calling Rule 4 4 Outgoing Calling Rule 4 4 1 New Calling Rules Navigation Setup gt Outgoing Calling Rules gt 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 There are two ways to make outgoing calls for the registered extension users e VoIP SIP trunk vi
63. to reach the Queue directly Queue Name The name of the Queue Strategy Type o Ring All Rings all available extensions o Round Robin Takes turns ringing each available extension o 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 o 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 LeaveWhenEmpty This option controls whether callers already on hold are forced out of a queue that has no agents Join Empty This option controls whether callers can join a call queue that has no agents Format Select the format in which call has to be recorded Enable Recording Enable this to record queue calls Member Timeout Select the seconds an agent s phone will ring before the queue tries to ring the next agent 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 Auto fill The queue will complete as many calls simultaneously to the available agents Auto Pause Pauses an agent if they fail to answer the call Report Hold time Reports the hold time of the agents before the caller is connected to the agents Key Pr
64. voicemail message 4 7 1 Configuring a Voicemail Group Navigation Voicemail Group This is where you setup your Voicemail Groups To create a Voicemail 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 4 7 2 Voicemail Group Details Voicemail s Group Extension The number to be allocated for the Voicemail Group Name of the Voicemail Group User Mailboxes The list of available extensions that could be part of this Voicemail Group The marked extensions in the list of extensions belong to this Voicemail Group User Manual v2 8 www cem solution net NANO 2 Setting up Features New Voice Mail Group Voiceflail Group s Extension I m User MailBoxes Ram 202 fam cancel 4 8 Music on Hold MOH Navigation SETUP 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 PSTN VoIP PBX 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 from the drop down box and select from the Sound files listed Click on Apply Changes tab to save the changes made You can even dele

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