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Users

The QuBe Users module creates administrators a login to the PBX


  • Click the Create User + button

  • Username

    • Enter username for user to login to PBX



  • Password

    • Enter password for user to login to PBX



  • First Name

    • Enter first name to identify user



  • Last Name

    • Enter last name to identify user



  • Email address

    • Enter email address for user



  • Groups

    • From the User Groups module, select the User Group this User will be included in

    • This is optional however for recommended for managing operator and queue panel users, allowing access to certain modules for users, and more



  • Admin Access

    • Check box if user will have all admin privileges



  • Click submit to save

User Groups

The QuBe User Groups module is used to manage users created in the User module. 


  • Click the Create User Group + button

  • Group Name


    • Enter the Group Name to identify the created group



  • Description


    • Enter the description of the group to identify the users and allowed modules (e.g. Sales, Panel Members, etc.)



  • Allowed Modules


    • All modules in the QuBe PBX are listed. Check off each module to allow users in the group access to the module



  • Users


    • Multi-select user(s) to allow access to the modules selected



  • Click submit to save

  • Text Hover

Custom Branding

​The QuBe Custom Branding module is used for Quest Blue customers who will use the QuBe PBX to supply to their own customers. 


  • Account Key

    • In order to use this feature you must enter your valid account key in the account key field

    • This is 100% white label, and 100% free when using QuestBlue SIP & DIDs



  • Company name


    • Enter your company name in here to appear on the QuBe



  • Company Website URL


    • Enter your company’s web address to be seen at the bottom of the PBX – must be in the form http://siteaddress.com



  • Company Logo


    • Upload the image to use as the company logo – this will replace the QuestBlue logo in the upper left corner



  • Company Small Logo

    • Upload a smaller logo to use for a minimized or mobile browser version of your PBX’s site



  • Click Submit to save


To ensure proper formatting of your logo the maximum height should be no more than 20 pixels. 

For mobile scaling you will also need to upload a separate smaller image. 

Bulk

The QuBe Bulk configuration is used to download templates from the extension module and from the inbound route module to be edited in bulk and to be uploaded back to the PBX. The upload section of the bulk configuration is to upload the template to the PBX. All the information uploaded must match the values accepted by the PBX, as shown in the drop downs.


  • Click the Create Speed Dial Number button

  • Name


    • Enter name of the entry



  • Number


    • Enter number that the speed dial code will call



  • Speed Dial Code


    • Create a Speed Dial Code different from any other code or extension on the PBX



  • Click the Submit button to save configuration


 


  • How to fill Destination


    • This dropdown shows you how to enter the template value to upload data for module destination



  • Inbound Route Available Values

    • This dropdown shows you how to enter the template value to upload data for module inbound routes



  • Extensions Available Values

    • This dropdown shows you how to enter the template value to upload data for module extensions



Disk Space Usage

​The QuBe Disk Space Usage module is for our customers who value monitoring their phone system. Many customers have a high call volume and depending on the PBX memory and hard drive space you are using you may need to monitor CDR reports and logs. Call recording is one of the biggest factors we see clog up systems for customers that manage their PBX. Setting alert limits at a certain percentage of usage the customers can be aware and prevent any issues that may clog up the PBX


  • Enable Disk Space Monitoring

    • Check the box to enable the disk space monitoring feature



  • Store CDR for (days)


    • Enter the maximum number of days to store CDR reports for



  • Store Call Recordings as CDR


    • Check the box to allow storage of call recordings in the CDR reports



  • Store Queue Log for (days)


    • Enter the maximum number of days to store queue logs for



  • Store Queue Recordings as Queue Log

    • Check the box to store queue recordings in the Queue Logs



  • Set Disk Space Usage Alert Limit (%)

    • Enter a percentage amount, and usage over that amount will alert the administrator



  • Set Disk Space Freeing Limit

    • Enter a percentage amount to being freeing space — logs and recordings will be deleted day by day until usage is less than alert limit



  • Click the Submit button to save


 

Feature Codes

The QuBe feature code module is used to create and edit codes for features available to users to dial from their registered phone. These feature codes can be to access voicemail, switch call flow between day or night mode, etc.


  • Click each section to expand and see feature code

  • Expand section also to edit or change feature code

  • Click the Submit button to save configuration

Speed Dial

​The QuBe Speed Dial module creates speed dial entries for users wanting to reach another number or extension.


  • Click the Create Speed Dial Number button

  • Name

    • Enter name for the entry



  • Number

    • Enter number the speed dial code will call



  • Speed Dial Code

    • Create a Speed Dial Code that is different from any other code or extension on the PBX



  • Click the Submit button to save

Recordings

The QuBe Recording module is used to upload recordings to the PBX to be used as announcements in other modules. Recordings can be uploaded to the PBX from this module or recordings can be recorded directly from this module and from a phone registered to the PBX


  • Click the Record From Extension button

  • The PBX will call the extension you enter to record from


    • Your Extension

      • Enter the extension number the PBX will call



    • File Name

      • Enter a file name to save the recording as





  • Click the Submit button. The PBX will call the extension

  • Answer the call and begin recording after the prompt. Hang up or press pound when finished 

  • Press the play button to listen to the recording 

  • Click Save Recording, or Cancel to record again

  • Click the Upload File button

  • Recording must be uploaded to the system before you can create the recording

    • Name

      • Name the recording



    • Click the Choose File button to upload recording from computer



  • Click Submit to save the recording to the PBX

  • Click the Create Recording button

  • Create a recording file which will play an uploaded recording or multiple uploaded recordings

  • Name

    • Name the recording



  • Description

    • Enter a unique description of the recording



  • Add the uploaded recording file saved on the PBX

  • Click Add More + to add additional recordings to sound when this record file is active

Time Settings

The QuBe Time Settings module sets the timezone to be used system-wide


  • Set New Timezone

    • Select the timezone for the PBX to use

    • System must reboot for changes to take effect



  • Click the Submit button to save configuration

Updates

The QuBe Updates module is used to update system packages to the most recent versions available. QuestBlue Systems will release an update for any new features or any bugs and performance improvements to always keep the QuBe PBX the best as we can for you!


  • Click the Check Updates button


    • Any new module updates available will display next to the appropriate module



  • Click Update All

Blacklist

The QuBe Blacklist module is used to block specific numbers from making incoming calls to any DIDs in the PBX. All numbers added to the blacklist are blocked from the entire PBX but these blocks can be removed using this module.


  • Click the Create Blacklist Number button

  • Number

    • Enter the number you wish the blacklist from calling the PBX



  • Description

    • Enter the description of the Blacklist entry



  • Click the Submit button to save configuration

Asterisk CLI

The QuBe Asterisk CLI is used to monitor the Asterisk command line without having to SSH into the server


  • Enter an Asterisk CLI command into the text box

  • Press Submit

  • Enter password for user to login to PBX

Extensions

The QuBe Extensions module is used to create extensions and customize individual extensions with voicemail options, follow me configuration, recording options, and more. Each extension created can be used to assign to a physical phone or a virtual phone, a soft phone. The extensions can be used as an endpoint in a call flow or it can be added to multiple modules.


  • Click the Create Extension button

  • Display Name

    • Enter a display name that will be associated with the extension. This will NOT be the Caller ID Name



  • Extension Number

    • Enter a unique extension number – we recommend 3 or 4 digit numbers



  • Extension Password

    • Extension password is the secret password to connect the extension in the PBX to the device (e.g. phone) that will be configured as the extension and is automatically filled out with a complex password



  • NAT

    • Set NAT based on PBX server configuration:

      • This phone will exist in local networks only

      • Pretend there is an rport parameter even if there isn’t (force_rport): Force rport to always be on

      • Send media to the port Asterisk receives it (comedia): Use rport if the remote side says to use it and perform comedia RTP handling

      • Set the force_rport option if Asterisk detects NAT (auto_force_rport): Force rport if Asterisk detects that an incoming SIP request crossed a NAT after being sent by the remote endpoint

      • Set the comedia option if Asterisk detects NAT (auto_comedia): Use comedia if Asterisk detects that an incoming SIP request crossed a NAT after being sent by the remote endpoint

      • Both force_rport and comedia options: refer to definitions

      • Both auto_force_rport and auto_comedia (recommended in common cases)





  • ​Outbound CID

    • ​Set outbound CID on extension. If you leave this blank, the PBX will assign an outbound CID based on route or trunk



  • User

    • ​Assign a user from the drop-down menu



  • Click Submit to save the initial setup of the extension


 


  • Extensions listed on front page:

  • Display Name

    • ​Display name assigned to the extension when created. This can be edited.



  • Extension

    • ​Extension number assigned to the extension when created. This is used internally



  • Status

    • Current registration status of extension



  • User

    • ​The user account the extension is assigned to



  • CID

    • The outbound CID assigned to the extension



  • No Answer

    • The extension’s destination if nobody answers



  • Busy

    • The extension’s destination if busy



  • Unavailable

    • ​The extension’s destination if set unavailable (DND, etc.)



  • Edit

    • ​This will open the page to edit the extension settings



  • Delete

    • ​This will delete the extension entirely – if there is no backup saved any deleted extension will be permanently lost



We suggest disabling the device associated with a deleted extension to avoid the phone attempting to register into the PBX with incorrect credentials and locking out your network IP

IP Trunks

The QuBe IP Trunks module is used to connect the PBX to QuestBlue systems SBC to allow the PBX to handle incoming and outgoing calls.

QuestBlue Trunk: As QuestBlue Systems is your provider, clicking the Create QuestBlue Trunk button will automatically fill in all of the appropriate trunk settings with QuestBlue’s specifications.


  • Click the Create QuestBlue Trunk Button

  • Trunk Name


    • Edit the trunk name if you wish to define it with a unique name



  • Outbound CID


    • If the outbound CID number is set in the trunk module it will be universal to all outbound calls



  • All SIP Settings are set by default to QuestBlue’s

  • For additional SIP settings drop down the SIP Settings section

  • ​Dialed Number Manipulation Rules are automatically defaulted to QuestBlue Specifications

  • Click the Submit button to save configuration


SIP Trunk: Set up the SIP trunk to your specifications.


  • Click the Create SIP Trunk Button

  • Trunk Name

    • Edit the trunk name if you wish to define it with a unique name



  • Outbound CID


    • If the outbound CID is set in trunk module it will be universal to all outbound calls



  • CID Options


    • CID options determine what outbound CID are allowed out from trunk



  • Trunk Type

    • Set the trunk type of either peer or user. Default is peer



  • Host

    • Host is the IP or DNS name of the SIP server



  • DTMF Mode

    • RFC2833 sends DTMF tones separately with specially encoded RTP packets. These are distinct audio packets but are within the same network connection.

    • Inband sends tones as inband audio with the voice stream. Dialing on the device will create DTMF tones.

    • Info is an out of band method which sends the DTMF signals over SIP using a separate network connection



  • NAT Settings

    • See extension page for explanation on NAT selection



  • Direct Media

    • Set YES or NO depending on how to pass trunk media



  • Qualify

    • Set YES or NO depending on how to enable peer qualifying



  • 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.



  • Authenticate Insecure

    • Authenticate insecure notifies how to authenticate to the peer



  • For additional SIP settings drop down the SIP Settings section

  • Enter Dialed Number Manipulation Rules so these rules can manipulate the outgoing number before sending it out of this trunk

  • ​Click the Submit button to save configuration


 

Inbound Routes

The QuBe Inbound Routes module allows calls to enter the PBX when the DID is dialed. The inbound route based on DID will follow the destination set within the specific inbound route.


  • Click the Create Inbound Route button

  • Route Name

    • Enter a name for this unique route



  • Incoming Number

    • Define the DID number you are creating this route for



  • CID Prefix

    • Allows incoming calls to this route to be prefixed with text



  • Music on Hold

    • Select music to play when incoming calls to this route are placed on hold



  • Call Recording

    • Controls recording based on selection on this route



  • Destination Type

    • ​Select the type of location to route incoming calls to through call flow



  • ​Destination

    • ​Select the location to route incoming calls to through call flow



  • Click the Submit button to save configuration

Outbound Routes

​The QuBe Outbound Route module is used to set which numbers based off of the dial plan can be dialed outbound from the PBX. From the outbound route you must set the trunk to send calls to, you can set an override to all caller ID's set in other modules, select the route type, and more.


  • Route Name

    • Enter route name



  • Route CID

    • Override Extension CID is an optional route for Caller ID to be used. If CID is entered, this setting will override all CIDs specified except:

      • Extension EMERGENCY Caller ID

      • Trunk CID

      • Forwarded call: CF, Follow Me, Ring Groups, etc.

      • Extension Outbound CID





  • Route Type

    • Drop down the menu to select YES or NO to override the extension’s outbound CID to the CID set in this route

    • Define Route Type if emergency or internal. If left blank the route will depend on dial pattern and order of weight



  • Call Recording

    • Enter call recording preference – this will override all other call recording settings



  • Trunk

    • Enter the trunk that will use this route



  • Dial Patterns

    • Enter the dial patterns to determine what number dialed will match the route



  • Click the Submit button to save configuration

Announcements

The QuBe Announcement module is used to add an announcement recording to the call flow. You can add an announcement before sending the call to the next destination. Typically an announcement may be used to announce to the caller the call may be recorded or a marketing announcement, etc.


  • Click the Create Announcement button

  • Name

    • Enter name to describe the announcement



  • Recording

    • Select the recording that will play when this announcement is used



  • Repeat Key

    • These options allow the announcement’s recording to repeat depending on selection

      • Allow skip:

        • Check box to allow caller to press any key to skip to the end of the recording and then continue to destination

        • Leave empty to prevent caller from skipping to the end of the recording. Caller will be forced to wait until recording is complete before proceeding to destination



      • Return to IVR:

        • Check box to allow caller to return to IVR, if previous destination was an IVR, instead of proceeding to set destination

        • Leave empty and caller will proceed to destination



      • Answer Channel:

        • We recommend checking this because the behavior is to answer the call and play announcement recording

        • If you leave this empty the recording would play as early media to the caller







  • Destination After Playback

    • Define where to route the caller after announcement is complete



  • Click the Submit button to save configuration

Call Flows

The QuBe Call Flow module was built to manually switch the call flow between a daytime and a night time call flow. Time Conditions do relate to a call flow module but the difference is Time Conditions is set to automatically change the call flow based on the Time Rules set in the Time Conditions module. The Call Flow module sets a daytime and nighttime call flow that can be manually toggle using the feature code.


  • Click the Create Speed Dial Number button

  • Call Flow Code


    • Enter a code to use for enabled and disabling the call flow

    • The code will be used in the format of (call flow feature code) + (call flow code)

    • Call flow feature code is *28



  • Description

    • Enter description to uniquely identify call flow



  • Enabled Mode Announcement

    • Will play selected recording when call flow is enabled



  • Disabled Mode Announcement

    • Will play selected recording when call flow is disabled



  • PIN Code

    • ​Enter a code required by users to authenticate call flow toggle



  • Enabled Mode

    • Defines where the call is sent during enabled call flow



  • Destination

    • Destination for enabled mode will route the call to the specific location within the module destination



  • Disabled Mode

    • Defines where the call is sent during disabled call flow



  • Destination

    • ​Destination for disabled mode will route the call to the specific location within the module destination



  • Click the Submit button to save configuration

Call Recordings

The QuBe Call Recording module sets ability to override any call recording settings in the call flow. This module can be used to specify certain call flows and whether to set override to call recording. You can add this module to any part of the call flow - if you add this module to the beginning part of the call flow you will get more of the active call recorded, but if you add this module towards the end of the call flow you can specify which part of the call flow to be recorded.


  • Click the Create Call Recording button

  • Name

    • Name the recording to identify the file



  • Call Recording

    • Set Call Recording Strategy by selecting one of the following options:

      • “Force” and “Never” override at a higher priority than “Yes” or “No”

      • “Yes” and “No” have equal weight

      • “Force” and “Never” override all settings in other modules

      • “Force” and “Never” have equal weight

      • “Force” and “Never” will override “Yes” or “No,” and “Yes” and “No” will never override a “Force” or “Never”

      • “Don’t Care” will not change any settings in other modules





  • Destination

    • ​Choose the destination the call will go to after it passes the Call Recording module



  • Click the Submit button to save configuration

Conferences

​The QuBe Conference module creates conference rooms to connect multiple callers in one virtual room. Conference rooms can be configured to required an admin and user PIN so that you can limit the callers that can connect to the conference room. You can create multiple conference rooms for different users or callers.


  • Click on the Create Conference button

  • Conference Number

    • Assign Conference number to use to dial into this conference room



  • Conference Description

    • Give the conference a name to identify it



  • User PIN

    • This is optional – This will require caller to enter a password to be a part of this conference

    • This PIN should be different than the Admin PIN



  • Admin PIN

    • This is optional however if “Leader Wait” is in use this is required – This will require the caller to enter a password to be a part of this conference



  • Join Message

    • When joining conference a message to be played. Leave black for no message.



  • Wait for Leader

    • Select Yes or No depending on if you wish to have caller wait until admin user arrives



  • Leader Leave

    • Select Yes or No depending on if admin user leaves all users will be removed from conference



  • Silence Optimization

    • Enabling this option will drastically improve performance and help remove the buildup of background noise from the conference. This option is highly recommended for large conferences, due to its performance improvements.



  • Talker Detection

    • Sets whether or not notifications of when a user begins and ends talking should be sent out as events over AMI. By default, no.



  • Quiet Mode

    • When set to “yes,” enter/leave prompts and user introductions are not played. By default, no.



  • User Count

    • Announces if the number of users in the conference should be announced to the caller. By default, no.



  • Announce Join

    • When enabled, this option prompts the user for their name when entering the conference. After the name is recorded, it will be played as the user enters and exits the conference. By default, no.



  • Music on Hold

    • Sets whether music on hold should be played when only one person is in the conference or when the user is waiting on a marked user to enter the conference. By default, no.



  • Music on Hold Class

    • Sets the music on hold class to use for music on hold.



  • Allow Menu

    • Choose whether to present a menu for users when asterisk key is entered



  • Record Conference

    • Records the conference call starting when the first user enters the room, and ending when the last user exits the room. By default, this option is disabled.



  • Max Members

    • Limits the number of participants for a single conference to a specific number. By default, conferences have no participant limit. After the limit is reached, the conference will be locked until someone leaves. Admin-level users are exempt from this limit and will still be able to join otherwise-locked, because of limit, conferences.



  • Mute on Join

    • Sets if the user should start out muted. By default, no




Admin Menu


  • Drop down menu for admin feature codes

    • Enter Conference

      • The code allows admin key to enter to conference menu



    • Toggle Mute

      • The code allows the toggle mute code to switch from mute on and mute off



    • Exit Conference Menu

      • The code allows admin exit conference menu code allows admin to exit conference menu



    • Decrease listening volume

      • The code decreases the callers listening volume – sound will be quieter



    • Increase listening volume

      • The code increases the callers listening volume – sound will be louder



    • Reset listening volume

      • This code will reset sound level to default



    • Decrease Talking Volume

      • The code decreases the callers talking volume – sound will be quieter to other callers



    • Increase Talking Volume

      • The code increases the callers talking volume – sound will be loud to other callers



    • Reset Talking Volume

      • This code will reset the sound level to default



    • Leave Conference

      • The code allows a user to exit the conference and continue execution in the dial plan



    • Admin Kick Last

      • This code allows an admin to remove the last participant from the conference



    • Admin Toggle Conference Lock

      • The code allows an admin to toggle locking and unlocking the conference



    • Admin Mute All

      • This code allows an admin to mute or unmute all user participants in the conference



    • Participant Count

      • The code announces the number of participants in the conference






User Menu


  • Drop down menu for user settings

    • Enter Conference

      • The code allows user to enter to conference menu



    • Toggle Mute

      • The code allows the toggle mute code to switch from mute on and mute off



    • Exit Conference Menu

      • The code allows user to exit conference menu code allows admin to exit conference menu



    • Decrease listening volume​

      • The code decreases the callers listening volume – sound will be quieter



    • Increase listening volume

      • The code increases the callers listening volume – sound will be louder



    • Reset listening volume

      • This code will reset sound level to default



    • Decrease Talking Volume

      • The code decreases the callers talking volume – sound will be quieter to other callers



    • Increase Talking Volume

      • The code increases the callers talking volume – sound will be loud to other callers



    • Reset Talking Volume

      • The code will reset the sound level to default



    • Leave Conference

      • The code allows a user to exit the conference and continue execution in the dial plan



    • Participant Count​

      • The code announces the number of participants in the conference





DISA

The QuBe DISA module is used to allow external callers to dial into the PBX system and make outbound calls. DISA - Direct Inward System Access This module is helpful to those remote users that need to make calls using the outbound CID of the PBX or to not use cell phone minutes to call outbound.


  • Click the Create DISA button

  • Name

    • Enter a unique name



  • PIN Code

    • Create a PIN for user to use DISA



  • Context

    • Context to use with dialed number. Default context ‘from-disa’ does not allow dialing an external number

    • from-users= ability to send calls outside the system



  • Response Timeout (seconds)

    • Maximum number of seconds call will wait until it hangs up



  • Digit Timeout (seconds)

    • Maximum number of seconds between digit presses when dialing



  • Call Recording

    • ​Enter call recording preference – this will override all other call recording settings



  • Caller ID Name

    • Caller ID name to use for a call from DISA



  • ​Caller ID Number

    • Caller ID number to use for a call from DISA



  • Click the Submit button to save configuration

Directory

The QuBe Directory module walks through how to create and maintain your Directory


  • Click on the Create Directory

  • Directory Name

    • Enter a descriptive name



  • Directory Description

    • Enter a description of the directory



  • Caller ID Prefix

    • This is optional – the prefix will be added to the caller ID when a call goes through the directory



  • Announcement

    • The greeting played when a caller goes through the directory. Leave blank for a default recording.



  • Invalid Retries

    • The number of entries a caller can retry input after dialing an invalid or non-existent entry



  • Invalid Retry Recording​

    • Recording to be played when the number of invalid retries is exceeded



  • Invalid Destination

    • Type of destination to send call when number of invalid retries is exceeded​



  • Destination

    • Specific destination calls are sent when number of invalid retries is exceeded



  • Return to IVR

    • Select YES or NO to return call to parent IVR



  • Announce Extension

    • Select YES or NO to allow user to be prompted with extension name



  • Add All Users

    • Add all extension users to directory



  • Directory Entries

    • For adding specific users to a directory. Select extensions from a drop-down menu to add them one by one to a directory.



  • Click the Submit button to save configuration 


 

Follow Me

The QuBe Follow me module allows calls to be forwarded to another internal number or external number. You can set the follow me module to have calls received to the extension to skip ringing the extension and go straight to call the number(s) in the follow me list, or you can set the call to ring the extension and the number(s) in the follow me list.


  •  Click the Edit button

  • SIP Device

    • Select the extension user to add Follow Me module settings



  • Enabled

    • Check this box to enable the follow me settings – leave empty if you wish to disable follow me settings



  • Initial Ring Time

    • This is the time to ring the primary extension before ringing the numbers in the follow me list



  • Follow Me List

    • List the extension or phone number you wish to ring when the primary extension is dialed. Important: phone number entries must be entered as 1XXXXXXXXXX# (with X representing a digit)



  • Ring Time

    • Enter the number of seconds to ring the numbers in the follow me list



  • Ring Strategy

    • ​Ringall: This will call all parties in the follow me list simultaneously

    • Ringall2: This will be the same ring strategy as ringall, however it will include the primary extension

    • Hunt: This will call the first number in the list for the duration of ring time, and then ring the first number and the second number in the list for the duration of the ring time, and then the first number, second number, third number for the duration of the ring time, etc.

    • Hunt2: Same as the hunt strategy, but with the primary extension included in the list​



  • Play Music on Hold​

    • If box is checked it will enable Music on Hold to play to the caller while the PBX dials the follow me list based on the ring strategy selected. Leave empty to disable music on hold in favor of ringing by default​



  • CID Prefix

    • ​This will prepend the call that goes through this follow me list with the prefix entered



  • ​CID Forward Mode

    • ​Select the method to send caller’s original CID when handled in the follow me module



  • Fixed CID

    • ​If Fixed CID is selected in the CID Forward Mode then enter the fixed CID here



  • ​No Answer

    • Select the type of destination to send the call if there is no answer​



  • Destination

    • Select the specific destination to send the call if there is no answer



  • Click the Submit button to save configuration

IVR

The QuBe IVR module is used for callers to dial an option from the list available


  •  Click the Create IVR button

  • IVR Name

    • Create a name for the IVR



  • Description

    • This is optional, to help identify the IVR



  • Allow Direct Dial

    • Select YES or NO to enable direct dial. If YES is selected, caller will have ability to dial an extension directly without using the IVR options



  • IVR Extension

    • Assign an extension to the IVR to allow internal access



  • Announcement

    • Choose the greeting to play when a caller reaches the IVR



  • Timeout

    • The number of seconds the PBX should wait without input before moving to the timeout destination



  • Timeout Digits

    • The number of seconds the PBX should wait between keypresses before assuming timeout



  • Invalid Retries

    • The number of times the caller can retry input after entering an invalid entry



  • Invalid Retry Recording

    • The recording to be played after an invalid entry



  • Append Announcement to Invalid

    • Select YES or NO to repeat the IVR announcement after invalid input. If YES the system will play the IVR announcement after the invalid retry recording.



  • Return on Invalid

    • Select YES or NO to return caller to previous parent menu. Only if caller has entered a sub-IVR



  • Invalid Recording

    • Recording to be played when invalid retries are exceeded



  • Invalid Destination

    • Type of destination to send calls if invalid retries are exceeded



  • Destination

    • Specific destination to send calls if invalid retries are exceeded



  • Timeout Retries

    • The amount of times the caller is able to timeout before call is sent to timeout destination



  • Timeout Retry Recording

    • The recording to play when a caller exceeds the limit set in timeout



  • Append Announcement to Timeout

    • Select YES or NO to repeat the IVR announcement after timeout. If YES the system will play the IVR announcement after the timeout recording.



  • Return on Timeout

    • Select YES or NO to return caller to previous parent menu. Only if caller has entered a sub-IVR



  • Timeout Recording

    • Recording to be played when timeout limit exceeded



  • Timeout Destination

    • Type of destination to send calls if timeout limit is exceeded



  • Destination

    • Specific destination to send calls if timeout limit is exceededIVR EntriesSelect entries one-by-one to add to the IVR menu



  • Visualize

    • After the IVR is created you can view the entire call flow using the visualize button



  • Click the Submit button to save configuration 


 

Misc. Destinations

The QuBe Misc Destinations module can be used to create a route destination to send calls to another location, whether the location is internal or external. You can use this module to have calls go to an external number such as a cell phone, or this module can be used to create a destination to use a feature code. Such as, if users need to access their voicemail box from an external phone you could set the Misc Destination as *98. Then set up an IVR with option # to call the Misc Destination - Voicemail which will allow the caller to dial into the voicemail box.


  • Click the Create Misc Destination button

  • Name

    • Enter a descriptive name



  • Number to Dial

    • Enter the extension, telephone number, or feature code to dial when the miscellaneous destination is used



  • CID Mode

    • Select the caller ID mode to use



  • Click the Submit button to save configuration

Voicemail

The QuBe Voicemail module allows you to manage all basic configurations of voicemail for individual extensions. Enable voicemail messages to go to email along with changing the voicemail account password.


  • Edit voicemail settings or view voicemail settings by clicking the edit button next to the corresponding extension

  • Enable

    • Check box to enable voicemail settings on extensionPassword Required From Same ExtensionCheck the box to require the user to enter their password when *97 is dialedThis option does not apply to *98



  • Email

    • Enter email address to receive voicemail message notifications to email



  • Email Attach

    • Check box to attach .wav voicemail message files to the email address entered above



  • Play CID

    • Enable to read back telephone number before playing voicemail



  • Play Envelope

    • ​Enable to play date and time before playing voicemail



  • Delete Listened

    • ​Enable to remove already listened voicemails from PBX

    • If ENABLED it is highly recommended to set Email and Email Attachment to ENABLED to avoid inadventently losing voicemails from the PBX permanently



  • VM Options

    • This is optional – enter voicemail options, separated by the pipe symbol |



  • VM Context​

    • This is voicemail context​

    • It is highly recommended to not make changes to this unless you already understand this function



  • Click the Submit button to save configuration

To use the Gmail account as relay for sending out Voicemail emails.


  • Allow less secure apps must be turned on in the gmail account.

    • yum install postfix mailx cyrus-sasl cyrus-sasl-plain

    • vi /etc/postfix/sasl_passwd



  • Add the following line according to your account settings:

    • :587

    • smtp_use_tls = yes

    • smtp_sasl_auth_enable = yes

    • smtp_sasl_security_options =

    • smtp_sasl_password_maps = hash:/etc/postfix/sasl_passwd

    • smtp_tls_CAfile = /etc/ssl/certs/ca-bundle.crt



  • Save and Close

    • postmap /etc/postfix/sasl_passwd

    • systemctl restart postfix.service



  • To Test

    • mail -s “Test subject” recipient@domain.com



  • You will be presented with a blank line for the CC field press enter. Type the body of your message type CTRL-D to send it.

Paging

The QuBe Paging module creates an extension number to dial with a group of internal users. When the extension number is dialed an intercom call is placed to all extensions in the paging group.


  • Click the Create Page Group + button

  • Enabled

    • Check this to enable the paging group



  • Page Group Extension

    • Assign a unique extension number so users can dial paging group internally



  • Description

    • Provide a description to identify the paging group



  • Page Group List

    • Enter the extension numbers to be called when the paging group is dialed



  • Ring Type

    • Intercom – One ring followed by automatic answer

    • Intercom with immediate answer – No ringing

    • Normal ringing until caller answers



  • Time For Normal Ring

    • Enter the time in seconds for ring time



  • ​Announcement​

    • Choose an announcement to play to callee



  • Skip Busy Extensions​

    • If user in page group list is on a call, incoming page will skip extension



  • Duplex

    • ​Select if all parties are muted or all parties are unmuted



  • CID Prefix

    • Add prefix to users receiving incoming calls from the page group



  • Click the Submit button to save configuration

Parking

Click here to edit.
The QuBe Parking module is used to as a "virtual" hold. When parking buttons are configured on the phone it allows one user to put a call on hold in the parking lot and for another user to pick up the call on hold in the parking lot. The user that placed the call on the parking lot can announce or call to another user who will pick up the call that there is a call waiting on parking button 71. The user will pick up the call by hitting the 71 park button on the phone. The parking buttons will need to be configured on the phone by user preference. The slots can range from one slot to multiple slots / buttons on the phone.


  • Click the Create Time Rules button

  • Description

    • Enter a description to identify the time rule




 

Non-Periodical Time Rules

Non-periodical rules will override periodical rules. Non-periodical rules are used for holiday dates or last-minute closures.


  • Date and Time to Start

    • Enter the date and time to start the time rule



  • Date and Time to Finish

    • Enter the date and time to finish the time rule




 

Periodical Time Rules

Periodical rules are used for daily hour time set and will be overridden by non-periodical rules.


  • Start time

    • Enter the time to begin the time rule



  • End Time

    • Enter the time to end the time rule



  • Week Days

    • Enter the week day to apply the start time and end time periodical time rule



  • Click the Submit button to save configuration 


Additional Parking

To create a new parking lot, go to Functions --> Parking. Then you just click "Create Parking Lot", and fill out the necessary settings. (making sure to name it something other than default)


After you create a new parking lot, you'll need to go to Connectivity --> Extensions. Once in here, find the extension(s) you wish to use with the said parking lot, click edit, scroll down to Advanced Settings and add a "parkinglot" with the parking lot name in "value". 
IMPORTANT: The parking lot selected (in the "value" section of advanced settings) must be all lowercase no matter the capitalization of the parking lot created.

Queues

The QuBe Users module creates administrators to login to the PBX


  • Click on Create Queue button

  • Name

    • Enter a name to identify the queue



  • Queue Extension

    • Assign extension number to dial this queue



  • Music Class

    • Set which music class to play for this queue



  • Announcement before Enter Queue

    • Select an announcement to play to caller before entering queue



  • Queue Strategy

    • Ring strategy to ring queue members

    • Ring all available members

    • Ring member which was least recently hung up by this queue

    • Ring the one with the fewest completed calls from this queue

    • Ring random member

    • Round robin with memory – Remember where call was left off last ring pass

    • Same as RRMemory, except the queue member order from config

    • Rings members in the order in which they were added

    • Rings Random member, but uses the members penalty as a weight



  • IVR

    • Ability to allow caller to use selected IVR as options – IVR must contain single digit options only



  • Auto Fill Free Members

    • Autofill will use queue strategy but allow multiple calls through at the same time until there is no calls waiting or no more available queue members



  • Maximum Queue Callers

    • Set the maximum amount of people who can wait in queue (Set 0 for unlimited)



  • Queue Weight – Higher More Important

    • Weight of queue when compared to other queues – the higher queue weight allows for queue to get available channels when the same channel is included in more than one queue



  • Service Level Time in Seconds

    • Used for service level statistics (Calls answered within service level time frame)

    • Penalty Members Limit

      • ​Limit to be set to disregard penalty settings when the queue has too few members – No penalty will be weighed if there are only X or fewer queue members (Default is 0)





  • Default Rule

    • If you wish to implement rule Timing Options




Timing Options


  • General Queue Timeout

    • Timeout represents the amount of time in seconds caller can stay in queue before caller is removed from queue



  • Timeout Before Ring Next Member

    • Timeout specific amount of time in seconds to try to ring queue member before considering queue member unavailable



  • Wait Until Retry Timeout

    • Amount of time in seconds system waits before trying all queue members again



  • Timeout Behavior

    • Select which timeout has higher priority



  • Time After Call to Accept Next Caller

    • Amount if time in seconds to wait to send call to free member (Default is 0 / No delay) 



  • Member Pause Behavior

  • Pause a Queue Member if they Fail to Answer a Call

    • Autopause will pause a queue member if they fail to answer a call



  • Delay Pause Since Last Successful Call

    • This will delay autopause of a queue member by number of seconds after taking a call



  • Pause Member if Busy Outside Queue

    • Autopause busy controls if queue member will be set to paused automatically if member’s device reports busy (Default set to no)



  • Pause Member if Unavailable

    • Autopause unavailable controls if queue member will be set to paused automatically if member’s device reports unavailable (Default set to no)




Periodic Announcements


  • Periodic Announcements – Played by File

    • Periodic announcements played by file



  • Periodic Announcements Frequency

    • Set how often to announce recording by time in seconds



  • Play Periodic Announcements in Random Order

    • Select YES or NO to set random order of announcements



  • Relative Periodic Announcements

    • If yes, periodic announcement frequency will be timed from end of announcement rather than the start of each announcement 




Announce Hold Time and Position to Caller


  • Announce Queue Position Frequency

    • The amount of time in seconds to announce queue position and / or estimated hold time



  • Minimum Time Between the Start of Each Queue Position

    • The minimum amount of time between the start of each queue position and / or estimated hold time announcement



  • Announce Hold time

    • Select YES or NO to include hold time in position announcements



  • Announce Position

    • This will announce position to caller



  • Announce Position Limit

    • If you have specified “limit” or “more” for the announce-position option, then the following value is what is used to determine what announcement to play to waiting callers. If you have set the announce-position option to anything else, then this will have no bearing on queue operation



  • Announce to First User

    • If set to yes, it will play the announcement to the first user waiting in the queue. This may result in an announcement played when a member attempts to connect to the waiting caller, which may delay the time before an agent and user begin to communicate



  • Round Announce Frequency to

    • Announce by the seconds as well as the minutes



  • Record “Thank you for your Patience”

  • Record “You are first in line”

  • Record “There are”

  • Record “Calls Waiting”

  • Record “The current Est. Hold Time is”

  • Record “Minute”

  • Record “Minutes”

  • Record “Seconds”

  • Record “Hold time” 


Announce Member


  • Announcement to Member when Caller is Connected

    • Announcement to play to member when the call is connected



  • Report User Hold Time to Member

    • Set this to yes if you wish to report caller’s hold time to member before they connect 




Record Option


  • Monitor Format

    • Format to record queue calls



  • Monitor Type

    • Type of recording queue calls




 Queue Empty Options and Busy In / User Handling


  • Join Empty

    • The “joinempty” option controls whether a caller may join a queue depending on several factors of member availability



  • Leave Empty

    • The “leavewhenempty” option controls whether a caller may remain in a queue he has already joined



  • Ring In Use Members

    • Select YES or NO if you want the queue to avoid sending calls to members whose devices are known to be unavailable



  • Connect Delay

    • If you wish to have a delay before the member is connected to the caller set the number of seconds to delay



  • Restart Timeout if Call was Rejected

    • If this is set to yes, the timeout for an agent to answer is reset if a busy or congestion is received




 Environment Options


  • Set Interface Variables

    • If set to yes, just prior to the caller being bridged with a queue member



  • Set Queue Entry Variables

    • If set to yes, just prior to the caller being bridged with a queue member



  • Set Queue Variables

    • If set to yes, the following variable will be set ust prior to the caller being bridged with a queue member and just prior to the caller leaving the queue



  • Run Macro When Member Connected

    • If set, run this macro when connected to the queue member you can override this macro by setting the macro option on the queue application



  • Run Macro When Member Connected

    • If set, run this gosub when connected to the queue member you can override this gosub by setting the gosub option on the queue application 




Queue Members


  • User

    • Select the user / extension number that will be included in this queue



  • Dynamic or Static

    • Select Dynamic if user will be able to login in and out of queue, or select Static if user will always remain in queue



  • Penalty

  • Login Name

  • Login PIN

  • ​​Click the Submit button to save configuration

Ring Group

The QuBe Ring Group module is used as a destination within the call flow to allow multiple extensions to ring and have the ability to pick up the incoming call. The ring group features include the ability to list internal extensions as well as external numbers in the follow me list. All numbers in the follow me list will ring depending how you select the ring strategy. The ring group is assigned an internal extension number for the ability to dial if a caller wants to reach one of the users in the follow me list.


  • Click the Create Ring Group button

  • Enabled

    • Check the enabled box



  • Ring Group Extension

    • Assign a unique extension number to dial this ring group



  • Ring Group List

    • Enter a list of extensions and/or phone numbers to ring when caller goes to ring group



  • Ring Strategy

    • Ring All – Call all parties in the list simultaneously

    • Hunt – Call the first party in the list; then the first and second; then the first, second, and third, etc.



  • Ring Time

    • Set the amount of time in seconds to ring the ring group list numbers



  • Announcement

    • ​Select an announcement to play to the caller before beginning to ring the ring group list



  • Play Music on Hold

    • Check this box if you want to enable music on hold



  • Music on Hold Classe​

    • Select the file to play if music on hold is enabled



  • CID Prefix

    • ​Add a CID prefix to prepend to the original caller ID



  • CID Forward Mode

    • ​This decides how to handle caller ID with ring group calls



  • Fixed CID​

    • This allows setting a fixed CID value



  • No Answer

    • Select the type of destination to route the call if no answer



  • Destination​

    • Select the specific destination to route the call if no answer



  • Click the Submit button to save configuration 

Time Conditions

The QuBe Time Conditions is built to control the call flow of the call based on the time range built in time rules. When an inbound call is set to time conditions as destination the system checks the time rule in the time condition, confirms the current time against the time rule, and based on the match it routes call to either destination in the time condition.


  • Click the Create Time Condition button

  • Description

    • Enter a description to identify this time condition



  • Time Rules

    • Select the time rule to apply to this time condition



  • Time Zone (Blank for Local)

    • Select the time zone



  • Destination Matches

    • The type of destination to send the call to if the time conditions match



  • Destination

    • The specific destination to send the call if the time conditions match



  • Destination Non-Matches

    • The type of destination to send the call if the time conditions do not match​



  • Destination

    • The specific destination to send the call if the time conditions do not match



  • Click the Submit button to save configuration

Time Rules

The QuBe Parking module is used to as a "virtual" hold. When parking buttons are configured on the phone it allows one user to put a call on hold in the parking lot and for another user to pick up the call on hold in the parking lot. The user that placed the call on the parking lot can announce or call to another user who will pick up the call that there is a call waiting on parking button 71. The user will pick up the call by hitting the 71 park button on the phone. The parking buttons will need to be configured on the phone by user preference. The slots can range from one slot to multiple slots / buttons on the phone.


  • Click the Create Time Condition button

  • Description

    • Enter a description to identify the time rule




Non-Periodical Time Rules

Non-periodical rules will override periodical rules. Non-periodical rules are used for holiday dates or last-minute closures.


  • Date and Time to Start

    • Enter the date and time to start the time rule



  • Date and Time to Finish

    • Enter the date and time to finish the time rule




Periodical Time Rules

Periodical rules are used for daily hour time set and will be overridden by non-periodical rules.


  • Start time

    • Enter the time to begin the time rule



  • End Time

    • Enter the time to end the time rule



  • Week Days

    • Enter the week day to apply the start time and end time periodical time rule



  • Click the Submit button to save configuration

Voicemail Blasting

The QuBe Voicemail Blasting module is used to send one voicemail message to multiple users. When creating a voicemail blast group you can create a group of users for departments or any specifics you choose and have that group receive the voicemail. There are multiple voicemail blast groups that can be created.


  • Click the Create VMBlast Group button

  • VMBlast Group Extension

    • Enter an extension to reach the voicemail blast group



  • Description

    • Enter a description to identify the voicemail blast group



  • Default

    • Check the box to make this voicemail blast group the default



  • Group List

    • Select one or more extensions to dial when the voicemail blast group is called



  • Password

    • This is optional – create a password to secure the voicemail blast group



  • Intro Audio

    • Select audio to play as the introduction

      • Say Number

      • Beep

      • Announcement





  • Announcement

    • ​If announcement is selected choose the announcement to play as the introduction audio



  • Click the Submit button to save configuration

Wake Up

The QuBe Wake up module was built for an alarm type call to come to an extension. You can create the settings to configure the wake up calls to your specifications.


  • Click the Create Wake Up Call button

  • Destination

    • Enter the destination to go when this wake up call is enabled



  • Description

    • Enter the description to identify the wake up call



  • Time

    • Enter the time to send the wake up call



  • Click the Submit button to save configuration

Channel SIP Settings

The QuBe Channel SIP Settings module allows you to customize the way your audio is handled as it is passed to and from the PBX through your trunk.


  • Default Context

    • Default context for incoming calls. Defaults to ‘from-sip-public’



  • Auto reject non-authorized request

    • When an incoming INVITE or REGISTER is to be rejected, for any reason, always reject with an identical response equivalent to valid username and invalid password instead of letting the requester know whether there was a matching user or peer for their request. This reduces the ability of an attacker to scan for valid SIP usernames. This option is set to “yes” by default.



  • Allow or reject guest calls

    • Allow or reject guest calls (default is no). If your Asterisk is connected to the Internet and you have allowguest=yes you want to check which services you offer everyone out there, by enabling them in the default context (see below)



  • Bind IP address

    • IP address to bind listen socket to (0.0.0.0 binds to all)



  • TCP Bind IP address

    • If TCP enabled. IP address to bind TCP listen socket to (0.0.0.0 binds to all)



  • External IP address

    • IP address and port number to be used when talking to a host outside the NAT



  • DNS SRV lookups on outbound calls

    • Enable DNS SRV lookups on outbound calls. Note: Asterisk only uses the first host in SRV records. Disabling DNS SRV lookups disables the ability to place SIP calls based on domain names to some other SIP users on the Internet. Specifying a port in a SIP peer definition or when dialing outbound calls will supress SRV lookups for that peer or call



  • Allow Call Events

    • Set to yes to receive events on AMI when a call is put on/off hold




Local Networks


  • Network Address

    • Enter internal network address



  • Network Mask

    • Enter network mask address




NAT and Media Settings


  • NAT

    • Asterisk may override the address/port information specified in the SIP/SDP messages, and use the information (sender address) supplied by the network stack instead. However, this is only useful if the external traffic can reach us. The following settings are allowed (both globally and in individual sections): ‘no’- Do no special NAT handling other than RFC3581; ‘force_rport’ – Pretend there was an rport parameter even if there wasn’t, ‘comedia’ – Send media to the port Asterisk received it from regardless of where the SDP says to send it. ‘auto_force_rport’ – Set the force_rport option if Asterisk detects NAT (default). ‘auto_comedia’ – Set the comedia option if Asterisk detects NAT



  • Direct Media

    • By default, Asterisk tries to re-invite media streams to an optimal path. If there’s no reason for Asterisk to stay in the media path, the media will be redirected. This does not really work well in the case where Asterisk is outside and the clients are on the inside of a NAT. In that case, you want to set directmedia=nonat. ‘yes’ – Asterisk by default tries to redirect the RTP media stream to go directly from the caller to the callee. Some devices do not support this (especially if one of them is behind a NAT). The default setting is YES. If you have all clients behind a NAT, or for some other reason want Asterisk to stay in the audio path, you may want to turn this off. This setting also affects direct RTP at call setup (a new feature in 1.4 – setting up the call directly between the endpoints instead of sending a re-INVITE). Additionally this option does not disable all reINVITE operations. It only controls Asterisk generating reINVITEs for the specific purpose of setting up a direct media path. If a reINVITE is needed to switch a media stream to inactive (when placed on hold) or to T.38, it will still be done, regardless of this setting. Note that direct T.38 is not supported. ‘nonat’ – An additional option is to allow media path redirection (reinvite) but only when the peer where the media is being sent is known to not be behind a NAT (as the RTP core can determine it based on the apparent IP address the media arrives from). ‘update’ – Yet a third option… use UPDATE for media path redirection, instead of INVITE. This can be combined with ‘nonat’, as ‘directmedia=update,nonat’. It implies ‘yes’. ‘outgoing’ – When sending directmedia reinvites, do not send an immediate reinvite on an incoming call leg. This option is useful when peered with another SIP user agent that is known to send immediate direct media reinvites upon call establishment. Setting the option in this situation helps to prevent potential glares. Setting this option implies ‘yes’. ‘yes’ – Enable the new experimental direct RTP setup. This sets up the call directly with media peer-2-peer without re-invites. Will not work for video and cases where the callee sends RTP payloads and fmtp headers in the 200 OK that does not match the callers INVITE. This will also fail if directmedia is enabled when the device is actually behind NAT



  • RTP Timeout

    • Terminate call if defined seconds of no RTP or RTCP activity on the audio channel when we’re not on hold. This is to be able to hangup a call in the case of a phone disappearing from the net, like a power loss or grandma tripping over a cable



  • RTP Hold Timeout

    • Terminate call if 300 seconds of no RTP or RTCP activity on the audio channel when we’re on hold (must be > rtptimeout)



  • RTP Keep alive

    • Send keep alives in the RTP stream to keep NAT open (default is off – zero)



  • Enable Jitter Buffer

    • Enable the use of a jitterbuffer on the receiving side of a SIP channel. Default set to “no”. An enabled jitterbuffer will be used only if the sending side can create and the receiving side can not accept jitter. The SIP channel can accept jitter, thus a jitterbuffer on the receive SIP side will be used only if it is forced and enabled




Audio Codecs​


  • ulaw

  • alaw

  • gsm

  • g726

  • g726aal2

  • adpcm

  • slin

  • lpc10

  • speex

  • ilbc

  • g722

  • testlaw


RTP Settings


  • Start RTP Port

    • Enter the RTP port to start



  • End RTP Port

    • Enter the RTP port to end



  • UDP checksum on RTP

    • Whether to enable or disable UDP checksums on RTP traffic



  • No “end” DTMF timeout

    • The amount of time a DTMF digit with no ‘end’ marker should be allowed to continue (in ‘samples’, 1/8000 of a second)



  • Time between rtcp reports (ms)

    • Milliseconds between rtcp reports(min 500, max 60000, default 5000)



  • Enable strict RTP protection

    • Enable strict RTP protection. This will drop RTP packets that do not come from the source of the RTP stream. This option is enabled by default



  • RTP sockets probation packets

    • Number of packets containing consecutive sequence values needed to change the RTP source socket address. This option only comes into play while using strictrtp=yes. Consider changing this value if rtp packets are dropped from one or both ends after a call is connected. This option is set to 4 by default



  • ICE Support

    • Whether to enable or disable ICE support. This option is disabled by default



  • STUN server

    • Hostname or address for the STUN server used when determining the external IP address and port an RTP session can be reached at. The port number is optional. If omitted the default value of 3478 will be used. This option is disabled by default. e.g. stundaddr=mystun.server.com:3478



  • TURN server

    • Hostname or address for the TURN server to be used as a relay. The port number is optional. If omitted the default value of 3478 will be used. This option is disabled by default. e.g. turnaddr=myturn.server.com:34780



  • TURN server username

    • Username used to authenticate with TURN relay server



  • TURN server password

    • Password used to authenticate with TURN relay server




Audio and Video Settings


  • Non-standard g726

    • If the peer negotiates G726-32 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-G726-32 instead



  • T38 Fax

    • This setting is available in the section as well as in device configurations. Setting this to yes enables T.38 FAX (UDPTL) on SIP calls; it defaults to off



  • Video Calls Support

    • Whether to enable or disable Video Calls support. This option is disabled by default




Video Codecs


  • h264

  • mpeg4

  • vp8

  • h263p

  • h261

  • h263


Notifications


  • Notify Ringing

    • Control whether subscriptions already INUSE get sent RINGING when another call is sent (default: yes). NOTE! This feature is used by panel



  • Notify Hold

    • Notify subscriptions on HOLD state (default: no). Turning on notifyringing and notifyhold will add a lot more database transactions if you are using realtime. NOTE! This feature is used by panel




Registration Settings


  • Registration Timeout

    • Retry registration calls timeout (default: 20)



  • Registration Attempts

    • Number of registration attempts before we give up 0 = continue forever, hammering the other server until it accepts the registration Default is 0 tries, continue forever



  • Registration Maximum Expiry

    • Maximum allowed time of incoming registrations (seconds)



  • Registration Minimum Expiry

    • Minimum length of registrations (default 60)



  • Registration Default Expiry

    • Default length of incoming/outgoing registration



  • SIP Advanced Settings​

    • Set any option here – all options defined here will override options set above



  • Click the Submit button to save configuration

  • Text Hover

Music on hold

The QuBe Music on Hold allows you to easily upload personal Music on Hold files to be played on the PBX. Music on Hold can be uploaded to the inbound route module, the ring group module, the queue module, and conference module.


  • Click the Create Music on Hold button

  • Name

    • Enter a Music on Hold class name



  • Mode

    • Select the Music on Hold mode​

      • Files: Play files from class directory

      • Custom: Use custom application defined by application string





  • Click the Submit button to save configuration

Firewall

The QuBe Firewall module was built to be able to manage the PBX firewall settings from the GUI. The firewall module removes having to access the command line to view and edit the firewall settings.

Firewall Status


  • Click on the Firewall Status button to view current IPTABLES


 

Trusted Interfaces


  • ​View the trusted interfaces and untrust interfaces​Interface

  • Shown is the interface connected to the server?

  • Status

  • Status will show if interface is trusted or untrusted​

  • Action

  • This will allow you to trust or untrust interface


 

Trusted Networks


  • ​​​Select the Add New button to add a trusted network IP

  • Click the Reload Rules button to reload the rules applied and saved at any time

  • Description

    • Provide a unique description to identify trusted network



  • IP address

    • Enter IP address of trusted network



  • Netmask

    • Enter netmask of trusted network




 

Trusted Domains


  • ​​​​​Select the Add New button to add a trusted domain

  • Click the Reload Rules button to reload the rules applied and saved at any time

  • ​Description

    • Provide a unique description to identify trusted domain



  • Domain name​

    • Enter Domain Name of trusted domain





External Devices


  • Select the Add New button to add a trusted network IP

  • Click the Reload Rules button to reload the rules applied and saved at any time

  • Description

    • Provide a unique description to identify external device



  • IP address

    • Enter IP address of external device



  • Netmask

    • Enter Netmask of external device​




 

Trusted Service Users


  • Select the Add New button to add a trusted network IP

  • Click the Reload Rules button to reload the rules applied and saved at any time

  • Description

    • Provide a unique description to identify trusted service user​



  • Service

    • Select service of trusted service user



  • IP address

    • Enter IP address of trusted service user



  • Netmask

    • Enter Netmask of trusted service user​​



  • Click the Submit button to save configuration

  • Text Hover

FAIL2BAN

The QuBe Fail2Ban module allows you to ignore IP addresses to allow access to the PBX. The module displays in the front page the amount of jailed IP addresses that have been banned totally and that are currently banned.


  • The Fail2Ban Stats page displays any jailed IP addresses

  • Create Fail2Ban Ignore Address

    • This button is located on the page you get after clicking the Fail2Ban Ignore List



  • IP Address

    • Enter IP address of network to add to Fail2Ban ignore



  • Netmask

    • Enter netmask of network to add to Fail2Ban ignore



  • Click the Submit button to save configuration

Back up and restore

The QuBe Backups and Restore module is used to create backups of the PBX and to upload backups onto the PBX to restore. The backup settings allow you to create backups on a schedule and customize what you are backing up to save. For a restore you can upload a backup file to the PBX from this module.

Backup Schedule


  • ​Enabled

    • Check the box to enable a backup schedule



  • Save Copy at FTP Server

    • Check the box to save a backup copy to an FTP server



  • Data

    • Select the data to include in the backup from the drop down list



  • Backup Period

    • Select whether to perform daily or weekly backup from the drop down list



  • Save Last

    • Enter the number of unique backups to save




 

FTP Server Settings


  • Enabled

    • Check the box to enable FTP server communication



  • FTP Server Address

    • Enter your FTP server address



  • Auth Username

    • Enter the authentication username to connect to the FTP server



  • Auth Password

    • Enter the authentication password to connect to the FTP server



  • Path

    • Enter the filepath to navigate to after connecting to the FTP server



  • Activate Passive Mode

    • Check the box to activate passive mode




 

Restore From File


  • Backup File

    • Click the Choose File button to upload a backup to restore this PBX




 

 

Create Backup


  • Data

    • ​From the drop down menu select the data to include in the backup file



  • Save Copy at FTP Server

    • Check the box to save a copy of the backup to the FTP server



  • Click the Submit button to save configuration

Voicemail Settings

The QuBe voicemail module walks you through how to setup voicemail settings


  • From Email

    • Set the sender email address



  • Attach File to Email

    • Select YES or NO from the drop down menu to attach voicemail .wav file to email



  • From String

    • Change the string in the from field



  • Email Subject

    • Enter the email subject that will display on all outbound emails



  • Email Body

    • Edit or enter the email body to notify the receiver of the new voicemail (512 character limit)



  • Click the Submit button to save configuration

For more information on how to take advantage of the "Speech Recognition" option in this menu, click below (requires access to Google API):

Asterisk Info

The QuBe Asterisk Info module displays the detailed current asterisk information broken down by most commonly used modules. All information displayed in the Asterisk Info module can also be found in the command line. Below is a list of the sub categories displayed when you select a given category.


  • Asterisk Info

    • Version

    • Uptime

    • Channels



  • Click on SIP Info button

    • Peers

    • Channels

    • Channel Stats

    • Registries

    • In Use

    • Subscriptions



  • Queues Info

    • Queues



  • Conference Info

    • Conferences



  • Voicemail Info

    • Voicemail Boxes



  • Subscription Info​

    • Hints



CDR Reports

The QuBe CDR Reports module is used to search through all incoming and outgoing calls in the PBX. The CDR reports show very detailed responsed as well as the call events. If extensions or part of the call flow is set to record the call you will be able to listen and download the recordings in the call detailed record / unique ID page.

Records Filter


  • Date

    • Enter the date from and date to for records filter



  • Source Number

    • Enter the source number for records filter



  • Source name

    • Enter the source name for records filter



  • Destination Number

    • Enter the destination number or outbound number called for records filter



  • Account Code

    • Enter the account code for records filter



  • Disposition

    • Select the final result of call or disposition for records filter



  • Duration

    • Enter the duration for records filter 




CDR Report


  • Call Date

    • The time and date stamp of the call



  • Unique ID Caller ID

    • The unique ID shows the Call Record in full, Queue Events if call entered a queue, and Channel events

    • The caller ID sent to the PBX



  • Source Number

    • The number the call originated from



  • Destination Number

    • The number the call terminated at



  • Last Application

    • The last destination of where the call ended



  • Duration

    • The total duration of the call



  • Billing Duration

    • The duration of the call that was billed



  • Disposition

    • The result of where the call was accepted or denied



  • Account Code

    • The account code is used to tag an extension for later identification in the CDR



  • ​Call Record

    • View the call record when you click on the unique ID on the call record



  • Queue Events

    • If call entered a queue then queue events will display in the unique ID call record – note, not all calls will show queue events only the calls that entered a queue

      • Call ID

      • Call Date

      • Queue Name

      • Agent

      • Event

      • ​Agent Name





  • Channel Events

    • The log / event of the call while active in the PBX

      • Event Type

      • Event Time

      • Userdeftype

      • CID_Name

      • CID_Number

      • CID_ANI

      • CID_RDNIS

      • CID_DNID

      • Extension

      • Context

      • CHAN Name

      • App Name

      • App Data

      • Unique ID

      • User FieldPeer





Logs

​The QuBe Logs is an effective way to view sections of the Asterisk log. This module is best intended for recent events in Asterisk, however you are able to view full Asterisk logs.


  • Click on Asterisk Messages Log button

    • The Asterisk message log will display all recent events



  • Click on Asterisk Full Log button

    • This will display the full Asterisk log



  • Config Generation Log button

    • This will display the config generation log



Recording reports

The QuBe Recording Reports allows recorded calls to be listed in the report for the PBX admin to manage. The report was built to clearly identify recorded calls by record type, time of call, duration, and description. The recording report allows you to play the recordings from the PBX without having to download multiple files. If you choose to download click the download button to save the recording to your computer.


  • File Name

    • The file name the system identifies the recording by



  • Record Type

    • The particular module this recording was tracking



  • Record Time

    • The time stamp of when the recording began



  • Duration

    • The total amount of time the recording lasts



  • Description

    • Information about where the call came from and where it went



  • Play Record

    • Listen to the recording, see the duration, and adjust the volume (Fig. 1)



  • You can also download the recording by using the icon on the last column

Queue calls report

The QuBe Queue Calls Reports module is used to search through all incoming calls in the PBX that have gone through the queue module. You can break down calls by date and DID number just like you can in the CDR Reports module however in the Queues Calls Report module you can also view the call reports by queue, as well as other specifications.

Records Filter


  • ​Date

    • Enter the date from and date to for records filter



  • Queue

    • Select one or multiple queues to view in the records



  • Agent Number

    • Select one or multiple agent numbers to view in the records



  • Agent Name

    • Select one or multiple agent names to view in the records



  • Caller Number

    • Enter the full 10 digit caller number to view in the records



  • DID Number

    • ​Select one or multiple DID numbers



  • End Reason

    • Select one or multiple reasons for why the call ended to view in the records




Report


  • ​Call ID

    • View unique ID under Call Detail Record



  • Call Date

    • Time stamp of call



  • Queue

    • The queue name of the call record



  • Agent

    • The agent number that handled the call



  • Agent Name

    • The agent name that handled the call



  • End Reason

    • The reason the call ended



  • Caller ID

    • The caller ID of the outbound call



  • DID Number

    • The DID of the call



  • Position

    • The last position of the call



  • Start Position

    • The position of when the call began



  • Wait Time

    • The total amount of time before the caller was connected to an agent



  • Hold Time

    • The total amount of time the caller was placed on hold



  • Ring Time

    • The total amount of ring time before the caller was connected to an agent



  • Call Time

    • The total amount of time elapsed before the call ended



  • Transfer To

    • The location the call was transferred to



  • IVR Used

    • The IVR the call came from, and was transferred to 



Queue Analytics

The QuBe Queue Calls Analytics module is used to view the statistics of the calls that have entered a queue in the PBX. The statistics are broken down by successful calls and failed calls as well as queue and agent statistics. The purpose of this module is to manage call centers or offices with many employees that use the queue to manage call flows and incoming callers.

Calls



  • Records Filter


  • Date

    • Enter the date from and date to for results



  • Queue

    • Select one or multiple queues for results



  • ​DID Number

    • Select one or multiple DIDs for results




  • Total Calls


  • Success

    • Listed under quantity is the total number of successful calls



  • Failed

    • Listed under quantity is the total number of failed calls



  • Total

    • Listed under quantity is the total number of calls




  • Successful Calls


  • Blind Transfer

    • Listed under quantity is the total number of blind transfer calls



  • Complete Agent

    • Listed under quantity is the total number of completed calls by agent



  • Complete Caller

    • Listed under quantity is the total number of completed calls by caller




  • Failed Calls


  • Abandon

    • Listed under quantity is the total number of abandoned calls



  • Exit with timeout

    • Listed under quantity is the total number of calls exited without timeout



  • Daily Statistic

    • As shown in column chart form, the total number of calls completed daily

    • As shown in pie chart form, the total number of calls failed and succeeded



  • Hourly Statistic

    • As shown in column chart form, the total number of calls completed hourly

    • As shown in pie chart form, the total number of calls failed and succeeded​




Successful Calls



  • Records Filter


  • Date

    • Enter the date from and date to for results



  • Queue

    • Select one or multiple queues for results



  • DID Number

    • Select one or multiple DIDs for results




  • Total Succeeded


  • Complete Caller

    • The number of succeeded calls that was completed with caller



  • Complete Agent

    • ​The number of succeeded calls that was completed with agent



  • Exit with Key​

    •  



  • Attended Transfer

    • When both you and the caller on on the line as the additional party is added to the call as well. 



  • ​Transfer

    • When you place the caller on a “hold” while you’re talking to the new party letting them know that they will be receiving a transfer



  • ​Blind Transfer​

    • When you do not alert the new party you are transffering the caller to that they are receiveing a transfer




  • Average Wait Time


  • Average Wait Time

    • The average wait time for successful calls



  • Average Ring Time

    • The average amount of time the call rings before answered



  • Average Call Time

    • The average amount of time a call lasts



  • Calls by DID, Queue, and Agent

  • Calls by Queue

    • Listed by queue shows the quantity, wait time, ring time, and call time of calls



  • Calls by DID

    • Listed by the DID shows the quantity, wait time, ring time, and call time of calls



  • Calls by Agent

    • Listed by the agent shows the quantity, wait time, ring time, and call time of calls




​Failed Calls



  • ​Records Filter


  • Date

    • Enter the date from and date to for results



  • Queue

    • Select one or multiple queues for results



  • DID Number

    • Select one or multiple DIDs for results




  • Total Failed


  • Abandon

    •  



  • SysCompat

    •  



  • Exit with Timeout

    •  



  • Exit empty

    •  




  • Average Wait Time


  • Abandon

    •  



  • SysCompat

    •  



  • Exit with Timeout

    •  



  • Exit empty

    •  




  • ​Abandoned Calls


    • All abandoned calls are shown by a column chart. Hover over each column to view the exact number of calls




  • Failed Calls by DID and Queue


  • Failed by Queue

    • Listed by queue shows the quantity and average wait time



  • Failed by DID

    • Listed by DID shows the quantity and average wait time




Queue Statistics



  • Queue​


  • Select the queue to view the statistics of selected queue

  • Total Failed

    • Listed by total calls, total failed calls, and total calls. Each total is then further detailed by the quantity, the wait time, and the call time



  • Success Calls

    • Listed by Blind Transfer and Completed Call by Agent. Each type of successful call is separated by total quantity, total wait time, total ring time, and total call time



  • Failed Calls

    • Listed by Abandoned Calls and Exit with Timeout Call. Each type of Failed call is separated by total quantity and total wait time



  • Calls by DID

    • Listed by DID number. Each DID is broken down by successful calls, failed calls, wait time, ring time, call time, and failed wait



  • Calls by Agent

    • Listed by agent name. Each agent is broken down by quantity, wait time, ring time, and call time




  • Records Filter


  • Date

    • Enter the date from and date to for results



  • Queue

    • Select one or multiple queues for results



  • DID Number

    • Select one or multiple DIDs for results




Agent Statistics



  • Agent Name


  • Select the agent to view the statistics of specified agent

    • Total success

      • The successful calls are listed by quantity, wait time, and call timeSuccess CallsListed by Completed by agent, blind transfer, and completed by caller. Each type of call is broken down by quantity, wait time, ring time, and call time



    • No Answer on Ring

    • Calls by DID

      • Listed by DID, each DID is broken down by quantity, wait time, ring time, and call time



    • Calls by Queue

      • Listed by Queue, each Queue is broken down by quantity, wait time, ring time, and call time 





Dashboard

The QuBe Admin Dashboard is used to show any alerts or configuration errors. The configuration errors are listed on the admin dashboard to alert the PBX Admin of the errors to avoid a faulty call flow. The Possible Destination Problems (Fig. 1) display show the errors listed by: function, label, field error, and error location​

The System Status Report displays real time system status information. The system report monitors /var/spool/asterisk/monitor to prevent and stay above any system failures.

Panel

The QuBe Operator Panel is built in and is included a no extra cost. Typically features like this in other graphical PBX software packages are at an added cost and difficult to install. We have removed the stress of getting a fully functioning operator panel working right from the start.


  • Extension

    • Color Identifier​​

      • All extensions in red are not registered

      • All extensions in green are registered





  • Line

    • Line one and line two show the active lines



  • Voicemail

    • The mail icon on the top right of each user show if there is an active voicemail message.

      • Blinking indicates new messages

      • Solid indicates no message however you can drag and drop an active call into the voicemail of the users





  • Trunk List

    • The list of all IP trunks in the PBX



  • Active Conference List

    • The active conferences are listed



  • Queue List

    • Callers in Queue

    • Callers Logged in

    • Callers Talk Time

    • Callers Longest Hold

    • Caller Available

    • Caller Hold Time




 


  • Your Extension

    • On the top right, select the Your Extension button to select a specific extension assigned to your user to view activity 



Queue admin

The QuBe Queue Admin module is used to visually manage the users in a queue. The Queue Admin module lists active calls and which queue the call(s) entered. Members / Users are listed by extension number and which queue it is a part of.


  • Members

    • Listed in the members section are all the other agents in the queue that the specified extension is in



  • Calls in Queues

    • This section shows the number of active calls in the specified queue that the extension is also listed in



  • Active Calls

    • This section shows the active calls with both the member and the caller, which queue the call is in, and the amount of time the call in the queue has been active



  • Queue List

    • ​Callers in Queue

    • Callers Logged In

    • Callers’ Talk Time

    • Callers’ Longest Hold

    • Caller Available

    • Caller Hold Time



Pop-up Config

The QuBe PBX Pop up Config module is currently being developed for our PBX. The Pop up Config module is used as a CRM integration with many customizable options! The Pop up Config module was designed with you in mind. 

The Pop up module is integrated with our Queue Admin module to allow you and your users to manage all incoming calls by a simple click.