VITAL PBX SETUP

Pickup Group specifies to which pickup groups an extension belongs. An extension can belong to multiple pickup groups. The call group and pickup group options allow users to pick up calls that are not directed to them by dialing a feature code (*08). Calls directed to any phone in a particular call group can be answered by any user who is a member of the corresponding pickup group. For example, a user in pickup group Support will be able to pick up any call directed to any phone in the Support call group. This can be useful for small office or home setups, where it is easier to simply pick up a call from another phone rather than forward that call to another extension. Note that a user can be part of a pickup group without being a member of the associated call group. For example, a senior staff member may be able to pick up any call directed to anyone in his department, but his department should not be able to pick up calls directed to the senior staff member.

General

VitalPBX Pickup Groups

  • Description*, a description to identify the pickup group, can consist of numbers and names.
Extension Section
  • Extension, any extension that you want to add to this group.
  • Member, when enabled, indicates that the extension is a member of the pickup group.
  • Allow Pickup, when enabled, indicates that this extension can pick up calls that are directed to this group.
  • Ring Groups offer the possibility that a call to be received by more than one internal extension. The option is used most often for picking up calls received on a certain line (Inbound Routes) and sending them to a certain destination (Welcome prompt, IVR and ring groups). Also, the group can be accessed internally, calling the code assigned to it.

    General

    VitalPBX Call Center - Ring Groups

    • Code*, number to dial to reach this service.
    • Description*, short description to identify this ring group.
    • Extensions, list of extension for this ring group.
    • External Numbers, list of external number for this ring group.
    • Ring Strategy, ring strategy of extension group.
    • Ring Time, ring time in seconds, MAX 160 seconds.
    • Class of Service, Class of Service to use for dial the external numbers listed in this ring group.
    • Music on Hold, music on hold to play.
    • CID Name Prefix, prefix to append to this ring group.
    • Allow Diversions, allows you to define if the diversions defined for the different extension members will be applied or not.
    • Mark Cancelled Calls as Answered, with this option enabled prevents the other phones to record a missed call when the call has been answered by another member listed on this ring group. This is a very useful setting when the ring strategy is set to “Ring All”
    Last Destination Section
    • Select Module, allows the user to choose from a drop-down list of available modules, which module should be activated.
    • Select Destination, this is the call target to which the module should be routed.
    • ACD (Automatic Call Distributor) is what distributes incoming calls in the order of arrival to the first available agent. The system answers each call immediately and, if necessary, holds it in a queue until it can be directed to the next available call center agent. Balancing the workload among agents ensures that each caller receives prompt and professional service.

      General

      VitalPBX Call Center - Queues

      Code*, number to reach this service.

      Description*, short Description to identify this queue.

      Strategy, this defines the strategy to ring this queue. Options are:

      • Ring All: Ring all available channels until one of the members answers the phone.
      • Least Recent: Ring interface which was least recently hung up by this queue.
      • Fewest Calls: Ring the one with fewest completed calls from this queue.
      • Random: Ring random interface.
      • Round Robin Memory: Round robin with memory, remember where we left off last ring pass.
      • Round Robin Ordered: Same as Round Robin Memory, except the queue member order from config file is preserved.
      • Linear: Rings interfaces in the order specified in this queue. If you use dynamic members, the members will be rung in the order in which they were added.
      • Weight Random: Rings random interface but uses the member’s penalty as a weight when calculating their metric.

      CID Name Prefix, prefix to append to this queue, typically indicate to the agents from which queue the call comes.

      Join Announcement, allowing you to define an announcement to be played to the caller immediately as they reach the queue.

      Agent Announcement, an announcement may be specified which is played for the member as soon as they answer a call, typically to indicate to them which queue this call should be answered as, so that agents or members who are listening to more than one queue can differentiated how they should engage the customer.

      Service Level, the idea is to define the maximum acceptable time for a caller to wait before being answered. You then note how many calls are answered within that threshold, and they go toward your service level. So, for example, if your service level is 60 seconds, and 4 out of 5 calls are answered in 60 seconds or less, your service level is 80%.

      Join Empty, If there are calls queued, and the last agent logs out, the remaining incoming callers will immediately be removed from the queue, and the Queue() call will return, If leavewhenempty” is set to “strict”.

      “joinempty” set to “strict” will keep incoming callers from being placed in queues where there are no agents to take calls. The Queue() application will return, and the dial plan can determine what to do next.

      This setting controls whether callers can join a queue with no members.

      There are three choices:

      • yes – callers can join a queue with no members or only unavailable members
      • no – callers cannot join a queue with no members
      • strict – callers cannot join a queue with no members or only unavailable members
      • loose – same as strict, but paused queue members do not count as unavailable (new in 1.6)

      Leave When Empty, used to control whether callers are kicked out of the queue when members are no longer available to take calls.

      Timeout Priority, this is used to control the priority of the two possible timeout options specified for a queue. The Queue (App) application has a timeout value that can be specified to control the absolute time a caller can be in the queue. The timeout value controls the amount of time (along with retry) to ring a member for. Sometime these values conflict, so you can control which value takes precedence.

      Queue Timeout, this will cause the queue to fail out after a specified number of seconds.

      Member Timeout, this specifies the number of seconds to ring a member’s device.

      Retry, specifies the number of seconds to wait before attempting the next member in the queue if the timeout value is exhausted while attempting to ring a member of the queue.

      Wrap-up time, the number of seconds to keep a member unavailable in a queue after completing a call. This time allows an agent to finish any post call processing they may need to handle before they are presented with the next call.

      Queue Callback, when you have the Queues Callback add-on installed, here you can select the Queue Callback for this Queue.

      Music on Hold, this option sets the class of music for this queue.

      Ring Busy Agent, this is used to avoid sending calls to members whose status is In Use.

      Record, record the calls in this queue.

      Members section
      • Extension, extension number for the Queue Member.
      • Penalty, within a queue, we can penalize members in order to lower their preference for being called when there are people waiting in a particular queue. For example, we may penalize queue members when we want them to be a member of a queue, but to be used only when the queue gets full enough that all our preferred agents are unavailable. This means we can have three queues (say, Support, Sales, and Billing), each containing the same three queue members: James Shaw, Kay Madsen, and Danielle Roberts. Suppose, however, that we want James Shaw to be the preferred contact in the Support Queue, Kay Madsen preferred in Sales, and Danielle Roberts preferred in Billing. By penalizing Kay Madsen and Danielle Roberts in Support, we ensure that James Shaw will be the preferred queue member called. Similarly, we can penalize James Shaw and Danielle Roberts in the Sales Queue, so Kay Madsen is preferred, and penalize James Shaw and Kay Madsen in the Billing Queue so Danielle Roberts is preferred.
      • Member Type, this decides if the member will be Dynamic or Static.
      • Allow Diversion, decide if the member executed or not the diversions when called.
      Final Destination*

      The destination if the call is not answered after the “Queue Timeout”

      • Select Module, allows the user to choose from a drop-down list of available modules, which module should be activated.
      • Select Destination, this is the call target to which the module should be routed.
      After Agent Hang-up Destination

      This option is generally used when performing quality surveys for customer service. Once the call is hung up by the agent, we can send this call to someone to survey the customer or to an IVR tree with all the options. Afterwards, in the case of the IVR tree, we can use the IVR stats add-on to export the information based on the options selected by the customers.

      • Select Module, allows the user to choose from a drop-down list of available modules, which module should be activated.
      • Select Destination, this is the call target to which the module should be routed.

      Announcement Settings

      VitalPBX Call Center - Queues Announcement Settings

      • Periodic Announcement, periodic announcement to provide to the caller. The system default message is “All representatives are currently busy assisting other callers. Please wait for the next available representative.
      • Periodic Announcement Frequency, this indicates how often we should make periodic announcements to the caller. Bear in mind that playing a message to callers on a regular basis will tend to upset them. Pleasant music will keep your callers far happier than endlessly repeated apologies or advertising, so give some thought to:
        • keeping this message short
        • Don’t play it too frequently.
      • Announce First User, if enabled, play announcements to the first user waiting in the Queue. This may mean that announcements are played when an agent attempts to connect to the waiting user, which may delay the time before the agent and the user can communicate.
      • Relative Periodic Announcement, if set to yes, the Periodic Announce Frequency timer will start from when the end of the file being played back is reached, instead of from the beginning.
      • Announce Hold Time, defines whether the estimated hold time should be played along with the periodic announcements.
      • Announce Position, defines whether the caller’s position in the queue should be announced. If you have any logic in your system that can promote callers in rank (i.e., high-priority calls get moved to the front of the queue), it is best not to use this option. Very few things upset a caller more than hearing that they’ve been moved toward the back of the line. Options are:
        • No: The position will never be announced
        • Yes: The caller’s position will always be announced
        • Limit: The caller will hear her position in the queue only if it is within the limit defined by Announce Position Limit.
        • More: The caller will hear her position if it is beyond the number defined by Announce Position Limit.
      • Announce Position Limit, used if you’ve defined Announce Position as either limit or more
      • Announce Frequency, defines how often we should announce the caller’s position and/or estimated hold time in the queue. Set this value to zero to disable. In a small call center, it is unlikely that the system will be able to make accurate estimates, and thus callers are more likely to find this information frustrating.
      • Min Announce Frequency, this specifies the minimum amount of time that must pass before we announce the caller’s position in the queue again. This is used when the caller’s position may change frequently, to prevent the caller hearing multiple updates in a short period of time.
      • Announce Round Seconds, if this value is nonzero, the number of seconds is announced and rounded to the value defined.

      Others

      VitalPBX Call Center - Queues Others Settings

      Members Settings
      • Auto Pause, this enables/disables the automatic pausing of agents who fail to answer a call. A value of All causes this agent to be paused in all queues that they are a member of. This parameter can be tricky in a live environment, because if the agent doesn’t know they’ve been paused, you could end up with agents waiting for calls, not knowing they’ve been paused. Never use this unless you have a way to indicate to the members that they’ve been paused or have a supervisor who is watching the status of the queue in real time.
      • Penalty Members Limit, a limit can be set to disregard penalty settings when the queue has too few members. No penalty will be weighed in if there are only X or fewer queue members.
      • Member Delay, this is used if you want a delay prior to the caller and queue member being connected to each other.
      • Timeout Restart, if set to yes, resets the timeout for an agent to answer if either a BUSY or CONGESTION status is received from the channel. This can be useful if the agent is allowed to reject or cancel a call.
      Other Queue Settings section
      • Queue Weight, this option defines the weight of a queue. A queue with a higher weight defined will get first priority when members are associated with multiple queues. Keep in mind that if you have a very busy queue with a high weight, callers in a lower-weigh queue might never get answered (or have to wait for a long time).
      • Queue Max Length, this value specifies the maximum number of callers allowed to be waiting in a queue. A value of zero means an unlimited number of callers are allowed in the queue.
      • Reset Stats, it allows you to select a Cron Profile to reset the queue stats periodically.
      • IVR, an IVR may be specified, in which if the user types a SINGLE digit extension while they are in the queue, they will be taken out of the queue and sent to that extension in this IVR.
      • VIP Customer, List of VIP Customers, these customers have more priority in this queue.
      • Autofill, the old behavior of the queue (autofill=no) is to have a serial type behavior in that the queue will make all waiting callers wait in the queue even if there is more than one available member ready to take calls until the head caller is connected with the member they were trying to get to.
      • Here you can create dial restrictions rules. These can be associated with a class of services.

        General

        VitalPBX Class of Services - Dialing Restriction Rules

        • Description*, short description to identify this “Dialing Restriction” – must be unique.
        • Custom Rules Context, this allows you to include a custom context with your own advanced rules.
        Rules section
        • Rules (Pattern), allows you to create extension patterns in your dial plan that match one or more possible dialed numbers. The pattern options are:
          • The letter X or x represents a single digit from 0 to 9.
          • The letter Z or z represents a single digit from 1 to 9.
          • The letter N or n represents a single digit from 2 to 9.
          • The period (.) character is a wildcard that matches one or more characters.
          • The exclamation mark (!) is a wildcard that matches zero or more characters.
          • [1237-9] matches any digit or letter in the brackets (in this example, 1,2,3,7,8,9)
          • [a-z] matches any lower-case letter
          • [A-Z] matches any UPPER-case letter
        • Allowed, allow/disallow this pattern.
        • Announcement, you can choose an announcement associated with this pattern.
        • Play Max Duration, play max duration only if max duration is greater than 0 or not in blank.
        • Max Duration, maximum call duration associated with this pattern.
        • Password, this determines if a password is required when using this pattern.
        • Callback is a call target that will immediately hang up on a caller, call them back, and then redirect the call to another call target. This is most often used to avoid long-distance charges for remote agents who do not have access to a VoIP endpoint. This is especially relevant in the case of mobile phones where incoming calls are usually significantly cheaper than outgoing calls. The callback target may connect the caller with any resource on VitalPBX (such as an extension, the voicemail messaging center, or a queue), or it may be used in conjunction with DISA to give the caller a dial tone on the system from which they can call any telephone number they wish.

          General

          VitalPBX Call Back

          • Description*, short Description to identify this callback.
          • Number, this is the phone number that VitalPBX will dial to reconnect with the caller after the call that initiated the callback is terminated. The number must be in a format that one of the outbound routes configured in the Outbound Routes section of VitalPBX can be matched with (for example, if there is no outbound route defined to match a 10-digit dialing pattern, entering 5551234567 for this field would render the callback configuration useless, as the outbound callback would never be completed). If the field is left blank, then VitalPBX will attempt to call back the caller ID number that initiated the callback.
          • Dial Prefix, this prepends the prefix to the number to dial.
          • Delay, delay before return call.
          • Class of Service, Class of Service for making the call.
          Destination* Section
          • Select Module, to choose which module should be activated.
          • Select Destination, to configure the call target that the caller will be connected to, once the callback dialog reconnects the caller to VitalPBX. Any existing call target can be used.

          A few common examples of when a callback target might be used are as follows:

          • A company where employees need the ability to check their voicemail from anywhere. Calling the toll-free company phone number costs the company too much money. A callback target could be set up to call back the incoming caller ID and be directed to the miscellaneous destination of *98. Callers would receive a call on the number they called from, would be prompted for their extension and their password, and would then have access to their voicemail messages.
          • A company receives better per-minute rates on calls made through its VoIP trunks than calls made through employees’ mobile phones. Employees’ mobile phones have free incoming calls. A callback target could be set up for each employee with a mobile phone to call back the employee’s mobile number. The callback would be directed to a DISA destination to give the employee a dial tone on the PBX (allowing them to dial out using the company’s VoIP trunks without using any outgoing mobile minutes).
          • A company that receives collect calls from anywhere in the world (such as a credit card company that needs to receive calls if a customer’s card is lost or stolen). The company reduces their costs if they use a VoIP trunk local to the country that the customer is in, rather than paying for the entire collect call at hefty international rates. A callback target could be set up to call back the incoming caller ID of the customer and be directed to a queue. The customer would receive a call to the number they called from and would be connected with a company representative as soon as one is available.
          • A conference room allows a group of people to participate in phone call.  The most common form of bridge allows participants dial into a virtual meeting room from their own phone.  Meeting rooms can hold dozens or even hundreds of participants.  This contrasts with three-way calling, a standard feature of most phone systems which only allows a total of three participants.  For most phone systems, conference bridging is an add-on feature that costs thousands of dollars.

            General

            VitalPBX Applications Conferences

            • Extension*, number to dial to reach this service. This is a number that internal endpoints can dial to reach this conference. Like the ring groups, this can be thought of as the extension number of the conference.
            • Description*, short description for identify this conference.
            • Max Members, this option limits the number of participants for a single conference to a specific number. After the limit is reached, the conference will be locked until someone leaves. Note however that an Admin user will always be allowed to join the conference regardless if this limit is reached or not.
            • Video Mode, Options:
              • None: No video sources are set by default in the conference. It is still possible for a user to be set as a video source via AMI or DTMF action at any time.
              • Follow Talker: The video feed will follow whoever is talking and providing video.
              • Admin: The first administrator who joins the conference with video capability is the only source of video distribution to all participants. If the administrator leaves, the next administrator to join after them becomes the source.
            • User PIN*, this is a numeric passcode that is used to enter the conference room. If a PIN is entered in this field, no one is able to join the conference room without entering the PIN.
            • Admin PIN, functions in the same way as the User PIN. The Admin PIN and User PIN should not be set to the same value. The Admin PIN is used in conjunction with the Wait Admin option explained further in this chapter, in order to identify the administrator or leader of the conference.
            • Language, here you can set the language used for voice prompts to the conference.
            • Record Conference, when set to yes, records the conference call starting when the first user enters the room, and ending when the last user exits the room.
            Conference Settings section
            • Music on Hold, the music on hold class to use for this conference.
            • Announce User Count, used for announcing the participant count to all members of the conference. If set to a number, then the announcement is only played when the number of participants is above the set number. Available options are yes, no, or a whole number. Default is no.
            • Music on Hold When Empty, when this option is enabled on-hold music will be played if there is only one caller in the conference room or if the conference has not started yet (because the leader has not arrived). If this option is disabled, no sound will be played during these situations.
            • User Count, when this option is enabled, the number of users currently in the conference room will be announced to each caller before they are bridged into the conference.
            • Announce Join/Leave, when enabled, this option will prompt the user for a name when entering the conference. After the name is recorded, it will be played when the user enters or exits the conference.
            • Announce Only User, sets if the only user announcement should be played when a user enters an empty conference.
            • Wait for Leader, if this option is enabled, the conference will not begin until the conference administrator joins the conference room. The administrator is identified by the Admin PIN. If other callers join the conference room before the leader does, they will hear on-hold music or silence until the conference begins (what they hear depends on the MoH When Empty setting explained earlier in this section). If this option is set to “No”, the callers will be bridged into the conference as soon as they call the conference room number.
            • Start Muted, when this option is enabled, all users joining the conference are initially muted.
            • Drop Silence, this option drops what Asterisk detects as silence from entering into the bridge. Enabling this option will drastically improve performance and help remove the buildup of background noise from the conference. Highly recommended for large conferences due to its performance enhancements.
            • Quiet, when this option is enabled, user introductions, enter prompts, and exit prompts are not played. There are some prompts, such as the prompt to enter a PIN number that will still be played regardless of how this option is set.
            • Kick Users, enabling this option will kick out all remaining users of the conference, after the last admin user leaves the conference.
            • Talk Detection, this option sets whether or not notifications of when a user begins and ends talking should be sent out as events over AMI.
            • Allow to Invite, if enabled, all the participants could press “**” or “0” to invite other people to this conference.

            The following codes can be entered by all conference participants:

            • *1 – toggles mute for the user. When enabled, anything the user says is not transmitted to the rest of conference members. If the conference is being recorded, anything said by a muted user is not part of the recording.
            • *4 – decreases receive volume. The user can tap this option to decrease the volume of what they are hearing. This does not affect what any other conference member hears. If a user is finding other conference members too loud, they can press *4 a few times to make the conference quieter for themselves.
            • *5 – increases receive volume. The user can tap this option to increase the volume of what they are hearing. This does not affect what any other conference member hears. If a user is having trouble hearing other members of the conference, they can press *5 a few times to make the conference louder for themselves.
            • *6 – decreases transmit volume. The user can tap this option to decrease the volume of what they are transmitting to the rest of the conference members. When this option is used, the user will sound quieter to all other conference members. If a user is much louder than the other members of a conference room, they can tap *6 few times to make their transmit volume quieter.
            • *7 – increases transmit volume. The user can tap this option to increase the volume of what they are transmitting to the rest of the conference members. When this option is used, the user will sound louder to all other conference members. If the conference members are having trouble hearing a particular user, that user can tap *7 a few times to make their transmit volume louder.
            • *8 – user can tap this code to leave the conference.
            • In addition, the admin has access to additional codes:
            • *2 – toggles the conference lock. When a conference is locked, no more callers may join. A locked conference must be unlocked for any new users to join. This option is only available to a conference administrator. If the conference does not have an admin PIN configured or the user has joined the conference as a user instead of an admin, this option is not available.
            • *3 – Kicks the last user who joined the conference from the conference room. The user will hear a message informing them that they have been kicked from the conference and that their call will be terminated. Note that if a conference is unlocked, the user may rejoin. The best way to remove an abusive conference user is to eject them and then immediately lock the conference. This option is only available to a conference administrator. If the conference does not have an admin PIN configured, or the user has joined the conference as a user.

Class of Service is a Group of settings that define the dial plan that has access to the extension.

General

VitalPBX Class of Services

  • Class of Service *, Class of Service Name (Must be Unique). Alphanumeric values with dash and underscore are allowed.
  • Description *, this is a short Description for identify this Class of Service.
  • Feature Category, features allowed for this Class of Service.
  • Dial Restrictions, dial restriction rules set for this Class of Service.
  • Route Selection, routes to use for this Class of Service.
  • Allowed Calls By, it defines the list of CoS to be allowed to call to this CoS when Private field is checked.
  • Private, it defines if extensions with this CoS may be called by others with different CoS. If is checked only calls with the same CoS or calls coming from CoS selected on Allowed Calls By field will be allowed. It applies for internal calls only.
  • In this module you can create groups of feature codes. This allows you to prevent some users from having access to some of the more sensitive feature codes.

    General

    VitalPBX Class of Services - Feature Categories

    • Description*, this is a short description to identify this feature category – must be unique.
    • Available Features, list of available feature codes.
    • Enabled Features, list of feature codes that have been included.
    • Ring Groups offer the possibility that a call to be received by more than one internal extension. The option is used most often for picking up calls received on a certain line (Inbound Routes) and sending them to a certain destination (Welcome prompt, IVR and ring groups). Also, the group can be accessed internally, calling the code assigned to it.

      General

      VitalPBX Call Center - Ring Groups

      • Code*, number to dial to reach this service.
      • Description*, short description to identify this ring group.
      • Extensions, list of extension for this ring group.
      • External Numbers, list of external number for this ring group.
      • Ring Strategy, ring strategy of extension group.
      • Ring Time, ring time in seconds, MAX 160 seconds.
      • Class of Service, Class of Service to use for dial the external numbers listed in this ring group.
      • Music on Hold, music on hold to play.
      • CID Name Prefix, prefix to append to this ring group.
      • Allow Diversions, allows you to define if the diversions defined for the different extension members will be applied or not.
      • Mark Cancelled Calls as Answered, with this option enabled prevents the other phones to record a missed call when the call has been answered by another member listed on this ring group. This is a very useful setting when the ring strategy is set to “Ring All”
      Last Destination Section
      • Select Module, allows the user to choose from a drop-down list of available modules, which module should be activated.
      • Select Destination, this is the call target to which the module should be routed.
    • ACD (Automatic Call Distributor) is what distributes incoming calls in the order of arrival to the first available agent. The system answers each call immediately and, if necessary, holds it in a queue until it can be directed to the next available call center agent. Balancing the workload among agents ensures that each caller receives prompt and professional service.

      General

      VitalPBX Call Center - Queues

      Code*, number to reach this service.

      Description*, short Description to identify this queue.

      Strategy, this defines the strategy to ring this queue. Options are:

      • Ring All: Ring all available channels until one of the members answers the phone.
      • Least Recent: Ring interface which was least recently hung up by this queue.
      • Fewest Calls: Ring the one with fewest completed calls from this queue.
      • Random: Ring random interface.
      • Round Robin Memory: Round robin with memory, remember where we left off last ring pass.
      • Round Robin Ordered: Same as Round Robin Memory, except the queue member order from config file is preserved.
      • Linear: Rings interfaces in the order specified in this queue. If you use dynamic members, the members will be rung in the order in which they were added.
      • Weight Random: Rings random interface but uses the member’s penalty as a weight when calculating their metric.

      CID Name Prefix, prefix to append to this queue, typically indicate to the agents from which queue the call comes.

      Join Announcement, allowing you to define an announcement to be played to the caller immediately as they reach the queue.

      Agent Announcement, an announcement may be specified which is played for the member as soon as they answer a call, typically to indicate to them which queue this call should be answered as, so that agents or members who are listening to more than one queue can differentiated how they should engage the customer.

      Service Level, the idea is to define the maximum acceptable time for a caller to wait before being answered. You then note how many calls are answered within that threshold, and they go toward your service level. So, for example, if your service level is 60 seconds, and 4 out of 5 calls are answered in 60 seconds or less, your service level is 80%.

      Join Empty, If there are calls queued, and the last agent logs out, the remaining incoming callers will immediately be removed from the queue, and the Queue() call will return, If leavewhenempty” is set to “strict”.

      “joinempty” set to “strict” will keep incoming callers from being placed in queues where there are no agents to take calls. The Queue() application will return, and the dial plan can determine what to do next.

      This setting controls whether callers can join a queue with no members.

      There are three choices:

      • yes – callers can join a queue with no members or only unavailable members
      • no – callers cannot join a queue with no members
      • strict – callers cannot join a queue with no members or only unavailable members
      • loose – same as strict, but paused queue members do not count as unavailable (new in 1.6)

      Leave When Empty, used to control whether callers are kicked out of the queue when members are no longer available to take calls.

      Timeout Priority, this is used to control the priority of the two possible timeout options specified for a queue. The Queue (App) application has a timeout value that can be specified to control the absolute time a caller can be in the queue. The timeout value controls the amount of time (along with retry) to ring a member for. Sometime these values conflict, so you can control which value takes precedence.

      Queue Timeout, this will cause the queue to fail out after a specified number of seconds.

      Member Timeout, this specifies the number of seconds to ring a member’s device.

      Retry, specifies the number of seconds to wait before attempting the next member in the queue if the timeout value is exhausted while attempting to ring a member of the queue.

      Wrap-up time, the number of seconds to keep a member unavailable in a queue after completing a call. This time allows an agent to finish any post call processing they may need to handle before they are presented with the next call.

      Queue Callback, when you have the Queues Callback add-on installed, here you can select the Queue Callback for this Queue.

      Music on Hold, this option sets the class of music for this queue.

      Ring Busy Agent, this is used to avoid sending calls to members whose status is In Use.

      Record, record the calls in this queue.

      Members section
      • Extension, extension number for the Queue Member.
      • Penalty, within a queue, we can penalize members in order to lower their preference for being called when there are people waiting in a particular queue. For example, we may penalize queue members when we want them to be a member of a queue, but to be used only when the queue gets full enough that all our preferred agents are unavailable. This means we can have three queues (say, Support, Sales, and Billing), each containing the same three queue members: James Shaw, Kay Madsen, and Danielle Roberts. Suppose, however, that we want James Shaw to be the preferred contact in the Support Queue, Kay Madsen preferred in Sales, and Danielle Roberts preferred in Billing. By penalizing Kay Madsen and Danielle Roberts in Support, we ensure that James Shaw will be the preferred queue member called. Similarly, we can penalize James Shaw and Danielle Roberts in the Sales Queue, so Kay Madsen is preferred, and penalize James Shaw and Kay Madsen in the Billing Queue so Danielle Roberts is preferred.
      • Member Type, this decides if the member will be Dynamic or Static.
      • Allow Diversion, decide if the member executed or not the diversions when called.
      Final Destination*

      The destination if the call is not answered after the “Queue Timeout”

      • Select Module, allows the user to choose from a drop-down list of available modules, which module should be activated.
      • Select Destination, this is the call target to which the module should be routed.
      After Agent Hang-up Destination

      This option is generally used when performing quality surveys for customer service. Once the call is hung up by the agent, we can send this call to someone to survey the customer or to an IVR tree with all the options. Afterwards, in the case of the IVR tree, we can use the IVR stats add-on to export the information based on the options selected by the customers.

      • Select Module, allows the user to choose from a drop-down list of available modules, which module should be activated.
      • Select Destination, this is the call target to which the module should be routed.

      Announcement Settings

      VitalPBX Call Center - Queues Announcement Settings

      • Periodic Announcement, periodic announcement to provide to the caller. The system default message is “All representatives are currently busy assisting other callers. Please wait for the next available representative.
      • Periodic Announcement Frequency, this indicates how often we should make periodic announcements to the caller. Bear in mind that playing a message to callers on a regular basis will tend to upset them. Pleasant music will keep your callers far happier than endlessly repeated apologies or advertising, so give some thought to:
        • keeping this message short
        • Don’t play it too frequently.
      • Announce First User, if enabled, play announcements to the first user waiting in the Queue. This may mean that announcements are played when an agent attempts to connect to the waiting user, which may delay the time before the agent and the user can communicate.
      • Relative Periodic Announcement, if set to yes, the Periodic Announce Frequency timer will start from when the end of the file being played back is reached, instead of from the beginning.
      • Announce Hold Time, defines whether the estimated hold time should be played along with the periodic announcements.
      • Announce Position, defines whether the caller’s position in the queue should be announced. If you have any logic in your system that can promote callers in rank (i.e., high-priority calls get moved to the front of the queue), it is best not to use this option. Very few things upset a caller more than hearing that they’ve been moved toward the back of the line. Options are:
        • No: The position will never be announced
        • Yes: The caller’s position will always be announced
        • Limit: The caller will hear her position in the queue only if it is within the limit defined by Announce Position Limit.
        • More: The caller will hear her position if it is beyond the number defined by Announce Position Limit.
      • Announce Position Limit, used if you’ve defined Announce Position as either limit or more
      • Announce Frequency, defines how often we should announce the caller’s position and/or estimated hold time in the queue. Set this value to zero to disable. In a small call center, it is unlikely that the system will be able to make accurate estimates, and thus callers are more likely to find this information frustrating.
      • Min Announce Frequency, this specifies the minimum amount of time that must pass before we announce the caller’s position in the queue again. This is used when the caller’s position may change frequently, to prevent the caller hearing multiple updates in a short period of time.
      • Announce Round Seconds, if this value is nonzero, the number of seconds is announced and rounded to the value defined.

      Others

      VitalPBX Call Center - Queues Others Settings

      Members Settings
      • Auto Pause, this enables/disables the automatic pausing of agents who fail to answer a call. A value of All causes this agent to be paused in all queues that they are a member of. This parameter can be tricky in a live environment, because if the agent doesn’t know they’ve been paused, you could end up with agents waiting for calls, not knowing they’ve been paused. Never use this unless you have a way to indicate to the members that they’ve been paused or have a supervisor who is watching the status of the queue in real time.
      • Penalty Members Limit, a limit can be set to disregard penalty settings when the queue has too few members. No penalty will be weighed in if there are only X or fewer queue members.
      • Member Delay, this is used if you want a delay prior to the caller and queue member being connected to each other.
      • Timeout Restart, if set to yes, resets the timeout for an agent to answer if either a BUSY or CONGESTION status is received from the channel. This can be useful if the agent is allowed to reject or cancel a call.
      Other Queue Settings section
      • Queue Weight, this option defines the weight of a queue. A queue with a higher weight defined will get first priority when members are associated with multiple queues. Keep in mind that if you have a very busy queue with a high weight, callers in a lower-weigh queue might never get answered (or have to wait for a long time).
      • Queue Max Length, this value specifies the maximum number of callers allowed to be waiting in a queue. A value of zero means an unlimited number of callers are allowed in the queue.
      • Reset Stats, it allows you to select a Cron Profile to reset the queue stats periodically.
      • IVR, an IVR may be specified, in which if the user types a SINGLE digit extension while they are in the queue, they will be taken out of the queue and sent to that extension in this IVR.
      • VIP Customer, List of VIP Customers, these customers have more priority in this queue.
      • Autofill, the old behavior of the queue (autofill=no) is to have a serial type behavior in that the queue will make all waiting callers wait in the queue even if there is more than one available member ready to take calls until the head caller is connected with the member they were trying to get to.
    • In the simplest of terms, a trunk is a pathway into or out of a telephone system. A trunk connects VitalPBX to outside resources, such as PSTN telephone lines or additional PBX systems to perform inter-system transfers. Trunks can be physical, such as a PRI or PSTN line, or they can be virtual by routing calls to another endpoint using Internet Protocol (IP) links.

      Trunks are the PBX equivalent of an external phone line. They are the links that allow your system to make calls to the outside world, and to receive calls from the outside world. Without a trunk, you cannot call anyone, and no one can call you. You can configure a trunk to connect with:

      • Any VoIP service provider
      • Any PSTN/Media Gateway, which allows you to make and receive calls over standard telephone lines from your local telephone company
      • Connect directly to another PBX.

      General

      VitalPBX External - Trunks

      • Technology, the type of trunk that you want to create.
        • PJSIP
        • SIP
        • IAX2
        • TELEPHONY (Shown if the DAHDI add-on is installed)
        • TENANT (Shown if a Tenant is created)
        • CUSTOM

      SIP, PJSIP and IAX2 trunks utilize the technologies of their namesakes. These trunks have the same highlights and pitfalls that extensions and devices using the same technology do. Telephony trunks require physical hardware cards for incoming lines to plug into. SIP trunks are the most widely adopted and compatible, but have difficulties traversing firewalls. IAX2 trunks are able to traverse most firewalls easily but are limited to Asterisk-based systems.

      Setting up a trunk is very similar to setting up an extension. All of the trunks share common setup fields, followed by fields that are specific to the technology of the trunk.

      • Description, a description to help identify this trunk
      • Class of Service, class of service to be used by this trunk.
      • Ring Timer, time to ring the trunk before determining that the call cannot be completed.
      • Dial Profile, there are many options that you can set on the outbound call, including call screening, distinctive ringing, and more. Go to Settings > Technology > Dial Profile for more information.
      • Profile, profile with common parameters for the technology selected.
      • Music on Hold, default music on hold for this trunk.
      • Codecs, list of allowed codecs for SIP trunks, in order of preference. Codecs that are not list will not be allowed for this trunk.Port, the port number we want to connect to on the remote side.
      • NAT, (Network Address Translation) is a technology most commonly used by firewalls and routers to allow multiple devices on a LAN with “private” IP addresses to share a single public IP address. A private IP address is an address, which can only be addressed from within the LAN, but not from the Internet outside the LAN. The Options are:
        • No: Do no special NAT handling other than RFC3581
        • Force: Pretend there was a rport parameter even if there was
        • Comedia: Send media to the port Asterisk received it from regardless of where the SDP says to send it.
        • Auto Force: Set the force_rport option if Asterisk detects NAT.
        • Auto Comedia: Set the comedia option if Asterisk detects NAT.
      • Get DID From, it allows you to define from which SIP header will be extracted the DID number.
      • Get CID From, it allows you to define from which SIP header will be extracted the caller ID info.
      • Trunk CID, this sets the default caller ID name and number that will be displayed to the called party. The Trunk CID will only be used if Overwrite CID field is set to Yes. Note that setting the outbound caller ID only works on digital lines (T1/E1/J1/PRI/BRI/SIP/IAX2), not POTS lines. The ability to set outbound caller ID must also be supported by your provider.
      • Name, a string that can be used to identify calls on this trunk. If this field is left blank, only the Trunk CID number will be sent.
      • Number, the telephone number that will be displayed by calls on this trunk.
      • DTMF Mode, set default dtmf-mode for sending DTMF. Default rfc2833|rfc4733, Options:
        • info: SIP INFO messages (application/dtmf-relay)
        • shortinfo: SIP INFO messages (application/dtmf)
        • inband: Inband audio (requires 64 kbit codec -alaw, ulaw)
        • auto: Use rfc2833|rfc4733 if offered, in-band otherwise
      • Overwrite CID, Overwrites the CID sent by the Extension or module. You got the following options:
      • No, no overwrite will take place and the CID number is preserved.
      • Yes, will overwrite any CID number sent through this route.
      • If not Provided, will overwrite the CID information if no External CID is provided.
      • Disable Trunk, this allows you to disable this trunk to be inaccessible.
      • Continue on Busy, it forces to continue the call to the next configured trunk when this trunk being busy. Note: The call will also continue to the next trunk if any error happens, even if this checkbox is not checked.

      SIP/PJSIP/IAX settings

      Device for Outgoing Calls (Peer)
      • Outgoing Username, Username the remote server should use to contact this PBX. It is also the device name that will be created.
      • Host, this is the IP address or DNS hostname of the SIP provider. This is the destination server or network that VitalPBX will send calls to when using this trunk.
      • Port, this value sets the default port to be accessed on the remote endpoint device. Only required for SIP trunks.
      • Local Secret, secret to be used for authentication requests from remote server.
        • Insecure: Allows relaxing authentication of incoming SIP requests. Options:
        • Port: Allow matching of peer by IP address without matching port number
        • Invite: Do not require authentication of incoming INVITES’
        • Port, Invite: The combination is the minimum security since no checking or port check or authentication to the INVITE message type.
      • Allow Inbound Calls, if checked, this device will be allowed also to accept calls.
      • Remote Username, authentication username for remote server
      • Remote Secret, the password credential used to authenticate this trunk against the provider
      • From User, the user credential used to authenticate this trunk against the provider
      • From Domain, as your provider knows your domain
      • Qualify, causes VitalPBX to regularly send a SIP OPTIONS command to check that the peer is still online. If the peer does not answer within the configured period, VitalPBX will consider the device to be off-line and not available for future calls.
      • IAX Trunking, allows sending voice of several calls in one IAX packet. It can significantly reduce the required network bandwidth.
      Device for Incoming Calls (User)
      • Username, the username credential used to contact this trunk
      • Host, the host they use to contact us (We could specify the “dynamic” option and leave open the possibility that any device connected to your machine without an IP in particular.)
      • Local Secret, secret to be used for authentication requests from remote server.
      • Insecure, Sets the level of authentication and verification established between machines when performing communication. Options are:
        • Port: Allow matching of peer by IP address without matching port number
        • Invite: Do not require authentication of incoming INVITEs
        • Port, Invite: The combination is the minimum security since no checking or port check or authentication to the INVITE message type.
      • IP Authentication, if checked, allows the incoming requests authentication by IP address in addition to the username authentication.
      • Qualify, make periodic checks to make sure that the user is alive.
      • IAX Trunking, allows sending voice of several calls in one IAX packet. It can significantly reduce the required network bandwidth.
      General Configurations (PJSIP)
      • Transport, explicit transport configuration to use.
      • Match, the value is a comma-delimited list of IP addresses or hostnames. IP addresses may have a subnet mask appended. The subnet mask may be written in either CIDR or dotted-decimal notation. Separate the IP address and subnet mask with a slash (“/”).
      • Contacts, Permanent contacts assigned to an AoR. You can define multiple contact addresses in SIP URI format. e.g.: sip:198.51.100.1:5060.
      Outbound Registration Settings (PJSIP)
      • Require Registration, it defines, if is required to register against the remote server or VoIP provider.
      • Permanent Auth Rejection, if this option is enabled and an authentication challenge fails, registration will not be attempted again until the configuration is reloaded.
      • Client URI, this is the address-of-record for the outbound registration (i.e. the URI in the to header of the REGISTER). For registration with an ITSP, the client SIP URI may need to consist of an account name or number and the provider hostname for their registrar, e.g. [email protected]. This may differ between providers.
      • Server URI, this is the URI at which to find the registrar to send the outbound REGISTER. This URI is used as the request URI of the outbound REGISTER request from Asterisk.
      • Contact User, this establishes the user portion from the SIP contact header of the SIP URI. This will affect the reached extension on the Dial Plan when the far end calls you at this registration.
      • Max Retries, maximum number of registration attempts.
      • Expiration, expiration time for registrations in seconds.
      • Retry Interval, interval in seconds between retries if outbound registration is unsuccessful.
      • Forbidden Retry Interval, it defines the time to wait before attempting registration again, after receiving a 403 Forbidden response. If 0 is specified, no retry will be made after receiving a 403 Forbidden response.
        • For registration to generic registrars, the client SIP URI will depend on networking specifics and configuration of the registrar.
        • For registration with an ITSP, the setting may often be just the domain of the registrar, e.g. sip:sip.example.com.
      Register String
      • Register String, the register line includes a host name (mydomain.com) which tells Asterisk where to send the registration request; the account number and password, for example: account:[email protected]:5060

      Telephony Settings

      General

      VitalPBX External - Trunks Telephony Settings

      DAHDI Trunk Parameters
      • Channel Group, the channel group used by this trunk.
      • Mode, selection mode for available channels.

      Advanced

      In Advanced Settings, you can add any valid value in the account settings of the trunk.

      VitalPBX External - Trunks Advanced

      • Type, Friend, User or Peer.
      • Parameter, any valid variable from Asterisk.
      • Value, any value for the Asterisk variable.
      • Enabled, enable or disable the custom setting.

      Dial Manipulation Rules

      VitalPBX External - Trunks Dial Manipulation Rules

      Dialing Manipulation Rules, that allows you to manipulate the dialed number depending of the trunk. e.g.: Suppose you have two providers, both have emergency calls service, but the number to dial is different for each one, for the first provider you must to dial 933 and for the second one you must to dial 944. So, you can configure in your outbound route the 911 and replace this number depending on the trunk on which the call is dialed through.

      Switch to Text Mode

      Due to many requests about configuring trunks in text mode like in other Asterisk distros, we have decided to allow you to create trunks just by writing or pasting the configuration of your provider in a text box. This is to help the customers who come from other distros to have a very easy transition. This option is only available for SIP and IAX2 Trunks.

      VitalPBX External - Trunks Text Mode

      Tenant – Technology

      If it is configured in the main tenant it is to be used as a gateway for the other Tenant, if it is configured in the secondary Tenants it is to allow calls between Tenants. This option will only appear if you have installed the Multi-Tenant add-on module.

      Custom – Technology

      They are used to support protocols type H323 or any other protocol that is not defined.

      IVR (Interactive Voice Response) allows you configure an auto attendant to answer calls and redirect the call-in response to input from the caller. An IVR system is often referred to as a digital receptionist. An IVR plays a pre-recorded message to the caller that asks them to press various buttons on their telephone depending on which department or person they would like to speak with. The IVR system will then route the call accordingly.

      VitalPBX’s IVR allows any digits to be defined for routing purposes. For example, pressing “1” could route the caller to the sales ring group. Destinations can be defined to receive the call if the IVR times out or does not receive valid input.

      It is important that you carefully plan the call flow and branching options for IVRs, while considering the user experience. IVRs use customized Announcements, so you will need to make sure that they are clear and meaningful, and configured to optimize the caller experience. Factors that you should consider include:

      • Handling the timeout when there is no input from the caller
      • Controlling the action to take if caller provides invalid user input
      • Allowing the caller to backtrack if s/he has made a mistake or gets lost
      • Allowing the caller to return to the IVR if voicemail is encountered
      • Whether or not to take advantage of time-based branching, by defining Time Groups, for normal office hours, that include start and end times, start and end days of the week, and much more
      • Defining a Time Condition, and setting one destination if the time matches and a different destination if the time does not match

      General

      VitalPBX Incoming Calls - IVR

      • Description*, short description to identify this IVR. This field is not parsed by VitalPBX.
      • Class of Service, here you can choose a Class of Service for this IVR.
      • Invalid Tries, number of invalid attempts allowed.
      • Welcome Message, welcome message, selected from a drop-down menu of pre-recorded messages that will be played to the caller when they enter the IVR.
      • Instructions Message, Message to be played after the welcome message. This message is useful for avoid repeating the welcome message on invalid/timeout event.
      • Invalid Retry Message, message, selected from a drop-down menu of pre-recorded messages, to be played when the IVR receives an invalid option.
      • Invalid Message, message, selected from a drop-down menu of pre-recorded messages, to be played when user exceeds the maximum number of attempts.
      • Timeout, this value is the maximum time, in seconds, that the system will wait for input from the caller. If this time passes without input, the call will fail over to the Timeout Destination that the user has defined.
      • Timeout Tries, is used to determine the number of times the IVR will repeat itself when no valid input is received. After the specified number of tries, the caller will be send to the Timeout Destination. The maximum number of loops allowed is five.
      • Timeout Retry Message, message, selected from a drop-down menu of pre-recorded messages, to be played when input has not been received within the period defined by Timeout. If the number of Timeout Tries has not yet been reached, then the user will be prompted to try again.
      • Timeout Message, message selected from a drop-down menu of pre-recorded messages, to be played when reaching the timeout.
      • Welcome after Timeout, when enabled, will return the user to the main IVR Welcome Message after playing the Timeout Retry Message.
      • Welcome after Retry, when enabled, will return the user to the main IVR Welcome Message after playing the Invalid Retry Message.
      • Direct Dial, this enables the caller to dial an extension directly from the IVR. If this option is not enabled, the caller will receive a message that they have provided invalid input when they enter an extension, even if the extension is valid.
      • Generate Stats, if is set on yes, it will be saved stats for each option marked. These statistics can be consulted in the IVR Stats module.
      Invalid Destination* Section
      • Select Module, allows the user to choose from a drop-down list of available modules, which module should be activated.
      • Select Destination, this is the call target to which the module should be routed.
      Timeout Destination* Section
      • Select Module, allows the user to choose from a drop-down list of available modules, which module should be activated.
      • Select Destination, this is the call target to which the module should be routed.
      IVR Entries

      This tab defines how to handle the user’s input.

      VitalPBX Incoming Calls - IVR Entries

      • Digit, digit to press.
      • Module, module to activate when the caller presses the appropriate digit.
      • Destination, destination to call when the caller presses the appropriate digit.
      • Enabled, enabled or disable this option.

      It is good practice to ensure that the user has an easy way of getting back to the previous menu. One simple way to do this would be to allow the user to press “*” and link that keypress to the parent IVR.

      Time conditions are a set of rules for hours, dates, or days of the week. A condition has two call targets each time. Calls sent to a time condition will be sent to one target if the time of the call matches one of the conditions, or to the other target if none of the conditions match. Each time condition can have multiple time definitions (known as time groups). Time conditions are often used to control how a phone system responds to callers inside and outside of business hours, and during holidays.

      Before we can set up a time condition call target, we need to define a set of time groups. Time groups are a list of rules against which incoming calls are checked. The rules specify a specific date or time, and a call can be routed differently if the time it comes in matches with one of the rules in a time group. Each time group can have an unlimited number of rules defined. It is useful to group similar sets of time rules together. For example, there may be one-time group for business hours in which the time that the business will be open will be defined. Another popular time group is for holidays, in which each holiday that falls on a business day is defined.

      General

      VitalPBX Incoming Calls - Time Groups

      • Description*, used to identify the time group, when selecting it during the setup of a time condition. This value is not parsed by VitalPBX.
      Schedules Section
      • Time to Start, time, in hours and minutes, that the time group should start.
      • Weekday Start, day of the week that the time group should start.
      • Month Day Start, day of the month that the time group should start.
      • Month Start, month of the year that the time group should start.
      • Time to finish, time, in hours and minutes, that the time group should end.
      • Weekday Finish, day of the week that the time group should end.
      • Month Day Finish, day of the month that the time group should end.
      • Month Finish, month of the year that the time group should end.

      Once a time group has been defined, a time condition can be set up as a call target.

      General

      VitalPBX Incoming Calls - Time Conditions

      • Toggle Code*, dial code to toggle the time condition state through the phone.
      • Description*, short Description to identify this Time Condition.
      • Time Group*, select a Time Group, from the drop-down list, that was created in the Time Groups dialog.
      • Time Zone, as an extended feature, you can set at which Time Zone the Time Condition will be running at.
      • Authorization Pin, optional password to protect from unauthorized people of modifying this time condition.
      • Status, allows you to override the default behavior of a time condition, Options:
        • Default: Default behavior
        • Temporary Matched/Unmatched: Overrides temporary the time condition and sends the calls to the matched/unmatched destination until the current time span has elapsed. After that, the behavior will return to default
        • Permanently Matched/Unmatched: Overrides permanently the time condition and sends the calls to the matched/unmatched destination until the override is removed.
      • BLF Inverted, by default the BLF light color is green when the time condition is matching and red when is not matching. Setting up this to “yes” will make that the behavior be the inverse of what is described above.
      Destination if Time Matches Section
      • Select Module, allows the user to choose from a drop-down list of available modules, which module should be activated.
      • Select Destination, this is the call target to which the module should be routed.
      Destination if Time does not Match Section
      • Select Module, allows the user to choose from a drop-down list of available modules, which module should be activated.
      • Select Destination, this is the call target to which the module should be routed.
      • Once a time group has been defined, a time condition can be set up as a call target.

        General

        VitalPBX Incoming Calls - Time Conditions

        • Toggle Code*, dial code to toggle the time condition state through the phone.
        • Description*, short Description to identify this Time Condition.
        • Time Group*, select a Time Group, from the drop-down list, that was created in the Time Groups dialog.
        • Time Zone, as an extended feature, you can set at which Time Zone the Time Condition will be running at.
        • Authorization Pin, optional password to protect from unauthorized people of modifying this time condition.
        • Status, allows you to override the default behavior of a time condition, Options:
          • Default: Default behavior
          • Temporary Matched/Unmatched: Overrides temporary the time condition and sends the calls to the matched/unmatched destination until the current time span has elapsed. After that, the behavior will return to default
          • Permanently Matched/Unmatched: Overrides permanently the time condition and sends the calls to the matched/unmatched destination until the override is removed.
        • BLF Inverted, by default the BLF light color is green when the time condition is matching and red when is not matching. Setting up this to “yes” will make that the behavior be the inverse of what is described above.
        Destination if Time Matches Section
        • Select Module, allows the user to choose from a drop-down list of available modules, which module should be activated.
        • Select Destination, this is the call target to which the module should be routed.
        Destination if Time does not Match Section
        • Select Module, allows the user to choose from a drop-down list of available modules, which module should be activated.
        • Select Destination, this is the call target to which the module should be routed.
        • This module is used for when you want the caller to hear a message, before being automatically transferred to a fixed destination.

          General

          VitalPBX Incoming Calls - Announcements

          • Description*, short description to identify this Announcement.
          • Custom Recording, here you can select a recording, from the drop-down list, to play in this Announcement.
          Destination after playing announcement* Section
          • Select Module, allows the user to choose from a drop-down list of available modules, which module should be activated.
          • Select Destination, this is the call target to which the module should be routed.
        • With the Passthrough add-on module, you can connect traffic between two trunks together. This allows you to monitor this traffic on the CDR, and even allow you to use VitalPBX as a man in the middle, and record calls for a third-party PBX system.

          PT Trunks

          General

          VitalPBX Pass-Through Trunk

          • Description, this is a brief description to recognize the Passthrough Trunks.
          • Private Trunk, this is the trunk that goes towards a private trunk, being another PBX, for example.
          • Public Trunk, this is a trunk that generally goes towards a PSTN.

          PT Extensions

          General

          VitalPBX Pass-Through Extensions

          • Extension Owner, name of the extension you wish to monitor.
          • Extension, the number of the extension to monitor.
          • Incoming DID, the incoming DID that is related to the Extension monitored.
          • Record Incoming, if this is enabled, then calls incoming from the declared DID towards the Extension Number will be recorded.
          • Record Outgoing, if this is enabled, you can record outgoing calls from the extension declared.
        • In this tab you will find the information about Music on Hold. As an extended feature, you will be able to use an application to playback the MoH. This can be used to stream music from a streaming server.

          General

          VitalPBX Settings Music on Hold

          Name, short description for identify this MoH Category.

          Mode, this defines the mode to play or retrieve music on hold.

        • Options:
          • Files: plays files from a directory in any media format supported by Asterisk.
          • Custom: run a custom application. For example, an online radio.

        Sort, sort the files to listen to.

        Sound file, you can use this to upload a wav or mp3 file.

        Default, if checked, the sounds from this music group will be used for the default music class on hold. Only one music group can be marked as default.

        Application, application that will be used to reproduce a broadcast. If the application is not provided, the mpg123 application will be used in the following format: mpg123 -q -r 8000 -f 8192 –mono -s.

        Transmission URL, URL that will be transmitted by the defined application.

        Format, the format option specifies the audio format that the application will provide to Asterisk.

        In this tab you will find the information about Record Management.

        General

        VitalPBX Settings Recording Managements

        • Name, short description for identify this recording.
        • Sound File, here you can upload a wav or mp3 file.
        Recording List
        • Recording, name of the sound file.
        • Name, description of the sound file.
        • Duration, duration of the recording.
        • Action
          • VitalPBX Gestión de Grabaciones Editar, edit description.
          • VitalPBX Gestión de Grabaciones Escuchar, listen recording.
          • VitalPBX Gestión de Grabaciones Grabar, record by phone.
          • VitalPBX Gestión de Grabaciones Borrar, delete recording.