Extensions

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Extensions allow you to configure extensions (users) and devices (telephones) in your system.

General

VitalPBX Extensions

 

 

  • Extension*, number to dial in order to reach this extension. The extension number must be unique, and should not conflict with an existing extension number, or any other number that is assigned to any other entity within the system, such as a conference, queue, ring group, feature code, etc. The value of this field cannot be changed after the extension has been saved.
  • Name*, name to identify this extension. This is generally the end user’s name or the location of the extension, e.g. Fernando Alonso or Server Room. This value will be displayed as the caller ID text for any calls placed from this extension to other users or devices on the PBX unless the Internal CID field contains a value.
  • CoS Name, the dial plan can be segmented into sections, called Classes of Service (CoS). CoS are the basic organizational unit within the dial plan, and as such, they keep different sections of the dial plan independent of each other. VitalPBX uses CoS to enforce security boundaries between the various parts of the dial plan, as well as to provide different classes of service to different groups of users.
  • Features Password, password to access certain system features and the control panel of the phone.
  • Email Address, email address to where the services messages will be sent.
  • Internal CID, internal Caller ID for the extension, consisting of two parts: the CID Name and the CID Number. This will define the caller ID text that is displayed when this user calls other (internal) users on the same PBX. This could be used when a user is part of a department in which callbacks should be directed to the department rather than directly to the user (such as a technical support department). This field is not mandatory. If the field is left blank, the user’s extension will be used to set the Outbound Caller ID text.
  • External CID, external Caller ID for the extension, consisting of two parts: the CID Name and the CID Number. This will define the caller ID text that is displayed when this user makes calls outside of the PBX. This could be used when a user is part of a department in which callbacks should be directed to the department rather than directly to the user (such as a technical support department). Setting the caller ID must be supported by the trunk service provider. This field is not mandatory, but if the field is left blank, the default caller ID name for the trunk placing the call will be used to set the caller ID name text.
  • Emergency CID, it allows to define the caller id that will be used in case of calling an emergency number.
  • Account Code, this field is used to populate the Account Code field of the Call Detail Record (CDR). If the field is left blank, the Account Code field of the CDR record will also be blank.
  • Language, specifies the language setting to be used for this extension. This will force all prompts specific to the user to be played in the selected language, provided that the language is installed, and voice prompts for the specified language exist on your server. This field is not required. If left blank, prompts will be played in the default language of the VitalPBX server.

Devices section

This section allows you to configure the device that is linked to the extension.

Technology, type of technology used by this device. The technology options are:

  • PJSIP, PJSIP device
  • SIP, SIP device
  • IAX2, IAX device
  • FXS, analog device
  • NONE, extension without device.
PJSIP
  • User Device*, username to be used when registering this device.
  • Password, password (secret) associated with this device. Passwords can be the weakest link on any externally accessible PBX system, as malicious users will attempt to locate extensions having weak passwords. Extensions that authenticate by using simple passwords such as “1234” stand a good chance of being compromised, allowing an attacker to place calls through your PBX. Pick strong passwords carefully and ensure that passwords are not given to anyone who does not need to know them. Passwords should be at least 8 characters long and should include a random mixture of letters (both upper- and lower-case), numbers, and special characters.
  • Profile, group of settings for this device. Each technology (PJSIP, SIP, IAX2, DAHDi, or None) must have at least one (default) profile that defines attributes for the technology. You can configure these profiles in the Settings->Technology Settings->Profiles menu.
  • Max Contacts, maximum number of contacts that can bind to an AoR.
  • Codecs, list of allowed codecs. The order in which the codecs are listed determines their order of preference. If you select at least one codec, the DISALLOW=ALL parameter will be added. This will ensure that the device will only use only the codecs that you specifically define for the device.
  • DTMF Mode, sets default dtmf-mode for sending Dual Tone Multi-Frequency (DTMF). The DTMF mode for a SIP device specifies how touchtone will be transmitted to the other side of the call. The default value is rfc4733. Available options are:
    • Rfc4733
    • info: SIP INFO messages (application/dtmf-relay)
    • shortinfo: SIP INFO messages (application/dtmf)
    • inband: Inband audio (requires 64 kbit codec -alaw, ulaw)
    • auto: Use rfc4733 if offered, in-band otherwise
  • Device Description, a short (optional) description to identify this device.
  • Emergency CID, this is the CID to be used when an Emergency Call is placed with this device.
  • Dispatchable Location, this is the location to be used whenever an Emergency Call is placed from this device.
  • Deny, in a user/peer definition, allows you to limit SIP traffic to and from this peer to a certain IP or network. This option should be in the format of an IP address and subnet, such as 192.168.25.10/255.255.255.255 (denies traffic from this specific IP address), or 192.168.1.0/255.255.255.0 (to disallow traffic for this extension from the IP range of 192.168.1.1 to 192.168.1.254). It is possible to enter a value of 0.0.0.0/0.0.0.0 to deny all the networks by default, and, to enter specific networks from which traffic can be accepted in the permit option. This option is commonly used to restrict endpoint usage to a particular network, so that if the endpoint is stolen or otherwise removed from the network, it cannot be used to place calls and will be essentially useless. This field is not required. If it is left blank, VitalPBX will not block traffic for this peer from any IP address.
  • Permit, in a user/peer definition, allows you to limit SIP traffic to and from this peer to a certain IP or network. For example, 192.168.10.0/255.255.255.0 allows traffic from any address on the 192.168.10.x network. The permit option is the opposite of the deny option. Specific IP addresses or networks can be added in this option to allow traffic for this extension from the entered IP/network. This field is not required. If it is left blank, traffic will be allowed from all IP addresses. Strengthen your system security by use of the deny and allow options, where possible. If the endpoint is static, we strongly recommend that you make proper use of the permit and deny options to ensure that traffic is only allowed from the specific address. Even if the endpoint is not static, but always resides on a known subnet, you should limit the allowed range to that specific subnet.
  • Ring Device, this determines whether incoming calls should cause the device to ring.
  • Send Push, if this is enabled, the PBX will attempt to send push notifications to mobile devices using VitXi Mobile.
  • VitXi Client, if this is enabled, this device can be used for VitXi Applications, be it WebRTC or Mobile Apps. When devices with this option enabled are use on VitXi Mobile, all of the premium features will be enabled on the softphone.
SIP
  • User Device*, username to be used when registering this device.
  • Password, password (secret) associated with this device. Passwords can be the weakest link on any externally accessible PBX system, as malicious users will attempt to locate extensions having weak passwords. Extensions that authenticate by using simple passwords such as “1234” stand a good chance of being compromised, allowing an attacker to place calls through your PBX. Pick strong passwords carefully and ensure that passwords are not given to anyone who does not need to know them. Passwords should be at least 8 characters long and should include a random mixture of letters (both upper- and lower-case), numbers, and special characters.
  • Profile, group of settings for this device. Each technology (SIP, IAX2, DAHDi) must have at least one (default) profile that defines attributes for the technology. You can configure these profiles in the Settings->Technology Settings->Profiles menu.
  • Codecs, list of allowed codecs. The order in which the codecs are listed determines their order of preference. If you select at least one codec, the DISALLOW=ALL parameter will be added. This will ensure that the device will only use only the codecs that you specifically define for the device.
  • NAT, (Network Address Translation) is a technology 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 Options:
    • Default: will use the default NAT settings set on the SIP Settings.
    • No: No special NAT handling other than RFC3581
    • Force: Pretend there was a rport parameter even if there wasn’t
    • Comedia: Send media to the rport 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
  • DTMF Mode, sets default dtmf-mode for sending Dual Tone Multi-Frequency (DTMF). The DTMF mode for a SIP device specifies how touchtone will be transmitted to the other side of the call. The default value is rfc2833. Available options are:
    • 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 if offered, in-band otherwise
  • Device Description, a short (optional) description to identify this device.
  • Deny, in a user/peer definition, allows you to limit SIP traffic to and from this peer to a certain IP or network. This option should be in the format of an IP address and subnet, such as 192.168.25.10/255.255.255.255 (denies traffic from this specific IP address), or 192.168.1.0/255.255.255.0 (to disallow traffic for this extension from the IP range of 192.168.1.1 to 192.168.1.254). It is possible to enter a value of 0.0.0.0/0.0.0.0 to deny all of the networks by default, and, to enter specific networks from which traffic can be accepted in the permit option. This option is commonly used to restrict endpoint usage to a particular network, so that if the endpoint is stolen or otherwise removed from the network, it cannot be used to place calls and will be essentially useless. This field is not required. If it is left blank, VitalPBX will not block traffic for this peer from any IP address.
  • Permit, in a user/peer definition, allows you to limit SIP traffic to and from this peer to a certain IP or network. For example, 192.168.10.0/255.255.255.0 allows traffic from any address on the 192.168.10.x network. The permit option is the opposite of the deny option. Specific IP addresses or networks can be added in this option to allow traffic for this extension from the entered IP/network. This field is not required. If it is left blank, traffic will be allowed from all IP addresses. Strengthen your system security by use of the deny and allow options, where possible. If the endpoint is static, we strongly recommend that you make proper use of the permit and deny options to ensure that traffic is only allowed from the specific address. Even if the endpoint is not static, but always resides on a known subnet, you should limit the allowed range to that specific subnet.
  • Ring Device, this determines whether incoming calls should cause the device to ring.
IAX2
  • User Device*, username to be used when registering this device.
  • Password, password (secret) associated with this device. Passwords can be the weakest link on any externally accessible PBX system, as malicious users will attempt to locate extensions having weak passwords. Extensions that authenticate by using simple passwords such as “1234” stand a good chance of being compromised, allowing an attacker to place calls through your PBX. Pick strong passwords carefully and ensure that passwords are not given to anyone who does not need to know them. Passwords should be at least 8 characters long and should include a random mixture of letters (both upper- and lower-case), numbers, and special characters.
  • Profile, group of settings for this device. Each technology (SIP, IAX2, Telephony, or None) must have at least one (default) profile that defines attributes for the technology. You can configure these profiles in the Settings->Technology Settings->Profiles menu.
  • Codecs, list of allowed codecs. The order in which the codecs are listed determines their order of preference. If you select at least one codec, the DISALLOW=ALL parameter will be added. This will ensure that the device will only use only the codecs that you specifically define for the device.
  • Device Description, a short (optional) description to identify this device.
  • Deny, in a user/peer definition, allows you to limit SIP traffic to and from this peer to a certain IP or network. This option should be in the format of an IP address and subnet, such as 192.168.25.10/255.255.255.255 (denies traffic from this specific IP address), or 192.168.1.0/255.255.255.0 (to disallow traffic for this extension from the IP range of 192.168.1.1 to 192.168.1.254). It is possible to enter a value of 0.0.0.0/0.0.0.0 to deny all of the networks by default, and, to enter specific networks from which traffic can be accepted in the permit option. This option is commonly used to restrict endpoint usage to a particular network, so that if the endpoint is stolen or otherwise removed from the network, it cannot be used to place calls and will be essentially useless. This field is not required. If it is left blank, VitalPBX will not block traffic for this peer from any IP address.
  • Permit, in a user/peer definition, allows you to limit SIP traffic to and from this peer to a certain IP or network. For example, 192.168.10.0/255.255.255.0 allows traffic from any address on the 192.168.10.x network. The permit option is the opposite of the deny option. Specific IP addresses or networks can be added in this option to allow traffic for this extension from the entered IP/network. This field is not required. If it is left blank, traffic will be allowed from all IP addresses. Strengthen your system security by use of the deny and allow options, where possible. If the endpoint is static, we strongly recommend that you make proper use of the permit and deny options to ensure that traffic is only allowed from the specific address. Even if the endpoint is not static, but always resides on a known subnet, you should limit the allowed range to that specific subnet.
  • Ring Device, this determines whether incoming calls should cause the device to ring.
FXS – (Only available if the DAHDI add-on is installed)
  • Channel*, the Telephony (DAHDI) channel, selected from the drop-down list, that should be associated with this device.
  • Profile, group of settings for this device. Each technology (SIP, IAX2, Telephony, or None) must have at least one (default) profile that defines attributes for the technology. You can configure these profiles in the Settings->Technology Settings->Profiles menu.
  • Device Description, a short (optional) description to identify this device.
  • Ring Device, this determines whether incoming calls should cause the device to ring.
NONE
  • Extensions that do not have a device, used for virtual voicemail or Hot Desking.

Voicemail

VitalPBX Extensions Voicemail

  • Enabled, enable or disable voicemail. If voicemail is not enabled, voicemail messages cannot be left for the user.
  • Attach Voicemail, Attach voicemail to email.
  • Delete, the voicemail is deleted from the server after the voicemail has been delivered. Be careful with this option, because VitalPBX will allow you to delete the message without guaranteeing that a copy of it has been attached to the email notification, or that the email has been delivered successfully. This could mean that after a message is left and a notification email is sent to the user, the actual voicemail that was left may no longer be accessible.
  • Voicemail Password, the numeric password to access the voicemail. The voicemail system will compare the password entered by the user against this value.
  • Zone Messages, time zone for messages. If not set, the time zone will be taken from the general settings section. Irrelevant if envelope is no.
  • Alias, an alternative name that can be used in the system-created phonebook, or for dialing using the Phonebook Directory feature code (411)
  • Allow to Call Back, if checked, users will be available to call back to the sender of a message. The specified Class of Service will need to be able to handle dialing of numbers in the format in which they are received (for example, the country code may not be received with the caller ID but might be required for the outgoing call).
  • Ask Password, it allows to define if the users who dials *97 to access to their own voicemail will be prompted to enter its voicemail password or not. This doesn’t apply for the “Remote Voicemail (*98)” feature.
  • Skip Instructions, if set to yes, it will skip the playback of instructions for leaving a message to the calling party.
  • Say CID, system will play back the caller ID number of the person who left the message prior to the message being played.
  • Say Duration, turn on/off the duration information before playing the voicemail message.
  • Envelope, this determines whether the user will hear the date and time that the message was left prior to hearing the voicemail message being played.
  • Hide from Directory, hide If set to yes, this name of this user will not be visible to the system-created phonebook, and you cannot dial to this user using the Phonebook Directory feature code (411).
  • Allow to Dial Out, if allowed, users can dial out from their mailboxes (option 4 from mailbox’s advanced menu). This is considered a very dangerous practice in a phone system (mainly because many voicemail users like to use 1234 as their password) and is therefore not recommended.
  • Generate Hint, if enabled, it will be possible to remotely monitor the voicemail status of this extension through a BLF key. To configure the BLF, you must to use the following format: vm_1234, where 1234 is the extension that will be monitored.

Recording

VitalPBX Extensions Recording

In this tab you will find the information about the recording telephone calls and dictation recording.

This group of fields allows a user to control the recording of incoming or outgoing calls. The user can either dial a feature code (*3) to selectively enable recording for the current call, never record calls, or always record calls.

  • Outgoing, record external outgoing calls.
  • Incoming, record external incoming calls.
  • Internal, record internal calls.
  • On Demand Recording, record calls on demand.
  • Dictation section
    • Enabled, activates the dictation service when set to Yes.
    • Format, recording audio format:
      • OGG Vorbis
      • GSM
      • WAV
    • Auto-Send Email, recording will be sent automatically once completed.

Advanced

VitalPBX Extensions Advanced

  • Ring Time, the number of seconds to ring the device before giving up and moving on to the next priority for the extension.
  • Call Limit, maximum number of simultaneous calls that can be received by this device.
  • Dial Profile, there are many options that you can set on the outbound call, including call screening, distinctive ringing, and more. Goto Settings/Technology/Dial Profile for more information.
  • Internal Auto Answer, automatic call answering can be requested from within the incoming call by using the SIP Alert-Info header. This can only be utilized when automatic call answering is allowed on the phone.
  • Music on Hold Class, this option specifies which music on hold class to suggest to the peer channel when this channel places the peer on hold.
  • Secretary Extension, functionality is used to re-route all incoming calls for this extension to the secretary’s extension. Only the secretary is allowed to make direct calls to this extension.
  • Caller ID on Diversions, this allows you to define which CID will be sent when the call is forwarded.
    • Caller, this will send the CID of the person calling.
    • Called, this will send the CID of the person receiving the call and has the diversion activated.
  • Fax Enabled, Enable/Disable fax.
  • Diversions Hints, this generates hints regarding status of the extension. For example, hints could be generated for diversions (DND, Call Forwarding, Personal Assistant and Boss/Secretary). Do not activate this option unless your phone has a console or keys for Hints. Activating this option can slow down the “Apply Changes” in the PBX and overload.
  • Block Spy Me, do not let other users to spy on this extension.
  • Send Caller ID, send, or hide, the Caller ID for this extension.
  • Call Waiting, if you uncheck this option, only one incoming call will be allowed to this extension.
  • Pin less, if enabled, the user of this extension will not be prompted to enter pin on outbound routes that have assigned a pin set.
  • Dynamic Routing, this allows you to enable or disable the dynamic routing towards this extension. If enabled, when an external party (that was previously called by this extension) calls back, the call will be routed directly to this extension.
Call Center Settings

This section contains two fields (Dynamic Queues, Static Queues) that allows you to assign or remove massively an extension to any queue or group of queues

  • Dynamic Queues, these are the agents who will be allowed to log in the call queue.
  • Static Queues, are agents that will always be in the queue, these agents do not need to log in.
User Portal
  • Enable Portal, allow users to login to Portal to configure their own extension.
  • Portal User, user for login as portal user.
  • Portal Password, password for access to portal area.
  • Fax Device, this option assigns a fax device to this extension, so they can send and receive faxes from the user interface using the Virtual Faxes add-on.
User Image section

Allows the user to select any image and associate it with the extension. It may be the photo of the owner of the extension, an avatar, or any other graphic; in PNG, JPG, or JPEG format. The size of the file must be less than 20 MB.

 

Follow Me

VitalPBX Extension Follow Me

  • Follow Me List, list of extensions and/or external numbers to be accessed by follow me.
  • Initial Ring Time, time in seconds to ring the primary extension before calling to the members on the follow-me list.
  • Ring Time, this is the time that the phone will be allowed to ring, without being answered, before continuing to an alternative destination.
  • Ring Strategy, here you define the strategy to ring this list.
  • One by One: ring all available number in the Follow List One by One.
  • Ring All: ring all available number in the Follow List at the same time.
  • Music on Hold, the Music on Hold class that should be used for the caller while they are waiting to be connected.
  • Call-from Prompt, you can select the default option to use the “Incoming call from” message prompt or use your own custom prompt.
  • No Recording Prompt, you can select to use the standard “You have an incoming call” message prompt when the caller elects not to leave their name or the option isn’t set for them to do so or use your own custom prompt.
  • Please Hold Prompt, you can select to use the standard “Please hold while we try and connect your call” message prompt or use your own custom prompt.
  • Status Prompt, you can select to use the standard “The party you’re calling isn’t at their desk” message prompt or use your own custom prompt.
  • Sorry Prompt, you can select to use the standard “I’m sorry, but we were unable to locate your party” message prompt or use your own custom prompt.
  • Enabled, it allows you to enable/disable the follow-me feature on this extension.
FollowMe Options
  • Record Caller’s Name, here you can record the caller’s name so it can be announced to the callee at each step.
  • Prompt Called, called party will be asked whether they wish to accept the incoming call.

Incoming Routes

VitalPBX Extensions Incoming Routes

The DID number for incoming calls, i.e. the inbound route that should be associated with this extension.

  • Description, a short description to identify the route.
  • DID Pattern, the DID number for incoming calls, i.e. the inbound route that should be associated with this extension.
  • CID Pattern, optional CID number to make route more specific.
  • Actions, go to Inbound Route module.
Welcome Email

Starting from Version 3 of VitalPBX, it is now possible to send a welcome email with various information about the new user extension. For this to work, we need to have the following configured.

  • Under Settings > PBX Settings > System General, enable the option “Send Welcome Email”.
  • Configure the email client with the account to send emails with, under Admin > System Settings > E-Mail Settings.
  • Make sure that the extension you created has an email assigned on the General Tab.
  • Note: If you wish to modify the contents of this email, you can do so under Admin > System Settings > Email Templates.

You can also see the QR code or send the Welcome Email Manually by pressing the QR button next to the Device Field.

VitalPBX Extensions Device QR Code

VitalPBX Extensions Send QR Code

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