Extensions

Extensions

Extensions

Extensions are associated with all UADs/Phones registered to the current slave tenant. On Multi Tenant EdgelessPBX, this menu item is only available when you are editing a tenant, since the master tenant is used for controlling the system behavior and tenants functionality.

Contents

  1. System
    1. Search
    2. Add/Edit Extension
    3. Adding Multi Extensions
    4. Advanced Options
    5. Permissions
  2. Ring Groups
    1. Add/Edit Ring Group
    2. Advanced Options
  3. Departments
    1. Add/Edit Department

System


System Extensions lists all local and remote UADs/Phones connected to the current tenant with the following details:

  • Name

    Full name of the user to which the device is registered
    (ex. Peter Doyle)
    (Display)

  • Extension

    UAD/Phone extension number
    (ex. 1111)
    (Display)

  • User Agent

    UAD/Phone type
    (ex. Yealink T38P)
    (Display)

  • Status

    UAD/Phone system status
    (ex. Active/Inactive)
    (Display)

  • Procotol

    Protocol used by the UAD/Phone
    (ex. SIP/IAX)
    (Display)

  • Edit

    Edit UAD/Phone configuration
    (Button)

  • Delete

    Delete UAD/Phone from the system
    (Button)

Search


The search bar filters extensions by name, e-mail, and number.

  • Search

    Provide a search phrase here and hit enter to filter the records.
    ([a-z][0-9])

  • Name

    The search filter should be applied to the names UADs are registered to.
    Check the box to search the names.
    (Check box)

  • E-mail

    The search filter should be applied to email addresses associated with the UADs.
    Check the box to search the email addresses.
    (Check box)

  • Number

    The search filter should be applied to extension numbers.
    Check the box to search extension numbers.
    (Check box)

  • MAC

    The search filter should be applied to MAC numbers of UADs.
    Check the box to search for MAC numbers
    (Check box)

Add/Edit Extension


The procedure for adding a new system extension is divided into two steps. In the first step, the UAD/Phone type and extension location need to be provided. In second step, basic UAD/Phone information such as user's name and email address is provided.

TIP:
By default, a 'Single Extension' will be created. 'Advanced Options' offer the facility to add multiple extensions as well. For more information, check the 'Adding Multi Extensions' chapter.

  • UAD (User Agent Device)

    Select the model of the new system UAD/Phone.
    If the UAD/Phone is not listed here, navigate to 'Settings: UAD' Edit the desired UAD/Phone and set its 'Status' to 'Active'. Now, the UAD/Phone will be available in this list.
    (ex. Linksys SPA-941)
    (Select box)

  • Location

    Select the location of the new UAD/Phone. Location refers to whether the UAD/Phone is in 'Local' or 'Remote' network.
    (ex. Local/Remote)
    (Select box)

In the second step, basic UAD/Phone information is set.
TIP:
Since this is an extension on a tenant you will see that the Username is prefixed with a tenant code, which is required for a UAD/Phone to register to the system. Nevertheless, when you register you will be able to dial other users on the tenant with only their extension number.

  • Extension

    System extension number
    By default, this field is automatically populated, but can be changed to any Extension number.
    (ex. Setting '1008' here will create a new system extension with the same network number. By default, this field is automatically populated, but can be changed to any Extension number).
    ([0-9])

  • Name

    Full name of the person using the Extension. This name is sent in a Caller ID information For example, setting name 'Joanna Cox' in this field will display the name on the other UAD/Phone display when the call is made.
    ([a-z][0-9])

  • E-mail

    Email address associated with the extension and used for various system notifications
    (ex. Setting 'joanna@domain.com' here will transfer all Voicemail notifications, Extension PIN and other details to this email)
    ([a-z][0-9])

  • Department

    Department to which extension will belong to. This is used so the Bicom Systems gloCOM can group extensions depending on which department they belong to.
    (ex. Select "Sales" and when sorted in gloCOM, this extension will be shown in Sales department group).
    (Select box)

  • Billing:

    Turn Billing on or off for the current extension
    (ex. Yes, No, N/A)
    (Option buttons)

  • Service Plan

    Service plan applied to the extension
    (ex. Select among available service plans to apply its rates to the extension)
    (Select box)

  • Slave

    Set whether two extensions should share the same billing funds. If ext 1000 has 100.00 in credit, enable this option and set 'Slave Account Code'='1001'. Now, any call made by these two extensions will take the credit off the 1000 extension.
    (Option buttons)

  • Username

    Username used by the UAD/Phone for the registration with the EdgelessPBX MT
    By default, this field is the Extension network number prefixed with tenant code and cannot be changed.
    (ex. In this case this value is set to '30010008').
    ([0-9])

  • Secret

    Secret/Password used by the UAD/Phone for registration with the EdgelessPBX MT
    By default, this field is automatically populated but can be changed to any value
    (ex. t8C1OGvK)
    ([a-z] [0-9])

  • PIN (Personal Identification Number)

    Four digit number used for account authorization.
    NOTE: This number must always be four (4) digits long
    (ex. If the PIN for this extension is set to '8474', provide it when asked for it by the EdgelessPBX MT when checking your Voice inbox or other Enhanced Services)
    ([0-9])

TIP:

  • After the extension is created, the 'Permissions' group will be editable for the administration.
  • Do not paste a value to the 'Name' and 'Email' fields, but please type it in. If these values are pasted, 'Advanced options' will need to be opened and the system will prompt for missing values.
  • Once the extension is created, the 'Save & Email' button becomes available. This command sends Extension details on the provided 'E-mail' address.

Adding Multi Extensions


There are two ways to add multiple extensions to EdgelessPBX MT:

  • Manually

    To create a list, manually provide details to 'Name', 'Email', 'Ext', 'Secret', 'PIN' [, 'MAC'] fields and click on the Add(+) icon.
  • Uploading a '.csv' file

    To Upload a '.csv' file:
    • Open a text editor on your desktop
    • Add the following lines, for example ('Name', 'Email', 'Exit', 'Secret', 'PIN', [, 'MAC'])

      John Doe,john@domain.com,4444,1234,4444
      Joanna Cox,joanna@domain.com,5555,2345,5555
    • Save file as 'ext.csv'
    • Click on the 'Browse' button
    • Select 'ext.csv' from the Desktop
    • Click on the 'Upload' button
    • New Extensions will be created
  • Single/Multiple Extension

    Switch between the single and multi extension adding options
    (Option button)

Advanced Options



Klicking Advanced Options button will show advanced configuration options and fields that were hidden previously.

  • UAD

    UAD (User Agent Device) are various IP phones, soft phones, ATA (Analog Telephone Adaptors), and IAD (Integrated Access Devices) used for system extensions. EdgelessPBX supports a wide range of UAD using SIP, IAX, MGCP, and ZAPTEL protocols. In case your phone is in supported UAD list, and UAD is enabled on your EdgelessPBX, you will be able to choose UAD that match the phone registering to the extension.

  • Location

    This option is related to Auto Provisioning function of EdgelessPBX, extensions located in same LAN as EdgelessPBX have to be set to Local while extensions connecting to EdgelessPBX from WAN should be set to Remote.

General



The following options are used frequently and are mostly required for normal extension operation. Some of these fields are pre-configured with the default values. It is not recommended to change these unless prompted to do so while saving the changes.

  • Extension

    Extension number. Number you dial on in order to reach local EdgelessPBX user associated with this extension.

  • Title

    Users title like Mrs, Mr, and such
    (ex. Mrs)
    ([a-z])

  • Name

    Name of user associated with extension e.g. John Lebowski.

  • E-mail

    E-mail address associated with user of the extension. For example, this e-mail address can be used to send out extensions account details to the user.

  • Location

    User's location
    (ex. Location like Street, City, or State)
    ([a-z][0-9])

  • Department

    Department in company with which user is associated with e.g. Development.
  • User Type

    Extensions can be set to make calls only, receive calls only or both make and receive calls
    (ex. Select 'User' to make the calls only; 'Peer' to receive the calls only; or 'Friend' for both, to make and receive calls)
    (Select box)

  • DTMF Mode (Dual Tone Multi-Frequency)

    A specific frequency, consisting of two separate tones. Each key has a specific tone assign to it so it can be easily identified by a microprocessor.
    This is a sound heard when dialing digits on touch-tone phones. Each phone has different 'DTMF Mode'.
    (ex. By default, this field is populated automatically for supported devices. If adding other UAD/Phone select between 'inband', , 'rfc2833' or 'info' options)
    (Select box)

  • RFC2833 Compensate (1.2):

    Compensate for pre-1.4 DTMF transmission from another Asterisk machine. You must enable this option or DTMF reception will not work.
    (ex. Yes, No, N/A)
    (Option Buttons)

  • Context

    Every system extension belongs to a certain system context. Context may be described as a collection/group of extensions. Default context used by the EdgelessPBX MT per tenant is 't-XXX' (where XXX is tenant number) and cannot be changed.

  • Status

    Extension status/presence on the network.
    Rather than deleting the extension and then recreating it again later on, the extension can be activated/deactivated using this field.
    (ex. Setting this field to 'Not Active' will disable all calls to this extension).
    Options:

    Active - Extension is active, it can make and receive calls.
    Not Active - Extension is not active and it can't make nor receive calls.
    Suspended - Extension is suspended and can't make calls to numbers other than those defined as Emergency Service numbers in Settings -> Servers -> Edit Server -> Locality (section) -> Emergency Services,

    (Select box)

  • Show In Directory:

    Whether the extension should be shown in the directory or not
    (ex. Yes, No, N/A)
    (Option buttons)

Authentication



These options are used for UAD/Phone authentication with EdgelessPBX MT

  • Username

    Username used by the UAD/Phone for the registration with EdgelessPBX MT
    (ex. By default, this field is the same as the extension network number and cannot be changed. In this case, this value is set to '1008').
    ([0-9])

  • Authname

    Name used for authentication with the sip provider
    Example:
    If you set this field to 12345, for example, the sent SIP header will look like 12345@sipprovider.com, for example
    ([0-9])

  • Auth

    Auth is the optional authorization user for the SIP server
    (ex. 44000)
    ([a-z][0-9])

  • Secret

    Secret/Password used by the UAD/Phone for registration with EdgelessPBX MT
    (ex. By default, this field is automatically populated, but can be changed to any value)
    ([a-z][0-9])



NOTE: In EdgelessPBX 3.8.2 we introduced strong password enforcement, which means that secret must meet certain criteria in order to be accepted otherwise, EdgelessPBX will display error message stating that secret is too weak.


Secret has to meet the following criteria in order to be accepted:

  • It must be at least 8 characters long
  • It must contain at least 1 uppercase
  • It must contain at least 1 lowercase
  • It must contain at least 1 digit
  • It must contain at least 1 special character ()
  • Allowed characters are: a-z, A-Z, 0-9, ! % * _ -

In order to make it easier for our users, we also implemented password generator, that will automatically generate strong password that meets above criteria with a single mouse click on a key icon located on a side of Secret field.

  • PIN (Personal Identification Number)

    Four digit number used for account authorization.
    NOTE: This number must always be four (4) digits long
    (ex. If the PIN for this extension is set to '8474', provide it when asked for it by EdgelessPBX MT when checking your Voice inbox or other 'Enhanced Services')
    (0-9)

IAX Extensions only

  • Auth Method

    Authorization method used for IAX extensions, can be set to:
    • none
    • plaintext
    • md5
    • rc4
    • rsa

  • RSA Key

    RSA key used for authorization, if preferred auth method is 'rsa' then RSA key needs to supplied to this field.
    ([0-9][a-z][A-Z])

  • Encryption

    Whether to enable encryption of IAX data stream. For this to work, you must choose 'md5' auth method, for example Yes, AES-128.

Billing


These options are used for billing of incoming and outgoing calls. The extension is assigned to a service plan and its call rates and additional billing options are set here as well.

  • Billing

    This option will enable/disable billing on extension.

  • Reset Inclusive

    Reset extension inclusive minutes, click on this button and confirm with 'Yes' to reset inclusive minutes.
    (Button)

  • Credit/Debit

    Opens a window for adding extension credit/debit.
    (Button)

Credit/Debit



  • Type

    Billing type, select whether billing is credit or debit.
    (Select box)

  • Amount

    Billing amount, if the billing type is in Euros, and you add 100 here, 100 Euros will be added to the extension amount.
    ([0-9])

  • Ref No

    Billing reference number, depending on how your company bills clients, the invoice number can be assigned here, for example.
    ([a-z][0-9])

  • Notes

    Additional billing notes.
    ([a-z][0-9])

  • Send

    This will finalize billing action, fill in all previous fields and click this button to add funds.
    (Button)

Once funds are added, the following details will be displayed:

  • Date: Time and date of the payment
  • User: The username used for login to the system of the user who added the funds
  • Ref No: Billing reference number
  • Notes: Additional billing notes
  • Amount: Amount of funds added
  • Type: Billing type

NOTE: Buttons Reset Inclusive and Credit/Debit will not be displayed unless Billing is enabled.

  • Slave

    Set whether two extensions should share the same billing funds. If ext 1000 has 100.00 in credit, enable this option and set 'Slave Account Code'='1001'. Now, any call made by these two extensions will take the credit off the 1000 extension.
    (Option buttons)

  • Master Account Code

    Set the master account code (extension number) from which the current extension is using funds. If ext 1000 has 100.00 in credit, enable this option and set 'Slave'='Yes' and set this field to '1001'. Now, any call made by these two extensions will take the credit off the 1000 extension.
    ([0-9])

  • Reminder Balance

    Account balance at which a reminder should be sent to the user. If this field is set to 10, the user will receive an email notification when the account balance reaches this amount.
    ([0-9])

  • Credit Limit

    The maximum amount that the system will extend to the billing account. If this field is set to '10' and the account balance has dropped down to '0', your account will still have '10' units in available funds.
    ([0-9])

  • Service Plan Date (dd-mm-YYYY)

    When the current Service Plan was set so inclusive minutes can be reset. If this field is set to 12-06-2008, inclusive minutes will be reset on 12th day of each month. So, if all 5 inclusive minutes were used by this day, inclusive minutes will be reset back to 5 minutes.
    ([dd-mm-YYY])

  • Enable Limits

    This option set the limits on the current extension to Yes, No, N/A.
    (Option buttons)

  • Limit Type


    This option set limits to be applied Daily or Monthly.
    (Select box)

  • Soft Limit

    Depending on Limit Type, when the extension reaches Soft Limit, it will email the person in charge of billing. Set 10 here if you want an email sent when the user hits that amount when calling.
    ([0-9])

  • Hard Limit

    Depending on the Limit Type, when an extension reaches Hard Limit, the system will block this extension from making any further calls. Set 20 here if you want the system to block this extension from making any calls.
    ([0-9])

  • Notification Email

    What email should be used when the user reaches its Soft Limit.
    (ex.admin@domain.com)
    ([a-z][0-9])

Billing Info


This information displays the extension's billing information: amount left, inclusive minutes left, etc.

  • Account Balance

    Displays the available account balance - the exact sum spent by the user.
    (ex.If the user has 100 units of credit, 100 units + the credit limit can be spent. If this amount displays a negative value (e.g. -4.00000) that means that the account balance has reached 0 and the credit limit is being used).
    (Display)

  • Available Funds

    Displays available account funds (account balance + credit limit).
    (ex. If the user account balance has 100 units + 10 credit limit units, 110 units will be displayed here).
    (Display)

  • Inclusive Minutes Left

    Displays the inclusive minutes left. As long as there is any inclusive time left, billing is not calculated for outgoing calls.
    (ex. You'll see the inclusive minutes left in the following form '0d 0h 4m 25s').
    (Display)

  • Creation Date

    Extension creation date.
    NOTE: If your system was updated to newer version, old extensions will have this field displaying 'unknown' and all new extensions will display extension creation date.
    (ex. 14-06-2007 12:30:36)
    (Display)

  • First Use Date

    Date/Time of the first extension use.
    (ex. 11 Jun 2007 18:58:25)
    (Display)

  • Last Use Date

    Date/Time of the last extension use.
    (ex. 11 Jun 2007 19:25:12)
    (Display)

Permissions


Destinations



These options grant/deny certain local/worldwide destinations, conferences, enhanced services, or call monitoring to your edited extension. If the image below is displayed, all destinations are allowed for the user extension. Should extension permissions be changed, click the 'Set destinations manually' button.

Manually, destinations are set through the following groups:


  • Remote - E164 PSTN destinations, ITSPs, other VoIP networks etc.
  • Local - All destinations within the system/network (Extensions, IVR, Queues, Conferences...).
  • Other Networks - Other PBX networks we are connected to.


Authorized

PIN Required

Not Authorized

Enhanced Services


Enhanced Services allows users to fully adjust settings like Caller ID, Call Pickup, Call Filters & Blocking, Call Forwarding etc.
For detailed information on Enhanced Services click the link below:
Enhanced Services

Network Related


These options set important network related values regarding NAT, monitoring and security.

  • Transport:

    Type of transfer protocol that will be used on EdgelessPBX.

    UDP (User Datagram Protocol) - With UDP, computer applications can send messages, in this case referred to as datagrams, to other hosts on an Internet Protocol (IP) network without prior communications to set up special transmission channels or data paths.
    TCP (Transmission Control Protocol) - provides reliable, ordered, error-checked delivery of a stream of octets between programs running on computers connected to an intranet or the public Internet.
    TLS (Transport Layer Security) - cryptographic protocol that provide communication security over the Internet.[1] They use asymmetric cryptography for authentication of key exchange, symmetric encryption for confidentiality, and message authentication codes for message integrity.

    Type: Checkbox

  • Encryption:

    This option enables or disables encryption in EdgelessPBX transport.
    Options: Yes, No, N/A.

  • NAT (Network Address Translation)

    Set the appropriate Extension - EdgelessPBX NAT relation.
    If extension 1000 is trying to register with the EdgelessPBX from a remote location/network and that network is behind NAT, select the appropriate NAT settings here:
    • yes - Always ignore info and assume NAT
    • no - Use NAT mode only according to RFC3581
    • Default (rport) - this setting forces RFC3581 behavior and disables symmetric RTP support.
    • Comedia RTP - enables RFC3581 behavior if the remote side requests it and enables symmetric RTP support.
    (Option buttons)

  • Direct Media

    This option tells the Asterisk server to never issue a reinvite to the client, if it is set to No. Select Yes if you want Asterisk to send reinvite to the client.
    (Option button)

  • Direct RTP setup:

    Here you can enable or disable the new experimental direct RTP setup. Setting this value to yes sets up the call directly with media peer-2-peer without re-invites. Will not work for video and cases where the callee sends RTP payloads and fmtp headers in the 200 OK that does not match the callers INVITE. This will also fail if directmedia is enabled when the device is actually behind NAT.
    Options: Yes, No, N/A

  • Qualify

    Timing interval in milliseconds at which a 'ping' is sent to the UAD/Phone or trunk, in order to find out its status(online/offline). Set this option to '2500' to send a ping signal every 2.5 seconds to the UAD/Phone or trunk. Navigate to 'Monitor: Extensions' or 'Monitor: Trunks' and check the 'Status' field.
    ([0-9])

  • Host

    Set the way the UAD/Phone registers to EdgelessPBX. Set this field to 'dynamic' to register the UAD/Phone from any IP address. Alternately, the IP address or hostname can be provided as well.
    ([dynamic][a-z][0-9])

  • Default IP

    Default UAD/Phone IP address. Even when the 'Host' is set to 'dynamic', this field may be set. This IP address will be used when dynamic registration could not be performed or when it times out.

    NOTE: UAD/Phone must be on static IP address.

    ([0-9])
  • Use RTP source address for T.38 packets (1.2)

    Use the source IP address of RTP as the destination IP address for UDPTL packets if the nat option is enabled. If a single RTP packet is received Asterisk will know the external IP address of the remote device. If port forwarding is done at the client side then UDPTL will flow to the remote device.
    (ex. Yes, No, N/A)
    (Option Buttons)

Caller ID



The caller's name and number displayed here are sent to the party you call and are shown on their UAD/Phone display. The information you see here is taken from the extension number and user name. To set different Caller ID information, please go to 'Enhanced services: Caller ID' and set new information there.


  • Set Caller ID

    Enable 'Caller ID' service
    (ex. Set this option to 'Yes' to enable the Caller ID service)
    (Option buttons)

  • Caller ID

    Extension Number and Name that are displayed on dialed party UAD/Phone display
    (ex. These options are read-only. Caller ID information can be changed only through 'Enhanced Services')
    (Read-only)

  • Caller ID Presentation

    The way Caller ID is sent by the Extension
    If EdgelessPBX MT is connected to a third-party software and there are problems with passing the Caller ID information to it, applying different 'Caller ID Presentation' methods should sort out the problem
    (ex. Presentation Allowed, Not Screened)
    (Select box)

  • Hide CallerID for Anonymous calls

    When you set this option to Yes, incoming calls with Anonymous as number but have CallerID set, are then formatted as Anonymous
    .
    (ex. Yes, No, N/A)
    (Option buttons)

  • Ringtone for Local calls

    If you know which phone is registered on this extension, you can set a custom ring tone for local calls.
    (ex. If your phone is SPA941 you could set )
    ([a-z][0-9])

  • Only Allow Trunk CallerID within DID range

    When you assign an extension to a customer and assign some DIDs to it, customer can make calls through that extension with CallerIDs that match its DID numbers. If a customer tries to make a call with a CallerID that doesn't match any of the DIDs assigned to him, the call will not be allowed.
    (ex. Yes, No, N/A)
    (Option buttons)

Call Properties



These options fine-tune incoming/outgoing call settings.

  • Ringtime

    UAD/Phone ring time.

Example:

Time in seconds that the UAD/Phone will ring before the call is considered unanswered (default: 32).
([0-9])

  • Incoming Dial Options

    Advanced dial options for all incoming calls.
    Example:
    Please see below for a detailed list of all available dial options (default: tr).
    ([a-z])

  • Outgoing Dial Options

    Advanced dial options for all outgoing calls.
    Example:
    Please see below for a detailed list of all available dial options (default: empty).
    (a-z)

  • VoiceMaster PIN

    This is a PIN number that is issued along with a dial string to the VoiceMaster system.
    (ex. 1234)
    ([0-9])

Dial Options:

  • t - Allow the called user to transfer the call by hitting #
  • T - Allow the calling user to transfer the call by hitting #
  • r - Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one answers. Use with care and don't insert this by default into all of your dial statements as you are killing call progress information for the user. Really, you almost certainly do not want to use this. Asterisk will generate ring tones automatically where it is appropriate to do so. 'r' makes it go the next step and additionally generate ring tones where it is probably not appropriate to do so.
  • R - Indicate ringing to the calling party when the called party indicates ringing, pass no audio until answered. This is available only if you are using kapejod's bristuff.
  • m - Provide Music on Hold to the calling party until the called channel answers. This is mutually exclusive with option 'r', obviously. Use m(class) to specify a class for the music on hold.
  • o - Restore the Asterisk v1.0 Caller ID behavior (send the original caller's ID) in Asterisk v1.2 (default: send this extension's number)
  • j - Asterisk 1.2 and later: Jump to priority n+101 if all of the requested channels were busy (just like behaviour in Asterisk 1.0.x)
  • M (x) - Executes the macro (x) upon connect of the call (i.e. when the called party answers)
  • h - Allow the called party to hang up by dialing *
  • H - Allow the caller to hang up by dialing *
  • C - Reset the CDR (Call Detail Record) for this call. This is like using the NoCDR command
  • P (x) - Use the Privacy Manager, using x as the database (x is optional)
  • g - When the called party hangs up, exit to execute more commands in the current context.
  • G (context^exten^pri) - If the call is answered, transfer both parties to the specified priority; however it seems the calling party is transferred to priority x, and the called party to priority x+1
  • A (x) - Play an announcement (x.gsm) to the called party.
  • S (n) - Hang up the call n seconds AFTER the called party picks up.
  • d: - This flag trumps the 'H' flag and intercepts any dtmf while waiting for the call to be answered and returns that value on the spot. This allows you to dial a 1-digit exit extension while waiting for the call to be answered - see also RetryDial
  • D (digits) - After the called party answers, send digits as a DTMF stream, then connect the call to the originating channel.
  • L (x[:y][:z]) - Limit the call to 'x' ms, warning when 'y' ms are left, repeated every 'z' ms) Only 'x' is required, 'y' and 'z' are optional. The following special variables are optional for limit calls: (pasted from app_dial.c)
    • + LIMIT_PLAYAUDIO_CALLER - yes|no (default yes) - Play sounds to the caller.
    • + LIMIT_PLAYAUDIO_CALLEE - yes|no - Play sounds to the called party.
    • + LIMIT_TIMEOUT_FILE - File to play when time is up.
    • + LIMIT_CONNECT_FILE - File to play when the call begins.
    • + LIMIT_WARNING_FILE - File to play as a warning if 'y' is defined. If LIMIT_WARNING_FILE is not defined, then the default behavior is to announce ('You have [XX minutes] YY seconds').
  • f - forces callerid to be set as the extension of the line making/redirecting the outgoing call. For example, some PSTNs don't allow Caller IDs from other extensions than the ones that are assigned to you.
  • w - Allow the called user to start recording after pressing *1 or what defined in features.conf, requires Set(DYNAMIC_FEATURES=automon)
  • W - Allow the calling user to start recording after pressing *1 or what defined in features.conf, requires Set(DYNAMIC_FEATURES=automon)

Groups



These options define who is allowed to pickup our calls, and whose calls we are allowed to pickup.

  • Call Group

    Set the Call Group that the extension belongs to. Similar to 'Context' grouping, only this option sets to which call group the extension belongs.
    (ex. 3)
    (Select box)

  • Pickup Group

    Set which groups the extension is allowed to pickup by dialing '*8'.
    (ex. 4)
    (Select box)

TIP: Grouping works only within a technology (SIP to SIP or IAX to IAX)

Example:

Extension A:

  • Call Group = 1
  • Pickup Group = 3,4

Extension B:

  • Call Group = 2
  • Pickup Group = 1
  • If A is ringing, B can pickup the ringing call by dialing '*8'.
  • If B is ringing, A cannot pickup the ringing call because B's Call Group = 2, and A can pickup only Call Groups 3 and 4.

NOTE: To be able to select Call Group and Pickup Group they have to be assigned to the tenant in Settings -> Tenants -> edit tenant -> Numbering Defaults (section) -> Call groups/Pickup groups.

Trunks



These options enable extensions to use custom default trunks for all outgoing calls.

  • Primary/Secondary/Tertiary Trunk:

    Set the default trunks for all routes dialed from this extension.
    If the connection is not established through the primary, the secondary trunk is used, etc. Default trunks can be set per extension and on the Settings->Default Trunk, on a Slave. Please look at the 'Precedence' section.
    (Select box)

  • Override System LCR

    This option tells the systems that when making calls, they should omit checking LCR.
    (Option buttons)

Call Control



These options set the number of simultaneous incoming and outgoing extension calls.

  • Incoming Limit:

    Sets the maximum number of simultaneous incoming calls. If an extension receives more incoming calls than set here, they are all redirected to the extension voice-box
    (ex. 2)
    ([0-9])

  • Outgoing Limit:

    Sets the maximum number of simultaneous outgoing calls. The outgoing call can be placed on hold and another call can be made from the same extension. However, this feature has to be supported by the UAD/Phone
    (ex. 2)
    ([0-9])

  • Play sound on exceeded limit:

    If you try to make more calls than allowed in the Outgoing Limit, you will be played a message that the limit has been exceeded.
    (ex. Yes, No, N/A)
    (Option button)

  • Send e-mail on exceeded limit:

    Whether or not to send a notification mail on the exceeded limit.
    (ex. Yes, No, N/A)
    (Option buttons)

  • Notification e-mail:

    E-mail address to which notification mail should be sent if the number of calls exceed the limit.
    (ex. user@domain.com)
    ([a-z][0-9] @)

IAX Extensions only



  • Notransfer

    Prohibit Asterisk from stepping out of the media path and connecting the two endpoints directly to each other. This, of course, affects your CDR and billing information
    (ex. Yes, No, N/A)
    (Option buttons)

  • Send ANI

    Whether to send ANI along with CallerID
    (ex. Yes, No, N/A)
    (Option buttons)

  • Trunk

    Whether to use IAX trunking. IAX Trunking needs support of a hardware timer
    (ex. Yes, No, N/A)
    (Option buttons)

Voicemail



These options mimic the functions of an answering machine but with many additional features added. Voice messages are saved on central file-system location instead on a UAD/Phone.

  • Accessing voice-box:

    To access a voice-box, dial '*123', enter the extension PIN, and follow the instructions.
  • Leaving a voice message:

    When the user is transferred to voice-box, 'Please leave your message after the tone. When done, hangup or press the # key' message will be heard. Two options are available:

    1. Leave a voice message (ended by pressing '#' key or by hanging up), or
    2. Reach an operator by dialing '0'
    If '0' is dialed, the 'Press 1 to accept this recording, otherwise please continue to hold' message will be heard. Two options are available:

    1. Press '1' to save your message, after which the operator will be dialed. The 'Please hold while I try that extension' message will be heard, or
    2. Continue to hold, which will delete any left messages, after which the operator will be dialed. 'Message deleted, please hold while I try that extension' message will be heard.
  • File - system usage:

    With continuous tone for 60 seconds:

    • wav49 = 91.0kb
    • wav = 863.0kb
    • gsm = 91.0kb
    With continuous silent tone for 60 seconds:

    • wav49 = 0.38kb
    • wav = 3.0kb
    • gsm = 0.32kb

  • Voicemail:

    Enable the Voicemail service.
    When the call is placed and no one picks up the handset after some time, the calling party will be transferred to the dialed extension voice box and offered to leave a voice message
    (ex. Yes, No, N/A)
    (Option buttons)

  • Greeting-Mode:

    If your Voicemail is turned on, you can set this option to yes to play a greeting and then a busy sound
    (ex. Yes, No, N/A)
    (Option buttons)

  • Mailbox

    Mailbox extension number
    (ex. This value is the same as the extension number and cannot be modified)
    (Read-only)

  • Name:

    Full name of the user associated with the voice box.
    (ex. This value is the same as the 'Name' field and cannot be modified).
    ([a-z][0-9])

  • PIN: (Personal Identification Number)

    Password used for accessing voicemail. The value of this field is set under 'Authentication: PIN'.
    (ex. When B wants to access his voicemail, he is asked to authenticate with personal 4(four) digit PIN).
    ([0-9])

  • E-mail:

    E-mail address associated with the voice box. The value of this field is set under 'General: E-mail'.
    (ex. When A calls B and leaves a voice message, B will get an email notification about new voice message on this email address).
    ([a-z] [0-9] [@._-])

  • Send e-mail

    Whether or not to send an e-mail to the address given above
    (ex. Yes, No, N/A)
    (Option buttons)

  • Pager e-mail:

    Pager e-mail address associated with the voice box.
    (ex. When A calls B and leaves a voice message, B will get a pager email notification about a new voice message on this email address).
    ([a-z] [0-9] [@._-])

  • Greeting message:

    Greeting message played to users upon entering the voice box.
    (ex. When A gets to B's voice box, the selected 'Greeting message' is played to A before he is allowed to leave a message).
    (Select box)

  • Skip Instructions:

    Skip the instructions on how to leave a voice message.
    (ex. Once user A reaches the dialed voice box, if this option is set to 'Yes', A will hear the 'Greeting message', and then be transferred directly to the 'beep' sound).
    (Option buttons)

  • Attach:

    Send the voice message as an attachment to the user's email.
    (ex. Once B gets the new voice message, if this option is set to 'Yes', the message sound file will be attached to the new voicemail notification email).
    (Option buttons)

  • Delete After E-mailing:

    Delete voice message after sending it as an attachment to the user's email.
    (ex. Once B gets the new voice message, if this option is set to 'Yes', the message will be deleted from the voice box after it has been emailed to B).
    (Option buttons)

  • Say Caller ID:

    Announce the extension number from which the voice message has been recorded.
    (ex. If this option is set to 'Yes', when checking voicemail, the 'From phone number {$NUMBER}' message will be heard).
    (Option buttons)

  • Allow Review mode:
    Allow B to review the voice message before committing it permanently to A's voice box.
    Example:
    B leaves a message on A's voice box, but instead of hanging up, he presses '#'. Three options are offered to B:
    • Press 1 to accept this recording
    • Press 2 to listen to it
    • Press 3 to re-record your message
    (Option buttons)

  • Allow Operator:

    Allow B to reach an operator from within the voice box.
    Example:
    B leaves a message on A's voice box, but instead of hanging up, B presses '#'.
    'Press 0 to reach an operator' message played (Once '0' is pressed, user is offered the following options):
    • Press 1 to accept this recording (If selected, 'Your message has been saved. Please hold while I try that extension' is played and operator is dialed)
    • Or continue to hold (If B holds for a moment, 'Message deleted. Please hold while I try that extension' is played and operator is dialed)
    (Option buttons)

  • Operatior Extension:

    Local extension number that acts as an operator.
    (ex. If A's voice box has an option 'Allow Operator' set to 'Yes', all users dialing '#0' inside the voice box will reach this operator extension).
    ([0-9])

  • Play Envelope message:

    Announces the Date/Time and the Extension number from which the message was recorded.
    (ex. Once the voice box is checked for new messages, if this option is set to 'Yes', 'Received at {$DATE} from phone number {$NUMBER}' will be played, giving more details about the message originator).
    (Option buttons)

  • Hide from directory:

    This option will allow you to hide your extension from the Directory/BLF list.
    (ex. Yes, No, N/A)
    (Option buttons)

  • Rings to answer

    Number of rings before Voicemail answers the call
    (ex. 5)
    ([0-9])

  • Voicemail Delay:

    How long to pause in seconds before asking the user for PIN/Password.
    (ex. Some UADs/Phones have a tendency to garble the beginning of sound files. Therefore, the user checking the voice box, when asked for a password, would hear '...sword' instead of 'Password'. Setting this field to 1-2 seconds will provide a long enough gap to fix this anomaly).
    ([0-9])

  • Timezone:

    Sets the correct date/time stamp.
    NOTE: Timezones are taken from '/usr/share/zoneinfo' system directory
    (ex. By setting the correct time zone, the user would always be notified of the exact date/time voice message was left on their box. Set the correct time zone if the user is located in a different time zone than EdgelessPBX MT).
    (Select box)

Speakerphone Page Auto-Answer SIP Header



These options allow the caller to use a UAD in a public announcement system. If the UAD fully supports this service, the call is accepted automatically and put on a loudspeaker.
Speakerphone Page Auto-Answer SIP header


  • Choose Device Type

    Set predefined UAD/Phone type for this extension.
    (ex. The header will be added automatically depending on the selected device).
    (Select box)

  • Custom Header

    Set a custom UAD/Phone header for this extension.
    (ex. If one of the predefined headers does not work, you might want to try setting a custom header for this service. The custom header line to be used 'Call-Info:;answer-after=0').
    ([a-z][0-9])

Codecs



Codecs are used to convert analog to digital voice signals and vice versa. These options set preferred codecs used by the extension.

TIP:
If some of the desired codecs is disabled (cannot be selected), navigate to 'Settings: Servers: Edit: Default Codecs' and enable them under the 'Local' group.

  • Disallow

    Set the codecs extension to not allowed to use.
    (ex. This field is very unique. In order to work properly, this setting is automatically set to 'Disallow All' and it cannot be modified).
    (Ready-only)

  • Allow

    Set the codecs extension to allowed to use.
    (ex. Only the codecs set under 'Settings: Server' will be available to choose from).
    (Check box)

  • Video Support

    Set this option to Yes to enable SIP video support.
    (Yes, No, N/A)

  • Force codec on outbound trunk channel

    With this option you can force codec use for outbound trunk calls.
    (ex. iLBC)
    (Select box)

  • Auto-Framing (RTP Packetization)

    If autoframing is turned on, the system will choose the packetization level based on remote ends preferences.
    (ex. If the remote end requires RTP packets to be of 30 ms, your EdgelessPBX system will automatically send packets of this size if this option is turned on. Default is set to 20 ms and also depends on the codecs minimum frame size like G.729 which has 10 ms as a minimum).
    (Option buttons)

Codecs:

  • ITU G.711 ulaw - 64 Kbps, sample-based, used in US
  • ITU G.711 alaw - 64 Kbps, sample-based, used in Europe
  • ITU G.723.1 - 5.3/6.3 Kbps, 30ms frame size
  • ITU G.726 - 16/24/32/40 Kbps
  • ITU G.729 - 8 Kbps, 10ms frame size
  • GSM - 13 Kbps (full rate), 20ms frame size
  • iLBC - 15Kbps,20ms frame size: 13.3 Kbps, 30ms frame size
  • Speex - 2.15 to 44.2 Kbps
  • LPC10 - 2.5 Kbps
  • H.261 Video - Used over ISDN lines with resolution of 352x288
  • H.263 Video - Low-bit rate encoding solution for video conferencing
  • H.263+ Video - Extension of H.263 that provides additional features that improve compression over packet switched networks.

Recording



This group of options is used for the recording of all incoming/outgoing calls.

TIP:

  • Laws in some countries may require notifying the parties that their call is being recorded.
  • Recorded calls, marked with icon, can be accessed from 'Self Care Interface' or 'Reports: CDR' EdgelessPBX' menu.
  • Call are recorded in audio format set under 'Settings: Servers: Recordings Format'.
  • Record Calls

    Enable call recording service. Select 'Yes' to enable the service. All incoming/outgoing calls will be recorded. If using call recording with many extensions, check server disk space from time to time. Please see below for bit rates table.
    (Options buttons)

  • Silent

    Set call recordings should be announced to the parties in a conversation. If Silent=No, calling parties will hear a 'Recorded' or 'This call is recorded' message before their conversation starts.
    (Options buttons)

Disk Space Used By Call Recording

With continuous tone for 60 seconds:

  • wav49 = 84.5kb
  • wav = 833.0kb
  • gsm = 85.0kb

With continuous silent tone (without sound) for 60 seconds:

  • wav49 = 84.0kb
  • wav = 827.0kb
  • gsm = 84.0kb

Auto Provisioning



These options enable EdgelessPBX MT to automatically provision the UAD/Phone. Configuration files are downloaded from EdgelessPBX MT's TFTP server

  • Auto Provisioning:

    Enable auto provisioning service for this extension
    (ex. Connect the UAD/Phone to EdgelessPBX MT without any hassle by providing UAD/Phone MAC address (and optionally adding the Static UAD/Phone IP address and network details))
    (Option buttons)

  • MAC Address (Media Access Control):

    UAD/Phone MAC address
    (ex. Provide the UAD/Phone address here. Its a 48-bit hexadecimal number (12 characters))
    ([a-z][0-9])

  • DHCP (Dynamic Hosts Configuration Protocol):

    Set whether the UAD/Phone is on DHCP or Static IP address
    (ex. Set DHCP = Yes if UAD/Phone is on dynamic or DHCP = No if UAD/Phone is on static IP address. If on static IP, you will have to provide more network details in the fields below).
    (Option buttons)

  • Static IP:

    Static UAD/Phone IP address
    (ex. DHCP = No, has to be set. Provide the UAD/Phone static IP address here)
    ([0-9][.])

  • Netmask:

    UAD/Phone netmask
    (ex. Netmask applied to UAD/Phone static IP address)
    ([0-9][.])

  • Gateway:

    Gateway IP address
    (ex. Local area network gateway IP address)
    ([0-9][.])

  • DNS Server1 and Server2 (Domain Name Server):

    DNS Server IP address
    (ex. Local area network DNS IP address (Usually the same as your gateway))
    ([0-9][.])

Presence



This option simply notifies you of whether device presence is enabled or disabled. Supported UADs can be seen in the Settings->UAD menu.

  • Presence Enabled:

    Returns the information whether the phone is on call, ringing, or offline (not registered).
    (ex. Select 'Yes' to enable presence support, but all UADs/Phones don't support this feature)
    (Option buttons)

  • Global Presence:

    Enables presence like above option but when this option is turned on, it will enable presence on all tenants on the system.
    (ex. Yes, No)
    (Option buttons)

Supported UADs:

  • Snom 190(Firmware >= 3.60s), 320/360(Firmware >= 4.1)
  • Polycom IP30x/IP50x/IP600
  • Xten EyeBeam
  • Grandstream GXP2000 (Firmware >= 1.0.1.13)
  • Aastra 480i
  • Aastra 9133i

CLI

        show hints
        -= Registered Asterisk Dial Plan Hints =-
        1009                : SIP/1009             State:Idle              Watchers  0
        2001                : SIP/2001             State:Idle              Watchers  0
        1020                : SIP/1020             State:InUse             Watchers  0
        1016                : SIP/1016             State:Unavailable       Watchers  0
        1008                : SIP/1008             State:Idle              Watchers  0
        1006                : SIP/1006             State:Unavailable       Watchers  0
        1000                : SIP/1000             State:Ringing           Watchers  0
        1003                : SIP/1003             State:Unavailable       Watchers  0
        1030                : SIP/1030             State:Unavailable       Watchers  0
        1234                : IAX2/1234            State:Unavailable       Watchers  0
        7777                : IAX2/7777            State:Idle              Watchers  0
        1017                : IAX2/1017            State:Unavailable       Watchers  0
        ----------------
        - 12 hints registered
    

User Agent Auto Provisioning Template



This option allows adding of additional settings to auto-provisioning template. Auto-provisioning settings are generally defined in the 'Settings: UAD' and are custom set for each device.


NOTE: Unless absolutely sure, do not change or add to this template.

Additional Config



This option is used for providing additional config parameters for SIP and IAX configuration files. Values provided here will be written into these configuration files.


NOTE: Unless absolutely sure, do not change or add to this template.

Ring Groups


Ring Groups are used to group a number of UADs/Phones into one network destination. Each Ring Group is assigned a network number which, once dialed, rings all extensions assigned to the group.

  • Ring Group:

    Ring group extension number
    (ex. Accounts)
    (Display)

  • Extension:

    Ring group extension number
    (ex. Once a user dials this number, all destinations assigned to the ring group will ring (e.g. 1111))
    (Display)

  • Destinations:

    Extension Numbers assigned to a ring group
    (ex. Once a ring group number is dialed, all destinations set here will ring at the same time (e.g. 1001, 1002, 1003...))
    (Display)

  • Last Destination:

    Last destination to be called if none of the destination extensions answer the call
    (ex. 1010)
    (Display)

  • Edit

    Edit the ring group configuration
    (ex. Click to edit the ring group configuration)
    (Button)

  • Delete

    Delete a ring group from the system
    Click to delete a ring group from the system
    (Button)

Add/Edit Ring Group



Clicking on the 'Add Ring Group' or 'Edit' button will open the following ring group options:

  • Name:

    Unique Ring group name
    (ex.Set 'Accounts' here to create the same ring group)
    ([a-z][0-9])

  • Extension:

    Unique network number associated with the Ring group
    (ex. When this number is dialed, all extensions associated with it will ring at the same time)
    ([0-9])

  • Extensions:

    System extensions associated with the ring group
    Example:
    Provide an extension list separated by commas here (e.g. 1001,1002,1003...). When a ring group 'Extension' number is dialed, all extensions set here will ring at the same time.
    NOTE: If all destinations fail after 'timeout', 'Last Destination' will be called.
    ([0-9])

  • Incoming Limit (per call):

    If you have a scenario where call is sent from the current ring group to the second one and the second one returns the same call back to first group, it will allow only this much loops.
    Example:
    If this is set to 1 as it is by default, and the current ring group sends the call to the next group (or any other object on the system), returning the same call from that object will not be permitted as same call can enter this group only once.
    NOTICE: system wide limitation for these 'loops' is 10.

Advanced Options


These options fine-tune ring group settings with additional options

  • Greeting:

    Greeting sound file played to callers when the Ring group is dialed
    Example:
    Select 'greeting-default-attendant', for example. Any user that calls this ring group will hear this sound file played to them before all ring group extensions are dialed
    (Select box)

  • Answer on undefined greeting:

    If this option is turned on, the ring group will not answer until the proper greeting is selected.
    (ex. Yes, No, N/A)
    (Option button)

  • Timeout Message

    Sound file played to caller if his call does not get answered by any of the ring group extensions.

    NOTE: Sound file must have 'announce-' name prefix (e.g. 'announce-unavailable')
    (ex. John dials ring group 1000, but nobody answered his call. The sound file selected here will be played to John and then his call will be transferred to 'Last Destination' extension)
    (Select box)

  • Loops:

    How many times to dial all extensions again if nobody answers
    (ex. John dials Ring group 1000, but nobody answers his call. If this option is set to '2', all extensions will be dialed one more time before transferring his call to 'Last Destination')
    ([0-9])

  • Timeout:

    How many seconds will all ring group extensions ring before the call is considered unanswered
    (ex. This option is set to 20. John dials ring group 1000. All Extensions will ring for 20 seconds before timeout occurs. Depending on whether 'Loop' option is set, all extension will be rung again, or John will be transferred to 'Last Destination')
    ([0-9])

  • Dial Options:

    Additional call options assigned to a ring group
    (ex. To play music to ring group callers, set this field to 'm($CLASS)', where m = MOH class e.g. m('default'). Please check details on the bottom)
    ([a-z])

  • Ring Strategy:

    This option regulates how extension in the Ring Group will be ringed.
    Example:
    • All - ring all extensions in the group
    • Leastrecent - ring extension with least answered calls
    • Round - ring each available extension
    • Round Memory - like round, except we remember where we left off the last ring pass
    (Select box)

  • Custom ringtone:

    Set a custom ringtone for the phones which are in this ring group
    (ex. More info can be found in this section: Call Filters & Blocking)
    ([0-9] [a-z])

  • Record Calls:

    Enable call recording service
    (ex. Select 'Yes' to enable the service. All incoming/outgoing calls will be recorded. If using call recording with many extensions, check server disk space from time to time. Please see below for bit rates table).
    (Option buttons)

  • Silent:

    Set whether call recordings should be announced to parties in a conversation.
    (ex. If Silent=No, calling parties will hear a 'Recorded' or 'This call is recorded' message before their conversation starts)
    (Options buttons)

  • Exit Digit:

    Exit digits that transfers the call to the 'Exit Destination'
    (ex. John dials ring group 1000. While all extensions are ringing, John presses the 'Exit digit' set here (e.g. 9) and his call is transferred to the 'Exit Destination').
    ([0-9])

  • Exit Extension:

    EdgelessPBX MT extension to which the call is transferred once the user dials the 'Exit Digit'
    (ex. John dials ring group 1000. While all extensions are ringing, John presses the 'Exit Digit' and his call is transferred to the 'Exit Destination' provided here (e.g. 2001))
    ([0-9])

  • Last Destination:

    Last destination to be dialed if none of the ring group extensions answer the call
    (ex. John dials Ring group 1000, but nobody answers his call. Sound file selected under 'Announce' is played to John and his call is transferred to the extension number set here).
    ([0-9])

  • Last Destination is voicemail:

    Choose whether you want calls to be redirected to the Last Destination or Last Destination voicemail
    (ex. Yes, No, N/A)
    (Option buttons)

  • Confirm Calls:

    Chose whether the called number in the ring group list should be asked to accept or refuse the call from ring group.
    (ex. Yes, No, N/A)
    (Option buttons)

  • Confirmation Message:

    Chose whether to play system default or some custom added sound asking if you want to answer or reject the call
    (ex. All sound files for this option should start with 'rg-announce')
    (Select box)

  • Call Answered Message:

    Chose whether to play system default or custom sound file which is presented to user when he accepts the call from ring group, but the call has already been answered by someone else
    (ex. All sound files for this option should start with 'rg-late-announce')
    (Select box)

Disk Space Used By Call Recording

With continuous tone for 60 seconds

  • wav49 = 84.5kb
  • wav = 833.0kb
  • gsm = 85.0kb

With continuous silent tone (without sound) for 60 seconds

  • wav49 = 84.0kb
  • wav = 827.0kb
  • gsm = 84.0kb

Dial Options:

  • t - Allow the called user to transfer the call by hitting #
  • T - Allow the calling user to transfer the call by hitting #
  • r - Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one answers. Use with care and don't insert this by default into all your dial statements as you are killing call progress information for the user. Really, you almost certainly do not want to use this. Asterisk will generate ring tones automatically where it is appropriate to do so. 'r' makes it go the next step and additionally generate ring tones where it is probably not appropriate to do so.
  • R - Indicate ringing to the calling party when the called party indicates ringing, pass no audio until answered. This is available only if you are using kapejod's bristuff.
  • m - Provide Music on Hold to the calling party until the called channel answers. This is mutually exclusive with option 'r', obviously. Use m(class) to specify a class for the music on hold.
  • o - Restore the Asterisk v1.0 Caller ID behaviour (send the original caller's ID) in Asterisk v1.2 (default: send this extension's number)
  • j - Asterisk 1.2 and later: Jump to priority n+101 if all of the requested channels were busy (just like behaviour in Asterisk 1.0.x)
  • M(x) - Executes the macro (x) upon connect of the call (i.e. when the called party answers)
  • h - Allow the callee to hang up by dialing *
  • H - Allow the caller to hang up by dialing *
  • C - Reset the CDR (Call Detail Record) for this call. This is like using the NoCDR command
  • P(x) - Use the Privacy Manager, using x as the database (x is optional)
  • g - When the called party hangs up, exit to execute more commands in the current context.
  • G(context^exten^pri) - If the call is answered, transfer both parties to the specified priority; however it seems the calling party is transferred to priority x, and the called party to priority x+1
  • A(x) - Play an announcement (x.gsm) to the called party.
  • S(n) - Hang up the call n seconds AFTER the called party picks up.
  • d: - This flag trumps the 'H' flag and intercepts any dtmf while waiting for the call to be answered and returns that value on the spot. This allows you to dial a 1-digit exit extension while waiting for the call to be answered - see also RetryDial
  • D(digits) - After the called party answers, send digits as a DTMF stream, then connect the call to the originating channel.
  • L(x[:y][:z]) - Limit the call to 'x' ms, warning when 'y' ms are left, repeated every 'z' ms) Only 'x' is required, 'y' and 'z' are optional. The following special variables are optional for limit calls: (pasted from app_dial.c)
    • + LIMIT_PLAYAUDIO_CALLER - yes|no (default yes) - Play sounds to the caller.
    • + LIMIT_PLAYAUDIO_CALLEE - yes|no - Play sounds to the callee.
    • + LIMIT_TIMEOUT_FILE - File to play when time is up.
    • + LIMIT_CONNECT_FILE - File to play when the call begins.
    • + LIMIT_WARNING_FILE - File to play as a warning if 'y' is defined. If LIMIT_WARNING_FILE is not defined, then the default behavior is to announce ('You have [XX minutes] YY seconds').
  • f - forces callerid to be set as the extension of the line making/redirecting the outgoing call. For example, some PSTNs don't allow callerids from other extensions than the ones that are assigned to you.
  • w - Allow the called user to start recording after pressing *1 or what defined in features.conf, requires Set(DYNAMIC_FEATURES=automon)
  • W - Allow the calling user to start recording after pressing *1 or what defined in features.conf, requires Set(DYNAMIC_FEATURES=automon)

Departments




Departments section will list all the departments present on this <%PRODUCT%> system, and give the ability to edit or add a new ones. Departments are used by Bicom Systems gloCOM to sort extensions based on the department they belong to.

Add/Edit Department


When you click on Add Department link or the edit button you will be presented with this screen:

  • Name:

    Name of the department
    (ex. Accounting)
    ([0-9][a-z])