MT 5.1 Extensions

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Revision as of 15:51, 28 November 2018 by Nedim Mehmedovic (Talk | contribs)

Extensions

Extensions are associated with all UADs/Phones registered to the current slave tenant. On Multi Tenant PBXware, 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.

  • Add Extension
Click Add Extension icon to get to the extension creation screen.
  • Search
Click Search icon in order to open search filter that would allow you to filter extensions per Name, E-mail, Extension or MAC address associated with that extension. Please note you must check or uncheck search criteria in order to specify what you would like to search.
  • CSV Download
By clicking CSV Download icon you will be able to download CSV file that contains all the extensions already created on your tenant. This file can be used to import all the extensions, with their basic details, to a new PBXware system.


System

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

System
  • 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)
  • MAC Address
MAC Address of UADs
  • Status
UAD/Phone system status
(ex. Active/Inactive)
(Display)
  • Procotol
Protocol used by the UAD/Phone
(ex. SIP/IAX)
(Display)
  • 5.0.edit.png Edit
Edit UAD/Phone configuration
(Button)
  • 5.0.delete.png Delete
Delete UAD/Phone from the system
(Button)

Search

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

Search form
  • 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)
  • Extension
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.

Add/Edit Extension

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).
Extension
(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 PBXware 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 PBXware 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 PBXware 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

Adding Multi Extensions

There are two ways to add multiple extensions to PBXware:

1. Manually
To create a list, manually fill in the 'Name','Email','Ext','Secret', 'PIN','Department', ['MAC'] fields and click on the Add(+) icon.
2. Creating and uploading '.csv' file
To Upload a '.csv' file:
  • Open a new file in your text editor
NOTE: Always use simple editor such as Notepad++ on windows, or Gedit on Linux, when editing 'csv' files. Do not use Excel as it may remove formatting which will render your file unusable.
  • Add the following lines:
"Name","Email","Ext","Secret","User Password","PIN","MAC","Department"


For example:

"Joanna Cox","joanna@domain.com","121","jvT-6%xa0Tfpa6Em","KF80x2qvGKbH-3","1111","001122AABBCC","Sales"
  • Save the file as 'multiple-extensions.csv'
  • Click on the 'Browse' button
  • Select 'multiple-extensions.csv' from your hard drive
  • Click the 'Upload' button
  • The New Extensions will be created

NOTE: PBXware requires strong password enforcement, which means that the secret must meet certain criteria in order to be accepted, otherwise PBXware will display an error message stating that secret is too weak.


The 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, ! % * _ -
  • Single/Multiple Extension
(Switch between the single and multi extension adding options)
(Options buttons)

Advanced Options

Advanced Options
Copy As New
Copy As New

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

New feature in PBXware 5,1 is "Copy As New", an option to create a new extension by making copies of an existing extension's settings, in order to speed up the process of adding new extensions to the system.


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.

General Options
  • Extension
Extension number. Number you dial on in order to reach local PBXware 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 PBXware 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 -> Tenants -> Edit Tenant -> Locality (section) -> Emergency Services,
(Select box)
  • Music on Hold:
On extension's level, Music on Hold is new feature introduced in version 4.1.
It allows users to choose different Music on Hold class for every extension. Select the MOH (Music On Hold) class name. All sound files belonging to this MOH class will be played to users dialing this extension.
NOTE: When you select MoH class, you need to enter 'm' in Incoming Dial Option field. Entering 'm' will provide Music on Hold to the calling party until the called channel answers.
  • Show In Directory:
Whether the extension should be shown in Remote Directory accessed through the deskphones interface.
(ex. Yes, No, N/A)
NOTE: Device must support this feature.

Authentication

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

Authentication settings
  • Username
Username used by the UAD/Phone for the registration with PBXware 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 PBXware MT
(ex. By default, this field is automatically populated, but can be changed to any value)
([a-z][0-9])
Authentication settings
  • Fields from Trunk are now available on Extension in Authentication Group:
  • Incoming IP addresses
Ip used by the phone for registration with PBXware
(ex. 192.168.x.x)
  • Insecure
security level for incoming connection
(drop down)
If both of the fields are set to the following:
  • Incoming IP addresses are empty
  • Host is dynamic

Insecure option will not apply. If either of two are set, insecure option will apply.

To enable this option go to Settings > Tenants > "Tenant name" Allow IP Address Authentication for Extensions set this to "YES"


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

Strong Password Requirements

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.

  • User Password
Password used for gloCOM registration with the MT PBXware.
  • Show QR Code
Show QR Code button will display QR code that can be scanned with gloCOM GO mobile application. This feature will make mobile app setup and registration process as fast and simple as possible. In order to setup gloCOM GO with QR Code scanning you will have to enter public address of your PBXware in QR Server field in Settings -> Tenants -> Network Info section.

NOTE: Once you are registered to PBXware with your gloCOM GO you will be asked to change automatically generated User Password in order to log in. Once this procedure is completed User Password will not be visible anymore and QR Code button will be hidden.

  • 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 PBXware MT when checking your Voice inbox or other 'Enhanced Services')
(0-9)

IAX Extensions only

IAX Authentication
  • 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 settings
  • 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

Credit/Debit window
  • 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.

Billing Info
  • 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

Permissions settings

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.

Destinations settings


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...).
  • Special Routes - Other PBX networks we are connected to.



Authorized
caption


PIN Required
caption


Not Authorized
caption

.

Enhanced Services

Enhanced Services allows users to fully adjust settings like Caller ID, Call Pickup, Call Filters & Blocking, Call Forwarding etc.

Enhanced Services



For detailed information on Enhanced Services click the link below:

Enhanced Services

Editions & Modules

gloCOM Editions

Here you can set which gloCOM editions the extension can use. The ALL option is set by default, so the extension can use all editions. You can uncheck the ALL option and choose which editions you want to enable per extension. Next to the edition name you can see how many licenses you have and how many extensions are using each license. If the number of extensions is exceeded, a red warning will appear. For example, 14 of 5 means that 14 extensions is using 5 Agent (call center) editions, and since the limit is exceeded, the red warning will appear.

Network Related

Network related settings

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

  • Transport:
Type of transfer protocol that will be used on PBXware.
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
  • WebRTC Enabled:
Navigate to Settings > Protocols > SIP and make sure TLS is set to “Yes”.
Under Extensions > Extension > Edit screen, make sure option WebRTC Enabled is set to Yes.
For Allowed Codecs make sure “Opus” is enabled and set.
Options: Yes, No, N/A
NOTE: In order for WebRTC to function properly, first thing is to make sure SSL certificates are set properly. Go to Setup Wizard > SSL Certification page and make sure to upload a valid certificate or to use automatic Let’s Encrypt service.
  • Encryption:
This option enables or disables encryption in PBXware transport.
Options: Yes, No, N/A.
  • NAT (Network Address Translation)
Set the appropriate Extension - PBXware NAT relation.
If extension 1000 is trying to register with the PBXware 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
  • No - this option tells the Asterisk to never issue a reinvite to the client
  • Yes - send reinvite to the client
  • No NAT only - allow reinvite when local, deny reinvite when NAT
  • Use UPDATE - use UPDATE instead of INVITE
  • No NAT, Update - use UPDATE when local, deny when NAT
(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.
In PBXware 5.1 'Qualify' is set to 8000 by default.
([0-9])
  • Host
Set the way the UAD/Phone registers to PBXware. 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])
  • Max Contacts
Maximum number of contacts per extension. This number is set as default globally.
This can be overwritten by changing max contacts number per Extension.
([0-9])

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.

Caller ID settings
  • 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 PBXware 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 <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 <Simple-2>)
([a-z][0-9])
  • Ringtone for Transferred calls
Ringtone for transfered calls, work same as Ringtone for local calls setting. Depending of your phone manufacturer you can send a string to the phone in order to use different ringtone than the one set on device. Once this string is set in Ringtone for transfered calls field (as well as in your device itself) it will be used for all calls that are transfered to your extension.
  • 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 caller ID will be reset to Anonymous.
(ex. Yes, No, N/A)
(Option buttons)
  • Trust Remote-Party-ID:
Defines whether PBXware allow Remote-Party-ID header
(Option button)
  • Send Remote-Party-ID:
Should 'Remote-Party-ID' be added to uri.
Options:
  • Use Remote-Party-Id - Use the "Remote-Party-ID" header to send the identity of the remote party
  • Use P-Asserted-Identity - Use the "P-Asserted-Identity" header to send the identity of the remote party
(Select Box)
  • Connected Line Updates:
This option is particularly useful as with some providers, if Use PAI is enabled, calls might start dropping short time after update is sent. Setting Connected Line Updates to No will prevent these call drops.
(Option button)
  • RPID with SIP UPDATE:
In certain cases, the only method by which a connected line change may be immediately transmitted is with a SIP UPDATE request. If communicating with another Asterisk server, and you wish to be able transmit such UPDATE messages to it, then you must enable this option. Otherwise, we will have to wait until we can send a reinvite to transmit the information.

Call Properties

These options fine-tune incoming/outgoing call settings.

Call Properties 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)

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.
Groups settings
  • 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.

Trunks settings
  • 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)
Routes table
  • Routes
Set the default extension trunks 'per Destination/Country provider' (Routes: Example). For some countries/providers to be reached through non-primary trunks, or when one of the default trunks needs to be given higher precedence over another, a route may be set for each destination provider
  • NOTE: These routes are not available if the system is set to use Simple Routing mode.
(Select box)
Routes table

Precedence

Settings
  • Default Trunks: All System calls go through the trunks defined here.
  • MiniLCR: Overrides 'Default Trunks' and sets a specific trunk for a destination.
Extensions
  • Trunks: Overrides 'Settings: Default Trunks'
  • Routes: Overrides 'Settings: MiniLCR'

Routes: Example The list of countries that start with the letter 'A' is displayed when you click on the 'A' in the upper navigation. After the countries are listed, click on one of them to see default trunks for their providers. Once the default trunk is selected for a provider, all calls made from that extension to the set provider will be made using the set trunk.

Call Control

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

Call Control
  • 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])
  • Busy level:
Maximum number of concurent calls until the user/peer is considered busy. This option is not intended for blocking calls, but for displaying user/peer status properly, for example in BLF.
  • Apply Busy Level for Incoming Calls:
If Apply Busy Level for Incoming Calls is set,and one receive an incoming call while extension is on a call, incoming call is blocked and redirected to Voicemail or any other option set for extension.(For this to work One needs to set option Busy Level to 1 under group Call Control).
  • 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

IAX Call Control
  • 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 settings
  • 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)
  • MWI extensions (comma separated):
If you want your extension to be notified for multiple Voicemails, you can enter them in this field.
(ex. 2002, 2003, 2004, 2005)
  • 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 {$NUMBER}' message will be heard).
(Option buttons)
  • Allow Review mode:
Allow B to review the voice message before committing it permanently to A's voice box.
Example:
B leaves a message on A's voice box, but instead of hanging up, he presses '#'. Three options are offered to B:
  • Press 1 to accept this recording
  • Press 2 to listen to it
  • Press 3 to re-record your message
(Option buttons)
  • Allow Operator:
Allow B to reach an operator from within the voice box.
Example:
B leaves a message on A's voice box, but instead of hanging up, B presses '#'.
'Press 0 to reach an operator' message played (Once '0' is pressed, user is offered the following options):
  • Press 1 to accept this recording (If selected, 'Your message has been saved. Please hold while I try that extension' is played and operator is dialed)
  • Or continue to hold (If B holds for a moment, 'Message deleted. Please hold while I try that extension' is played and operator is dialed)
(Option buttons)
  • Operator Extension:
Local extension number that acts as an operator.
(ex. If A's voice box has an option 'Allow Operator' set to 'Yes', all users dialing '#0' inside the voice box will reach this operator extension).
([0-9])
  • Play Envelope message:
Announces the Date/Time and the Extension number from which the message was recorded.
(ex. Once the voice box is checked for new messages, if this option is set to 'Yes', 'Received at {$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 PBXware 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 settings

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: Tenants: 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 PBXware 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.

Hot-desking

Hot-desking
  • Automatic Logout (hours)
Sets automatic log-out time in hours for devices set up for hot-desking.

Recording

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

Recording settings

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' PBXware' menu.
  • Call are recorded in audio format set under 'Settings: Tenants: 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 PBXware MT to automatically provision the UAD/Phone. Configuration files are downloaded from PBXware MT's TFTP server

Auto Provisioning settings
Additional MAC Addresses
  • Auto Provisioning:
Enable auto provisioning service for this extension
(ex. Connect the UAD/Phone to PBXware 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])
  • Additional MAC Addresses
Ability to use multiple MAC addresses per one Extension. This provides the ability to auto provision multiple phones attached to the same Extension.
  • 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.

Pressence Enabled
  • Presence Enabled:
Returns the information whether the phone is on call, ringing, or offline (not registered).
(ex. Select 'Yes' to enable presence support, but not every UAD/Phone support this feature)
(Option buttons)
  • Global Presence:
Enables presence like above option but when this option is turned on, it will enable presence for all tenants on the system. Please note, enabling this option will prevent you from monitoring presence status for extensions on same tenant, unless you enter tenant prefix in front of extension number (i.e. 2001001 for extension 1001 on tenant 200.). However, setting Global Presence to Yes, will prevent you to use BLFs to monitor parking lots on your tenant as their numbers do not have tenant prefix.
(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

UAD Auto Provisioning Template Text box

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

Additional Config text box

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

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.

  • Group Name:
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)
  • 5.0.edit.png Edit
Edit the ring group configuration
(ex. Click to edit the ring group configuration)
(Button)
  • 5.0.delete.png Delete
Delete a ring group from the system
Click to delete a ring group from the system
(Button)


Add/Edit Ring Group

Add/Edit Ring Group

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

  • Group 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])
  • Destinations:
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

Advanced 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])
  • Replace Caller ID:
Replaces the caller ID with the custom data provided here. This is used when you want all incoming calls to your Ring Group to have this value displayed as a caller ID information. Along with the custom data, you can use the '%CALLERID%' variable, which displays the calling party phone number.
NOTE: Please make sure you enter this information as it is written down, otherwise it will not work properly. (ex. Providing a 'USDID' here, will display 'USDID' on your phone display, for all calls coming to this Ring Group. Providing 'USDID %CALLERID%', will display 'USDID 55510205' on your phone display, where 55510205 is calling party phone number).
NOTE: If custom data from previous NOTE does not work for you, try with this variable 'USDID<%CALLERIDNUM%>'
  • 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)
  • Play Periodic Beep
To enable this feature, enter time in seconds to define how often periodic sound signal will be played to informed parties that call recording is enabled.
For example, enter 60 to enable this feature and play periodic signal every 60 seconds.
  • 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:
PBXware 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)

Paging Groups

Paging Groups

Paging Groups feature works similar to standard paging, except this feature allows you to organize extensions to multiple paging groups and assign unique number to each of them. As this feature is used with access code *600, paging group number is entered after the access code.

For example, if we assign number 300 to paging group and add 4 extensions to it, once we dial *600300 we will be able to broadcast the message over intercom to all the extensions added to paging group 300.

  • Group Name
Paging group name
(Display)
  • Number
Paging Group number
(ex.300, Once the user dials *600 + this number, all extensions assigned to this paging group will be paged)
(Display)
  • Destinations
Extensions associated with a paging group
(ex. 102,103,105...)
(Display)
  • 5.0.edit.png Edit the paging group configuration
(ex. Click to edit the paging group configuration)
(Button)
  • 5.0.delete.png Delete the paging group from the system
(ex. Click to delete a paging group from the system)
(Button)


Add/Edit Paging Group

Add/Edit Paging Group

Clicking on 'Add Paging Group' or 'Edit' will open the following options:

  • Group Name
Unique Paging group name
(ex. Set 'Test Paging group' here to create the same paging group)
([a-z][0-9])
  • Number
Paging group extension number
([0-9])
  • Destinations
System extensions associated with the paging group
(ex. 101, 102, 103... Selected extensions will be paged)
  • Quiet mode
Does not play beep to paged extensions.


Departments

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

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])


Hot Desking

Hot Desking is a feature that allows your business the practice of not assigning permanent desks in a workplace, so that employees may work at any available desk.

From a managerial perspective, Hot Desking is attractive because it can cut overhead costs significantly. However, the concept won't work in environments where employees are expected to be in the office most of the time. Furthermore, for some employers, the cost benefits are outweighed by the lack of ability to monitor employee activities. For employees, the system also has both drawbacks and benefits. Many workers are reluctant to give up personal space but others are happy to have more flexibility.

Hot Desking, as PBXware feature, is simple as it can be. By dialing proper access code (*555 by default) on any office phone, user will go to an IVR, where it will be asked for extension and pin. Once proper extension / pin combination is entered, phone will be rebooted and auto provisioned with new extension.

If there was any phone already registered with same extension, it will reboot too and auto provisioned with dynamic extension. If extension is in use, phone will reboot once call ends. Phones provisioned with dynamic extension will not be able to dial anything, but Hot Desking IVR. Actually, whatever user dial, it will end up in Hot Desking IVR.

NOTE: To be able to use Hot Desking feature in PBXware 5, it have to be enabled in your PBXware license. For more information on how to get Hot Desking enabled, please contact your account manager.

Hot Desking
  • MAC
MAC address of the device.
  • Device
Hot Desking Device
  • Extension
Extension's number associated with MAC/Device.
  • Enabled
Status of Hot Desking device.

CSV upload

There is an option to upload CSV file instead of manually entering phones. Click Browse button, select the .csv file from your hard drive and press upload button. Please make sure that CSV file contain comma separated MAC address of the phone and device type per each line (MAC,device), like in example below:

001565267db9,yealinkt42p
0004f23fd871,polycomvvx300

In case you are not sure what to enter for your device name, check the Settings -> UAD, edit UAD for your device model and in General (section) check exact name set under Internal UAD name for Auto-Provisioning field.


CSV download

Once you have created list of Hot Desking devices you might get in to position where you would like to export it to a new system, for testing purposes or in case of a migration for example. CSV Download button gives you an option to download a full list of hot desking devices from a tenant, in CSV format, which can later be used to recreate the list on a new system/tenant.


Download CSV Template

As described earlier you can add large number of hot desking devices at once, by creating a CSV file. Download CSV Template button will present you with a file that already contains necessary headers which should help you create CSV file easier.


Add/Edit Hot Desking Device

Add Hot Desking Device
  • MAC address:
MAC address of the UAD.
  • Device:
Select a device for Hot Desking.
NOTE: Hot Desking feature currently works only with these devices:
  • Yealink - all T4x devices)
  • Polycom - VVX devices (firmware v.4.x)
  • Network:
Network refers to whether the UAD/Phone is in the 'Local' or 'Remote' network.


PIN Based Devices

Every extension has its own unique PBD pin. When you are making a call from a device, that is set as a Pin Based device, after you enter the desired extension number it will ask you to enter your PBD pin. This pin will identify the user on the system and then dialling will proceed as if the user was dialling from his own extension. Billing, CDRs and everything else will apply for the user extension and not for the extension of the Pin Based device. To make a call with extension that is connected with a Pin Based device you must enter the PBD pin for that extension.

For this update, you must resave all your extensions so that new configuration can be applied !

Editing extensions PBD
Add PIN Based Device
PIN Based Device


Setup

Editing extensions PBD PIN field is in the Pin Based Devices section on the Extensions Edit page and it is a required field. PBD PIN must be 5 digits long and unique for every extension.


Adding a new pin based device

To add a new PIN Based device you need to navigate through the main menu, going to Extensions -> Pin Based Devices. A page will open where you can add new, edit, view and delete PIN Based devices.

Clicking on the “Add Pin Based Device” button will open a new page. Here you can enter the name for your new PIN Based device, the extension which will be used by the device and if it should be Active or not. If the Active field is set to No or Not Set, PIN Based Dialling will not be applied for that extension, meaning that the PBD PIN will not be requested when making calls and Billing, CDRs etc… will be applied normally.


PBD local extensions

PIN Based dialling is disabled for local extensions. Entering PBD Pin for local calls is not required.


PBD access codes

Using access codes requires proper identification of the user with PBD pin. Most of the access codes are related per user, which means that they can be used only if they are enabled under enhanced services. Dialing access codes will result in requiring to enter the PBD pin so that system can identify the user and do required actions.


Access codes that can be used with PBD:

  • General Voicemail = *124
  • Call Park = 700
  • Call Park Start = 701
  • Call Park End = 720
  • Speed Dial = *130
  • Call Pickup = *8
  • Asterisk Call Pickup = *88




Next -> 5.Trunks