Asterisk features

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Overview

Asterisk features 내용 정리.

Basic

Asterisk 는 DTMF 을 통해서 동작하는 기능을 설정할 수 있도록 지원한다(a.k.a feature codes).

크게 다음과 같은 기능들을 지원한다.

  • Feature Code Call Transfers
  • One-touch Features
  • Call Pickup
  • Built-in Dynamic Features
  • Custom Dynamic Features
  • Call Parking

Feature Code Call Transfers

Call transfer 란, 하나의 party에서 다른 party 로 연결을 시도하는 것을 의미한다. Call transfer 에는 크게 다음과 같은 타입이 있다

  • Blind transfer
A blind or unsupervised transfer is where the initiating party is blind to what is happening after the transfer. They are removed from the process as soon as they initiate the transfer. It is a sort of "fire and forget" transfer.
  • Attended transfer
Variations on attended transfer behavior
An attended or supervised transfer happens when one party transfer another party to a new location by first dialing the transfer destination and only completing the transfer when ready. The initiating party is attending or supervising the transfer process by contacting the destination before completing the transfer. This is helpful if the transfer initiator wants to make sure someone answers or is ready at the destination.

Configuring Transfer Features

Transfer feature 를 사용하기 위해서는 다음 세가지 조건들이 충족되어야 한다.

  • The transfer type must be enabled and assigned a DTMF digit string in features.conf or per channel - see Dynamic DTMF Features.
  • The channel must allow the type of transfer attempted. This can be configured via the Application invoking the channel such as Dial or Queue.
  • The channels involved must be answered and bridged.

Enabling blind or attended transfers

In features.conf you must configure the blindxfer or atxfer options in the featuremap section. The options are configureed with the DTMF character string you want to use for accessing the feature. 

[featuremap]
blindxfer = #1
atxfer = *2

Now that you have the feature enabled you need to configure the dialplan such that a particular channel will be allowed to use the feature.

As an example if you want to allow transfers via the Dial application you can use two options, "t" or "T".

  • t : Allow the called party to transfer the calling party by sending the DTMF sequence defined in features.conf. This setting does not perform policy enforcement on transfers initiated by other methods.
  • T : Allow the calling party to transfer the called party by sending the DTMF sequence defined in features.conf. This setting does not perform policy enforcement on transfers initiated by other methods.

Setting these options for Dial in extensions.conf would look similar to the following. 

exten = 102,1,Dial(PJSIP/BOB,30,T)

- The same arguments("t" and "T") work for the Queue and Dial application.

The Asterisk should be restarted or relevant modules should be reloaded for changes to take effect.

Feature codes for attended transfer control

There are a few additional feature codes related to attended transfers. These features allow you to vary the behavior of an attended transfer on command. They are all configured in the 'general' section of features.conf.

  • Aborting an attended transfer
Dialing the atxerabort code aborts an attended transfer. Otherwise there is no way to abort an attended transfer.
  • Completing an attended transfer
Dialing the atxfercomplete code completes an attended transfer and drops out of the call without having to hang up.
  • Completing an attended transfer as a three-way bridge
Dialing the atxferthreeway code completes an attended transfer and enters a bridge with both of the other parties.
  • Swapping between the transferee and destination
Dialing the atxferswap code swaps you between bridges with either party before the transfer is complete. This allows you to talk to either party one at a time before finalizing the attended transfer.

Example 

[general]
atxferabort = *3
atxfercomplete = *4
atxferthreeway = *5
atxferswap = *6

Configuring attended transfer callbacks

By default Asterisk will call back the initiator or the transfer if they hang up before the target answers and the answer timeout is reached. There are a few options for configuring this behavior.

  • No answer timeout
atxfernoanswertimeout allows you to define the timeout for attended transfers. This is the amount of time (in seconds) Asterisk will attempt to ring the target before giving up.
  • Dropped call behavior
atxferdropcall allows you to change the default callback behavior. The default is 'no' which results in Asterisk calling back the initiator of a transfer when they hang up early and the attended transfer times out. If set to 'yes' then the transfer target channel will be immediately transferred to the channel being transferred as soon as the initiator hangs up.
  • Loop delay timing
atxferloopdelay sets the number of seconds to wait between callback retries. This option is only relevant when atxferdropcall=no (or is undefined).
  • Number of retries for callbacks
atxfercallbackretries sets the number of times Asterisk will try to send a failed attended transfer back to the initiator. The default is 2.

Example 

[general]
atxfernoanswertimeout = 15 
atxferdropcall = no 
atxferloopdelay = 10 
atxfercallbackretries = 2

Behavior Options

These options are configured in the "[general]" section of features.conf.

  • General transfer options
;transferdigittimeout = 3 ; Number of seconds to wait between digits when transferring a call
; (default is 3 seconds)
  • Attended transfer options
;xfersound = beep ; to indicate an attended transfer is complete
;xferfailsound = beeperr ; to indicate a failed transfer
;transferdialattempts = 3 ; Number of times that a transferer may attempt to dial an extension before
; being kicked back to the original call.
;transferretrysound = "beep" ; Sound to play when a transferer fails to dial a valid extension.
;transferinvalidsound = "beeperr" ; Sound to play when a transferer fails to dial a valid extension and is out of retries.

Basic Transfer Examples

In the previous section, we configured #1 and *2 as our features codes. We also passed the "T" argument to the Dial application at 102 to allow transfers by the calling party.

Our hypothetical example includes a few devices.

  • PJSIP/ALICE at extension 101
  • PJSIP/BOB at extension 102
  • PJSIP/CATHY at extension 103

Making a blind transfer

For blind transfers we configured the #1 feature code.

Example 

- ALICE dials extension 102 to call BOB
- ALICE decides to transfer BOB to extension 103, so she dials #1. Asterisk will play the audio prompt "transfer".
- ALICE enters the digits 103 for the destination extension.
- Asterisk immediately hangs up the channel between ALICE and BOB. Asterisk creates a new channel for BOB that is dialing extension 103.

Making an attended transfer

For attended transfer we configured *2 as our feature code.

Example 

- ALICE dials extension 102 to call BOB and BOB answers.
- ALICE decides to transfer BOB to extension 103, so she dials *2. Asterisk plays the audio prompt "transfer".
- ALICE enters the digits 103 for the destination extension. Asterisk places BOB on hold and creates a channel for ALICE to dial CATHY.
- CATHY answers : ALICE and CATHY talk. ALICE decides to complete the transfer and hang up the phone.
- Asterisk immediately hangs up the channel between ALICE and BOB. Asterisk plays a short beep tone to CATHY and then bridges the channels for BOB and CATHY.

One-Touch Features

Once configured these features can be activated with only a few or even one keypress on a user's phone. They are often called "one-touch" or "one-step" features.

All of the features are configured via options in the featuremap section of features.conf and require arguments to be passed to the applications invoking the target channel.

Available Features

  • automon : (One-touch Recording) Asterisk will invoke Monitor on the current channel.
  • automixmon : (One-touch Recording) Has the same behavior as automon, but uses MixMonitor instead of Monitor.
  • disconnect : (One-touch Disconnect) When this code is detected on a channel that channel will be immediately hung up.
  • parkcall : (One-touch Parking) Sets a feature code for quickly parking a call.
Most parking options and behavior are configured in res_parking.conf in Asterisk 12 and newer.

Enabling the Features

Configuration of features.conf

the options are configured in features.conf in the featuremap section. They use typical Asterisk configuration file syntax. 

[featuremap]
automon = *1
automixmon = *3
disconnect = *0
parkcall = #72

Assign each option the DTMF character string that you want users to enter for invoking the feature.

Dialplan application options

For each feature there are a pair of options that can be set in the Dial or Queue applications. the two options enable the feature on either the calling party channel. If heither option of a pair are set then you will not be able to use the related feature on the channel.

  • automon
W : Allow the calling party to enable recording of the call.
w : Allow the called party to enable recording of the call.
  • automixmon
X : Allow the calling party to enable recording of the call.
x : Allow the called party to enable recording of the call.
  • disconnect
H : Allow the calling party to hang up the channel.
h : Allow the called party to hang up the channel.
  • parkcall
K : Allow the calling party to enable parking of the call.
k : Allow the called party to enable parking of the call.

Example 

Set the option as you would any application option.

exten = 101,1,Dial(PJSIP/ALICE,30,X)

This would allow the calling party, the party dialing PJSIP/ALICE, to invoke recording on the channel.

Using the Feature

One you have configured features.conf and the option in the application then you only have to enter the feature code on your phone's keypad during a call.

Call Pickup

Call pickup allows you to answer a call while it is ringing another phone or group of phones(other than the phone you are sitting at).

Requesting to pickup a call is done by two basic methods.

by dialplan using the Pickup or PickupChan applications.
by dialing the extension defined for pickupexten configured in features.conf.

Which calls can be picked up is determined by configuration and dialplan. 

Call pickup support added in Asterisk 11

See also

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