Asterisk features

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.

See also

• https://wiki.asterisk.org/wiki/display/AST/Features - Features