Dialplan conference dialout

A customer is calling for company X. Company X is not responding and forwards the call to company Y. Company Y picks up the phone and asks the customer who he/she is looking for.

Company Y puts the customer on hold and calls Company X (i.e. other employer) and asks if he/she wants to take a call from a customer. If he/she agrees, Company Y merge the customer with Company X and leaves the call.

So:

  • Put the current (incoming)call on hold (customer)
  • Dial an outgoing number (company X)
  • Merge the outgoing call with the incoming call (customer with company X)
  • Leave the call (company Y)

How can I enable this in my Asterisk configuration? Do I need to make a ConferenceBridge? And how can i tell Asterisk to dial out to a number so I can speak to the person first and then join them together? :smile:

Blind and Attended Transfer options are available most of the time on your Soft-phone or IP phone. You can also setup features codes for this tasks, check /etc/asterisk/features.conf

Thank you for your answer. I didn’t had a features.conf but i added it in the config folder. I loaded the features.conf in Asterisk CLI (config reload features.conf) and checked if it was loaded (config list).

But when i want to make the transfer, I get the message:
“No attended transfer feature code found”

features.conf:

;
; Sample Call Features (transfer, monitor/mixmonitor, etc) configuration
;

; Asterisk 12 Note - All parking lot configuration is now done in res_parking.conf

[general]
transferdigittimeout => 3      ; Number of seconds to wait between digits when transferring a call
                                ; (default is 3 seconds)
xfersound = beep               ; to indicate an attended transfer is complete
xferfailsound = beeperr        ; to indicate a failed transfer
pickupexten = *8               ; Configure the pickup extension. (default is *8)
pickupsound = beep             ; to indicate a successful pickup (default: no sound)
pickupfailsound = beeperr      ; to indicate that the pickup failed (default: no sound)
featuredigittimeout = 1000     ; Max time (ms) between digits for
                                ; feature activation  (default is 1000 ms)
recordingfailsound = beeperr   ; indicates that a one-touch monitor or one-touch mixmonitor feature failed
                                ; to be applied to the call. (default: no sound)
atxfernoanswertimeout = 15     ; Timeout for answer on attended transfer default is 15 seconds.
atxferdropcall = no            ; If someone does an attended transfer, then hangs up before the transfer
                                ; target answers, then by default, the system will try to call back the
                                ; person that did the transfer.  If this is set to "yes", the ringing
                                ; transfer target is immediately transferred to the transferee.
atxferloopdelay = 10           ; Number of seconds to sleep between retries (if atxferdropcall = no)
atxfercallbackretries = 2      ; Number of times to attempt to send the call back to the transferer.
                                ; By default, this is 2.
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.
atxferabort = *1               ; cancel the attended transfer
atxfercomplete = *2            ; complete the attended transfer, dropping out of the call
atxferthreeway = *3            ; complete the attended transfer, but stay in the call. This will turn the call into a multi-party bridge
atxferswap = *4                ; swap to the other party. Once an attended transfer has begun, this options may be used multiple times

; Note that the DTMF features listed below only work when two channels have answered and are bridged together.
; They can not be used while the remote party is ringing or in progress. If you require this feature you can use
; chan_local in combination with Answer to accomplish it.

[featuremap]
blindxfer => #1                ; Blind transfer  (default is #) -- Make sure to set the T and/or t option in the Dial() or Queue() app call!
disconnect => *0               ; Disconnect  (default is *) -- Make sure to set the H and/or h option in the Dial() or Queue() app call!
automon => *1                  ; One Touch Record a.k.a. Touch Monitor -- Make sure to set the W and/or w option in the Dial() or Queue() app call!
atxfer => *2                   ; Attended transfer  -- Make sure to set the T and/or t option in the Dial() or Queue()  app call!
parkcall => #72                ; Park call (one step parking)  -- Make sure to set the K and/or k option in the Dial() app call!
automixmon => *3               ; One Touch Record a.k.a. Touch MixMonitor -- Make sure to set the X and/or x option in the Dial() or Queue() app call!

[applicationmap]
; Note that the DYNAMIC_FEATURES channel variable must be set to use the features
; defined here.  The value of DYNAMIC_FEATURES should be the names of the features
; to allow the channel to use separated by '#'.  For example:
;
;    Set(__DYNAMIC_FEATURES=myfeature1#myfeature2#myfeature3)
;
; (Note: The two leading underscores allow these feature settings to be set
;  on the outbound channels, as well.  Otherwise, only the original channel
;  will have access to these features.)
;
; The syntax for declaring a dynamic feature is any of the following:
;
;<FeatureName> => <DTMF_sequence>,<ActivateOn>[/<ActivatedBy>],<Application>[,<AppArguments>[,MOH_Class]]
;<FeatureName> => <DTMF_sequence>,<ActivateOn>[/<ActivatedBy>],<Application>[,"<AppArguments>"[,MOH_Class]]
;<FeatureName> => <DTMF_sequence>,<ActivateOn>[/<ActivatedBy>],<Application>([<AppArguments>])[,MOH_Class]

;
;  FeatureName   -> This is the name of the feature used when setting the
;                   DYNAMIC_FEATURES variable to enable usage of this feature.
;  DTMF_sequence -> This is the key sequence used to activate this feature.
;  ActivateOn    -> This is the channel of the call that the application will be executed
;                   on. Valid values are "self" and "peer". "self" means run the
;                   application on the same channel that activated the feature. "peer"
;                   means run the application on the opposite channel from the one that
;                   has activated the feature.
;  ActivatedBy   -> ActivatedBy is no longer honored.  The feature is activated by which
;                   channel DYNAMIC_FEATURES includes the feature is on.  Use predial
;                   to set different values of DYNAMIC_FEATURES on the channels.
;                   Historic values are: "caller", "callee", and "both".
;  Application   -> This is the application to execute.
;  AppArguments  -> These are the arguments to be passed into the application.  If you need
;                   commas in your arguments, you should use either the second or third
;                   syntax, above.
;  MOH_Class     -> This is the music on hold class to play while the idle
;                   channel waits for the feature to complete. If left blank,
;                   no music will be played.
;

;
; IMPORTANT NOTE: The applicationmap is not intended to be used for all Asterisk
;   applications. When applications are used in extensions.conf, they are executed
;   by the PBX core. In this case, these applications are executed outside of the
;   PBX core, so it does *not* make sense to use any application which has any
;   concept of dialplan flow. Examples of this would be things like Goto,
;   Background, WaitExten, and many more.  The exceptions to this are Gosub and
;   Macro routines which must complete for the call to continue.
;
; Enabling these features means that the PBX needs to stay in the media flow and
; media will not be re-directed if DTMF is sent in the media stream.
;
; Example Usage:
;
;testfeature => #9,peer,Playback,tt-monkeys  ;Allow both the caller and callee to play
;                                            ;tt-monkeys to the opposite channel
;
; Set arbitrary channel variables, based upon CALLERID number (Note that the application
; argument contains commas)
;retrieveinfo => #8,peer,Set(ARRAY(CDR(mark),CDR(name))=${ODBC_FOO(${CALLERID(num)})})
;
;pauseMonitor   => #1,self/callee,Pausemonitor     ;Allow the callee to pause monitoring
;                                                  ;on their channel
;unpauseMonitor => #3,self/callee,UnPauseMonitor   ;Allow the callee to unpause monitoring
;                                                  ;on their channel

; Dynamic Feature Groups:
;   Dynamic feature groups are groupings of features defined in [applicationmap]
;   that can have their own custom key mappings.  To give a channel access to a dynamic
;   feature group, add the group name to the value of the DYNAMIC_FEATURES variable.
;
; example:
; [myGroupName]         ; defines the group named myGroupName
; testfeature => #9     ; associates testfeature with the group and the keycode '#9'.
; pauseMonitor =>       ; associates pauseMonitor with the group and uses the keycode specified
;                       ; in the [applicationmap].

What am I doing wrong? :smile:

Please provide the the contents of the Asterisk log including this message, and the version of Asterisk you are using, as I cannot find this message in the source code of the the version that I looked at.

If you are using SIP phones, you should be able to do this using SIP transfers rather than feature codes.

This is the wrong forum for asking for help. It is intended for discussions.