First of all i want to apologize if I’m asking the same thing than other person but I cant find a solution on other posts.
I have already working a “background” to wait for users input:
exten => s,n,Background(make-your-selection)
If anyone press 1, 2, 3… application is working so asterisk is recognizing tones.
I have also tested call transfer with zoiper with button transfer.
So, tones are recognized by asterisk and call transfer is working.
When I try to use attended call transfers by pressing keys (*2) nothing happens, well, the other party is listening the two tones (I have try this with zoiper, xlite and analog phone).
How much time should I wait between *2 and the extension I want to call?
Should I be hearing tones?
Should I configure parkedcalls in some way to make get attended call transfers to be working or this is another feature nothing to do with “attended call transfers”?
Thanks for your help. I’m new in asterisk and I’m a noob!!!
My features is:
Builtin Feature Default Current
Pickup *8 *8
Blind Transfer # #1
Attended Transfer *2
One Touch Monitor
Disconnect Call * *0
Park Call *1
[quote=“Mustafa_saa”]if you want to use attended transfer make sure to make the dial application something like this
exten => 2222,1,dial(SIP/2222,15,tTr)[/quote]
I have already this but it is not working…
My Attended Transfer is *2
When I’m on a call I do *2 and then my extension (for example 9021) so y press *29021 and I can hear on both sides 6 tones.
Should I been hearing this?
Should I wait between *2 and extension?
Should I introduce extension and then *2, or *2 extension *2 … ?
to be sure that attended transfer has been initiated, remove the comment in feature.conf from this line:
xfersound = beep ; to indicate an attended transfer is complete
actually if it’s working then when you hit *2 you should hear beep and the other party should be put onhold.
can you plot your extensions.conf here?
on the other hand have you restart your asterisk box after activating the attended transfer feature?
[quote=“Mustafa_saa”]to be sure that attended transfer has been initiated, remove the comment in feature.conf from this line:
xfersound = beep ; to indicate an attended transfer is complete
actually if it’s working then when you hit *2 you should hear beep and the other party should be put onhold.
[/quote]
done, after removing “;” i have done a killall asterisk and then run again asterisk to be sure (can I use just asterisk -r and then reload isn’t it?)
I can not hear anything after hiting *29022 (9022 is an extension)
[general]
allow=all
context=default
maxexpirey=180
defaultexpirey=160
tos=reliability
static=yes ; These two lines prevent the command-line interface
writeprotect=yes ; from overwriting the config file. Leave them here.
I’m not sure what’s going on there. This should work, i’ve copied your configuration into my Server and attended transfer works perfectly.
i’d like to check the DTMF mode for your sip account, but you said that Asterisk is recognizing digits. you might try that anyway.
Also you might plot your feature.conf just for a sanity check
;
; Sample Call Features (parking, transfer, etc) configuration
;
[general]
parkext => 700 ; What extension to dial to park
parkpos => 701-720 ; What extensions to park calls on. These needs to be
; numeric, as Asterisk starts from the start position
; and increments with one for the next parked call.
context => parkedcalls ; Which context parked calls are in
;parkingtime => 45 ; Number of seconds a call can be parked for
; (default is 45 seconds)
;courtesytone = beep ; Sound file to play to the parked caller
; when someone dials a parked call
; or the Touch Monitor is activated/deactivated.
;parkedplay = caller ; Who to play the courtesy tone to when picking up a parked call
; one of: parked, caller, both (default is caller)
;adsipark = yes ; if you want ADSI parking announcements
;findslot => next ; Continue to the ‘next’ free parking space.
; Defaults to ‘first’ available
;parkedmusicclass=default ; This is the MOH class to use for the parked channel
; as long as the class is not set on the channel directly
; using Set(CHANNEL(musicclass)=whatever) in the dialplan
;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)
;featuredigittimeout = 500 ; Max time (ms) between digits for
; feature activation (default is 500 ms)
;atxfernoanswertimeout = 15 ; Timeout for answer on attended transfer default is 15 seconds.
; 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 #)
disconnect => *0 ; Disconnect (default is *)
;automon => *1 ; One Touch Record a.k.a. Touch Monitor
atxfer => *2 ; Attended transfer
parkcall => *1 ; Park call (one step parking)
i’ve created exactly a similar scenario to yours (i’ve used your features.conf file) and everything went OK.
So the only possible purpose why this is not working from my point of view is the dtmf mode for your sip account.
(can I use just asterisk -r and then reload isn’t it?)