[Asterisk-Users] Call Transfer does not work

[jbauer at teec.com]

Hi !

I am trying to transfer calls between internal SIP softclients, but it does
not work. Every time I press a key on the softclient, the CLI shows the
following output:

Attempting native bridge of SIP/456-9ee0 and SIP/173-f586

This is my extensions.conf:

[macro-voicemail]
exten => s,1,Dial(${ARG1},5,Ttr)
exten => s,2,Goto(status-${DIALSTATUS},1)
exten => status-BUSY,1,VoiceMail(b${MACRO_EXTEN})
exten => status-BUSY,2,Playback(vm-goodbye)
exten => status-BUSY,3,Hangup()
exten => status-NOANSWER,1,VoiceMail(u${MACRO_EXTEN})
exten => status-NOANSWER,2,Playback(vm-goodbye)
exten => status-NOANSWER,3,Hangup()

[internal]
exten => _ZXZ,1,Macro(voicemail,SIP/${EXTEN})

And this is the part of the features.conf I changed (just uncommented that
part)

[featuremap]
blindxfer => #1                 ; Blind transfer
disconnect => *0                ; Disconnect
automon => *1                   ; One Touch Record
atxfer => *2                    ; Attended transfer

None of the shortcuts in [featuremap] works.

What am I doing wrong?

Regards, Jens