Flash Operator Panel Attendant Transfer

[danielem <danielem@xxxxxxxxxxxx>]

Hello everybody! I'm trying to setup attendant transfer with fop (ver.
0.29), but I have a problem:
when I drag the destination phone (224) on mine (223), the person which
I was talking is hanged up (225), and the conversation continue only
with me and the reached person. In my op_server.cfg file I specified
this lines:

...
conference_context=conferences
barge_rooms=900-902
barge_muted=1
...
reverse_transfer=0
...
attendant_hold_extension=s
attendant_hold_context=hold
...

In my extensions.conf I have:

[conferences]
exten => 900,1,Meetme(900|qMAx)
exten => 900,2,Hangup
exten => 900,10,Meetme(900|qMAx)
exten => 900,11,Hangup

exten => 901,1,Meetme(901|qMAx)
exten => 901,2,Hangup
exten => 901,10,Meetme(901|qMAx)
exten => 901,11,Hangup

[hold]
exten => s,1,MusicOnHold

[test]
exten => _2X.,1,Answer
exten => _2X.,n,Dial(SIP/${EXTEN},60,t)
exten => _2X.,n,hangup

In my op_buttons.cfg:

[Sip/223]
Position=92
Label="223"
Extension=223
Context=test
Icon=1

[Sip/224]
Position=n
Label="224"
Extension=224
Context=test
Icon=1

[Sip/225]
Position=n
Label="225"
Extension=225
Context=test
Icon=1

This is the debug output of op_server.pl (debug 2):

127.0.0.1       -> Server:
127.0.0.1       -> Action: Redirect
127.0.0.1       -> Channel: SIP/225-b5e0b6d8
127.0.0.1       -> Exten: s
127.0.0.1       -> Context: hold
127.0.0.1       -> ActionID: 1234attendant
127.0.0.1       -> Priority: 1

127.0.0.1       -> Server:
127.0.0.1       -> Action: Originate
127.0.0.1       -> Channel: SIP/224
127.0.0.1       -> Exten: 900
127.0.0.1       -> Context: conferences
127.0.0.1       -> Priority: 1
127.0.0.1       -> Async: True
127.0.0.1       -> ActionID: attendant-GENERAL

Response: Error
ActionID: 1234attendant
Message: Channel does not exist: SIP/225-b5e0b6d8
Server: 0

And this is the output of my asterisk server:

 -- Executing [223@xxxx:1] Answer("SIP/225-b5e0b6d8", "") in new stack
    -- Executing [223@xxxx:2] Dial("SIP/225-b5e0b6d8", "SIP/223|60|t")
in new stack
    -- Called 223
    -- SIP/223-081797b0 is ringing
    -- SIP/223-081797b0 answered SIP/225-b5e0b6d8
== Spawn extension (test, 223, -1) exited non-zero on 'SIP/225-b5e0b6d8'
  == Auto fallthrough, channel 'SIP/225-b5e0b6d8' status is 'ANSWER'
    -- Executing [900@xxxxxxxxxxx:1] MeetMe("SIP/223-081797b0",
"900|qMAx") in new stack
  == Parsing '/etc/asterisk/meetme.conf': Found
    -- Created MeetMe conference 1023 for conference '900'
    -- Started music on hold, class 'default', on SIP/223-081797b0


Can someone help me, please?