[asterisk-users] Call does not go voicemail

[thelma at sys-concept.com]

Call is not forwarded to voicemail in below dial plan, why?

exten => 4,1,Dial(${FD_L1},25,trw)
exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?line2)
exten => 4,n(line2),Dial(${FD_L2},20,trw)
exten => 4,n,Voicemail(4)
exten => 4,n,Hangup()

    -- Called SIP/4
    -- SIP/4-00000288 is ringing
    -- Nobody picked up in 25000 ms
    -- Executing [4 at extensions:2] GotoIf("IAX2/home_server-6364", "0?line2") in new stack
    -- Executing [4 at extensions:3] Dial("IAX2/home_server-6364", "SIP/54,20,trw") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/54
    -- SIP/54-00000289 is ringing
  == Spawn extension (extensions, 4, 3) exited non-zero on 'IAX2/home_server-6364'
    -- Hungup 'IAX2/home_server-6364'

Extension 4 is not BUSY (just nobody pickup the call) so why isn't call going to "Voicemail" it shouldn't ring FD_L2 (SIP/54)
Why isn't it going to "Voicemail"?

-- 
Thelma