[AsteriskBrasil] erro ao transferir chamadas

[Josimar B. S.]

Pessoal... to tentando a um tempão a transferência de chamadas e ainda não
consegui. Um pergunta: Tem como eu puxar uma ligação de um ramal?.
O q tenho é o seguinte:
uso o trixbox trixbox CE current release is 2.6.0.7

Asterisk 1.4.18-3



Edit: features_featuremap_additional.conf
transferdigittimeout => 10
featuredigittimeout = 3000

blindxfer=##
atxfer=*2
automon=*1
disconnect=**

ligo do 774 para o 711

no 711 eu aperto '#' e nao '##' e da o som de transferencia

disco 773 e da que o numero discado nao está em serviço no ramal 774 e no
711 da tom de ocupado

segue o log:


    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
  recordingcheck|20080506-082331|1210073011.26: Inbound recording not
enabled
    -- AGI Script recordingcheck completed, returning 0
    -- Executing [s em macro-record-enable:5] NoOp("SIP/774-093cfaf0", "No
recording needed") in new stack
    -- Executing [s em macro-exten-vm:9] Macro("SIP/774-093cfaf0",
"dial||tTr|711") in new stack
    -- Executing [s em macro-dial:1] GotoIf("SIP/774-093cfaf0", "1?dial") in
new stack
    -- Goto (macro-dial,s,3)
    -- Executing [s em macro-dial:3] AGI("SIP/774-093cfaf0", "dialparties.agi")
in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
  dialparties.agi: Starting New Dialparties.agi
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'admin' logged on from 127.0.0.1
  dialparties.agi: Caller ID name is 'comp josimar' number is '774'
  dialparties.agi: Methodology of ring is  'none'
    --  dialparties.agi: Added extension 711 to extension map
    --  dialparties.agi: Extension 711 cf is disabled
    --  dialparties.agi: Extension 711 do not disturb is disabled
    --  dialparties.agi: dbset CALLTRACE/711 to 774
    --  dialparties.agi: Filtered ARG3: 711
  == Manager 'admin' logged off from 127.0.0.1
    -- AGI Script dialparties.agi completed, returning 0
    -- Executing [s em macro-dial:7] Dial("SIP/774-093cfaf0", "SIP/711||tTr")
in new stack
    -- Called 711
    -- SIP/711-093cc3f8 is ringing
  == Connect attempt from '127.0.0.1' unable to authenticate
    -- SIP/711-093cc3f8 answered SIP/774-093cfaf0
  == Parsing '/etc/asterisk/manager.conf': Found
  == Connect attempt from '127.0.0.1' unable to authenticate
    -- Started music on hold, class 'default', on SIP/774-093cfaf0
    -- <SIP/711-093cc3f8> Playing 'pbx-transfer' (language 'pt_BR')
    -- Stopped music on hold on SIP/774-093cfaf0
  == Channel 'SIP/774-093cfaf0' jumping out of macro 'dial'
  == Channel 'SIP/774-093cfaf0' jumping out of macro 'exten-vm'
    -- Executing [7777 em from-internal-xfer:1] Goto("SIP/774-093cfaf0",
"from-pstn|s|1") in new stack
    -- Goto (from-pstn,s,1)
    -- Executing [s em from-pstn:1] NoOp("SIP/774-093cfaf0", "No DID or CID
Match") in new stack
    -- Executing [s em from-pstn:2] Answer("SIP/774-093cfaf0", "") in new stack
    -- Executing [s em from-pstn:3] Wait("SIP/774-093cfaf0", "2") in new stack
    -- Executing [s em from-pstn:4] Playback("SIP/774-093cfaf0",
"ss-noservice") in new stack
    -- <SIP/774-093cfaf0> Playing 'ss-noservice' (language 'pt_BR')
  == Parsing '/etc/asterisk/manager.conf': Found
  == Connect attempt from '127.0.0.1' unable to authenticate
    -- Executing [s em from-pstn:5] SayAlpha("SIP/774-093cfaf0", "") in new
stack
  == Auto fallthrough, channel 'SIP/774-093cfaf0' status is 'ANSWER'







-- 
_________________________________________
Josimar B. S.
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