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 [EMAIL PROTECTED]:5] NoOp("SIP/774-093cfaf0", "No
recording needed") in new stack
-- Executing [EMAIL PROTECTED]:9] Macro("SIP/774-093cfaf0",
"dial||tTr|711") in new stack
-- Executing [EMAIL PROTECTED]:1] GotoIf("SIP/774-093cfaf0", "1?dial") in
new stack
-- Goto (macro-dial,s,3)
-- Executing [EMAIL PROTECTED]: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 [EMAIL PROTECTED]: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 [EMAIL PROTECTED]:1] Goto("SIP/774-093cfaf0",
"from-pstn|s|1") in new stack
-- Goto (from-pstn,s,1)
-- Executing [EMAIL PROTECTED]:1] NoOp("SIP/774-093cfaf0", "No DID or CID
Match") in new stack
-- Executing [EMAIL PROTECTED]:2] Answer("SIP/774-093cfaf0", "") in new
stack
-- Executing [EMAIL PROTECTED]:3] Wait("SIP/774-093cfaf0", "2") in new stack
-- Executing [EMAIL PROTECTED]: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 [EMAIL PROTECTED]:5] SayAlpha("SIP/774-093cfaf0", "") in new
stack
== Auto fallthrough, channel 'SIP/774-093cfaf0' status is 'ANSWER'
--
_________________________________________
Josimar B. S.
_______________________________________________
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http://www.voipmania.com.br
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