[AsteriskBrasil] Busca do Saber II
Danilo do Vale
danilo_vale em uol.com.br
Quarta Novembro 23 14:48:15 BRT 2005
Pessoal,
Estou querendo aprimorar os recursos da minha configuração atual, mas
estou tendo dificuldade. Abaixo esta os problemas que estou enfrentando
junto com as configurações no asterisk, alem da versão do asterisk e o
linux usado.
Instalei o asterisk versão 1.09 ( Asterisk CVS HEAD built by
root em localhost.localdomain on a i686 running Linux on 2005-10-11
15:24:03 UTC), com linux Fedora 4 atualizado (Linux version
2.6.13-1.1526_FC4 (bhcompile em hs20-bc1-6.build.redhat.com) (gcc version
4.0.1 20050727 (Red Hat 4.0.1-5)) #1 Wed Sep 28 19:15:10 EDT 2005)
O objetivo da configuração feita é para implantar os seguintes recursos:
Primeiro: Identificação de chamadas da rede publica (PSTN). Comprei
*CONVERSOR DE SINALIZAÇÃO DTMF PARA FSK - IDENTECH *para isto.
Resultado: Não esta identificando o numero do originador. Teste
realizado numa linha da Telemar.
-- Starting simple switch on 'Zap/1-1'
Oct 9 12:50:09 NOTICE[7256]: chan_zap.c:6001 ss_thread: Got event 18
(Ring Begin)...
Oct 9 12:50:10 NOTICE[7256]: chan_zap.c:6001 ss_thread: Got event 2
(Ring/Answered)...
Oct 9 12:50:10 ERROR[7256]: callerid.c:274 callerid_feed: fsk_serie
made mylen < 0 (-9)
Oct 9 12:50:10 WARNING[7256]: chan_zap.c:6031 ss_thread: CallerID feed
failed: Success
Oct 9 12:50:10 WARNING[7256]: chan_zap.c:6075 ss_thread: CallerID
returned with error on channel 'Zap/1-1'
-- Executing Goto("Zap/1-1", "recepcao|s|1") in new stack
-- Goto (recepcao,s,1)
-- Executing Dial("Zap/1-1", "SIP/3000|30") in new stack
-- Called 3000
-- SIP/3000-a016 is ringing
Starting simple switch on 'Zap/1-1'
Oct 9 12:51:03 NOTICE[7285]: chan_zap.c:6001 ss_thread: Got event 18
(Ring Begin)...
Oct 9 12:51:04 NOTICE[7285]: chan_zap.c:6001 ss_thread: Got event 2
(Ring/Answered)...
Oct 9 12:51:08 NOTICE[7285]: chan_zap.c:6001 ss_thread: Got event 18
(Ring Begin)...
-- Executing Goto("Zap/1-1", "recepcao|s|1") in new stack
-- Goto (recepcao,s,1)
-- Executing Dial("Zap/1-1", "SIP/3000|30") in new stack
-- Called 3000
-- SIP/3000-7a99 is ringing
Starting simple switch on 'Zap/1-1'
Oct 9 12:52:18 NOTICE[7318]: chan_zap.c:6001 ss_thread: Got event 18
(Ring Begin)...
Oct 9 12:52:19 NOTICE[7318]: chan_zap.c:6001 ss_thread: Got event 2
(Ring/Answered)...
Oct 9 12:52:23 NOTICE[7318]: chan_zap.c:6001 ss_thread: Got event 18
(Ring Begin)...
-- Executing Goto("Zap/1-1", "recepcao|s|1") in new stack
-- Goto (recepcao,s,1)
-- Executing Dial("Zap/1-1", "SIP/3000|30") in new stack
-- Called 3000
-- SIP/3000-388e is ringing
Teste realizado numa linha da Telefonica:
Oct 23 11:36:55 NOTICE[6751]: chan_zap.c:6001 ss_thread: Got event 18
(Ring Begin)...
-- Executing Goto("Zap/1-1", "recepcao|s|1") in new stack
-- Goto (recepcao,s,1)
-- Executing Dial("Zap/1-1", "SIP/3000|30") in new stack
-- Called 3000
-- SIP/3000-80e8 is ringing
Segundo: Utilizar as opções avançadas do voicemail opção 3
* *1* Reply - Não consigo utillizar esta opção em contexto
diferentes, apesar de poder chamar entre os ramais normalmente
apesar de petencer a contexto diferentes Exemplo: ramal 3000
pertence ao contexto ramais e ramal 5000 pertence ao contexto ddd
Resultado:
Executing Dial("SIP/3000-2ab3", "SIP/5000|30|tTmw}") in new stack
-- Called 5000
-- Started music on hold, class 'default', on channel 'SIP/3000-2ab3'
-- SIP/5000-9a3f is ringing
-- SIP/5000-9a3f answered SIP/3000-2ab3
-- Stopped music on hold on SIP/3000-2ab3
-- Attempting native bridge of SIP/3000-2ab3 and SIP/5000-9a3f
Executing Dial("SIP/5000-2080", "SIP/3000|30|tTmw}") in new stack
-- Called 3000
-- Started music on hold, class 'default', on channel 'SIP/5000-2080'
-- SIP/3000-c765 is ringing
-- SIP/3000-c765 answered SIP/5000-2080
-- Stopped music on hold on SIP/5000-2080
-- Attempting native bridge of SIP/5000-2080 and SIP/3000-c765
Parsing '/var/spool/asterisk/voicemail/ddd/5000/INBOX/msg0000.txt': Found
-- No mailbox number '3000' in context 'ddd', no reply sent
* *2* Call back(1) - Não sei como fazer aplicar este recurso
Channel 'SIP/3000-844d' sent into invalid extension '5000' in context
'fromvm', but no invalid handler
* *3* Envelope - Esta funcionando
* *4* Outgoing call(1) - Não sei como fazer aplicar este recurso
Terceiro: Usar o recurso de parking. Disco do softphone (ramal 3000)
para o ata ramal 5000, e feito a ligação perfeitamente. No ramal 5000
aperto flash e disco 700 e escuto 701, coloco o telefone do ramal 5000
no gancho e vou para o telefone do ramal 6000 (ata). Quando disco 701 no
telefone do ramal 6000 para puxar a ligação da tom de ocupado. Noto que
a ligação fica presa no telefone do ramal 5000, pois levando o gancho do
ramal 5000 a ligação entre o ramal 5000 e 3000 permanece. Agora não se
configuração ou a forma de usar o parking ou ambos que esta dando errado
Resultado:
-- Executing Dial("SIP/3000-d00c", "SIP/5000|30|tTmw}") in new stack
-- Called 5000
-- Started music on hold, class 'default', on channel 'SIP/3000-d00c'
-- SIP/5000-b1a7 is ringing
-- SIP/5000-b1a7 answered SIP/3000-d00c
-- Stopped music on hold on SIP/3000-d00c
-- Attempting native bridge of SIP/3000-d00c and SIP/5000-b1a7
-- Started music on hold, class 'default', on channel 'SIP/3000-d00c'
-- Executing Park("SIP/5000-86f2", "") in new stack
== Parked SIP/5000-86f2 on 701. Will timeout back to extension [ddd]
s, 1 in 45 seconds
-- Added extension '701' priority 1 to parkedcalls
-- Playing 'digits/7' (language 'br')
-- Playing 'digits/0' (language 'br')
-- Playing 'digits/1' (language 'br')
-- Started music on hold, class 'default', on channel 'SIP/5000-86f2'
== Spawn extension (ddd, s, 1) exited KEEPALIVE on 'SIP/5000-86f2'
-- Stopped music on hold on SIP/5000-86f2
== SIP/5000-86f2 got tired of being parked
Oct 23 13:33:50 NOTICE[8193]: chan_sip.c:9604 handle_response_peerpoke:
Peer '3000' is now TOO LAGGED! (1242ms / 500ms)
Oct 23 13:34:00 NOTICE[8193]: chan_sip.c:9598 handle_response_peerpoke:
Peer '3000' is now REACHABLE! (10ms / 500ms)
-- Stopped music on hold on SIP/3000-d00c
== Spawn extension (ramais, 5000, 1) exited non-zero on 'SIP/3000-d00c'
Quarto: Como implantar as funções de siga-me , não pertube, chamada a
tres. Quais configurações devo fazer no extensions.conf ou outra
configuração a ser mexida
Resultado: Não sei implantar
### zaptel.conf ###
fxsks=1
loadzone=br
defaultzone=br
### zapata.conf ###
[channels]
language=br
context=entrada
signalling=fxs_ks
musiconhold=default
immediate=yes
callwaitingcallerid=no
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
busydetect=yes
busycount=6
callprogress=no
rxgain=0
txgain=0
callgroup=1
pickupgroup=1
hidecallerid=no
callerid=asreceived
usecallerid=yes
cidsignalling=fsk
cidstart=ring
group=1
channel=1
### extensions.conf ###
[general]
static=yes
writeprotect=no
autofallthrough=no
clearglobalvars=no
priorityjumping=no
[globals]
recepcao=SIP/3000
vendas=SIP/4000
financeiro=SIP/5000
suporte=SIP/6000
linhas=ZAP/g1
[entrada]
include=expediente|7:30-17:30|mon-fri|*|*
include=foradoexpediente|17:31-7:29|mon-fri|*|*
include=foradoexpediente|*|sat-sun|*|*
[expediente]
exten=s,1,Goto(menuprincipal,s,1)
[foradoexpediente]
exten=s,1,Goto(recepcao,s,1)
[menuprincipal]
exten=s,1,Answer
exten=s,2,Set(TIMEOUT(response)=5)
exten=s,3,Background(beep)
exten=1,1,Goto(vendas,s,1)
exten=2,1,Goto(financeiro,s,1)
exten=3,1,Goto(suporte,s,1)
exten=t,1,Goto(recepcao,s,1)
exten=i,1,Goto(recepcao,s,1)
exten=_40XX,1,Dial(SIP/${EXTEN},30,tTmw})
exten=_40XX,2,Voicemail(u${EXTEN})
exten=_40XX,3,Voicemail(b${EXTEN})
exten=_40XX,4,Hangup
exten=_50XX,1,Dial(SIP/${EXTEN},30,tTmw})
exten=_50XX,2,Voicemail(u${EXTEN})
exten=_50XX,3,Voicemail(b${EXTEN})
exten=_50XX,4,Hangup
exten=_60XX,1,Dial(SIP/${EXTEN},30,tTmw})
exten=_60XX,2,Voicemail(u${EXTEN})
exten=_60XX,3,Voicemail(b${EXTEN})
exten=_60XX,4,Hangup
exten=_30XX,1,Dial(SIP/${EXTEN},30,tTmw})
exten=_30XX,2,Voicemail(u${EXTEN})
exten=_30XX,3,Hangup
exten=_30XX,4,Voicemail(b${EXTEN})
exten=_30XX,5,Hangup
exten=9090,1,Goto(caixademensagem,s,1)
exten=9090,2,Hangup
exten=22,1,Wait(2)
exten=22,2,Record(serviconaoautorizado:gsm)
exten=22,3,wait(2)
exten=22,4,Playback(serviconaoautorizado)
exten=22,5,Wait(2)
exten=22,6,Hangup
include=ramais
include=parkedcalls
[caixademensagem]
exten=s,1,VoiceMailMain
exten=s,2,Hangup
[restrito]
include=ramais
[vendas]
exten=s,1,Dial(${vendas},30)
exten=s,2,Voicemail(u${vendas})
exten=s,3,Voicemail(b${vendas})
exten=s,4,Hangup
exten=s,103,Voicemail(u${vendas:4})
exten=s,104,Hangup
exten=s,105,Voicemail(b${vendas:4})
exten=s,106,Hangup
[financeiro]
exten=s,1,Dial(${vendas},30)
exten=s,2,Voicemail(u${financeiro:4})
exten=s,3,Hangup
exten=s,4,Voicemail(b${financeiro:4})
exten=s,5,Hangup
exten=s,103,Voicemail(u${financeiro:4})
exten=s,104,Hangup
exten=s,105,Voicemail(b${financeiro:4})
exten=s,106,Hangup
[suporte]
exten=s,1,Dial(${suporte},30)
exten=s,2,Voicemail(u${suporte:4})
exten=s,3,Hangup
exten=s,4,Voicemail(b${suporte:4})
exten=s,5,Hangup
exten=s,103,Voicemail(u${suporte:4})
exten=s,104,Hangup
exten=s,105,Voicemail(b${suporte:4})
exten=s,106,Hangup
[recepcao]
exten=s,1,Dial(${recepcao},30)
exten=s,2,Voicemail(u${recepcao:4})
exten=s,3,Hangup
exten=s,4,Voicemail(b${recepcao:4})
exten=s,5,Hangup
exten=s,103,Voicemail(u${recepcao:4})
exten=s,104,Hangup
exten=s,105,Voicemail(b${recepcao:4})
exten=s,106,Hangup
[ramais]
exten=0,1,Answer
exten=0,2,Set(TIMEOUT(digit)=5)
exten=0,3,Set(TIMEOUT(response)=5)
exten=0,4,Playtones(dial)
exten=0,5,Read(primeiro,,1)
exten=0,6,StopPlaytones
exten=0,7,Read(check,,3)
exten=0,8,Read(resto,,6)
exten=0,9,GotoIf($["${primeiro}${check}" = "0800"]?10:11})
exten=0,10,Dial(ZAP/g1/${primeiro}${check}${resto},30,m)
exten=0,11,Playback(serviconaoautorizado)
exten=0,12,Hangup
include=menuprincipal
[local]
exten=0,1,Answer
exten=0,2,Set(TIMEOUT(digit)=5)
exten=0,3,Set(TIMEOUT(response)=5)
exten=0,4,Playtones(dial)
exten=0,5,Read(primeiro,,1)
exten=0,6,StopPlaytones
exten=0,7,Read(resto,,7)
exten=0,8,GotoIf($["${primeiro}" = "9"]?11:10})
exten=0,9,GotoIf($["${primeiro}" = "8"]?11:10})
exten=0,10,Dial(ZAP/g1/${primeiro}${resto},30,m)
exten=0,11,Playback(serviconaoautorizado)
exten=0,12,Hangup
include=menuprincipal
include=local
[ddd]
exten=0,1,Answer
exten=0,2,Set(TIMEOUT(digit)=5)
exten=0,3,Set(TIMEOUT(response)=5)
exten=0,4,Playtones(dial)
exten=0,5,Read(primeiro,,1)
exten=0,6,StopPlaytones
exten=0,7,Read(resto,,12)
exten=0,8,Dial(ZAP/g1/${primeiro}${resto},30,m)
include=menuprincipal
include=ddd
[ddi]
exten=0,1,Answer
exten=0,2,Set(TIMEOUT(digit)=5)
exten=0,3,Set(TIMEOUT(response)=5)
exten=0,4,Playtones(dial)
exten=0,5,Read(primeiro,,1)
exten=0,6,StopPlaytones
exten=0,7,Read(resto,,18)
exten=0,8,Dial(Zap/g1/${primeiro}${resto})
include=menuprincipal
include=ddi
### sip.conf ###
[general]
port=5060
bindaddr=0.0.0.0
context=restrito
disallow=all
canreinvite=no
externip=jr.homelinux.com
localnet=192.168.22.3/255.255.255.0
defaultexpirey=120
maxexpirey=400
language=br
[3000]
context=ramais
type=friend
username=3000
secret=3000
host=dynamic
dtmfmode=rfc2833
qualify=500
disallow=all
allow=ilbc
allow=ulaw
nat=yes
canreinvite=no
callgroup=1
pickupgroup=1
callerid=franscisco
[3030]
context=ramais
type=friend
username=3030
secret=3030
host=dynamic
dtmfmode=rfc2833
qualify=500
disallow=all
allow=ilbc
allow=ulaw
nat=yes
canreinvite=no
callgroup=1
pickupgroup=1
[4000]
context=local
type=friend
username=4000
secret=4000
host=dynamic
dtmfmode=rfc2833
qualify=500
disallow=all
allow=ilbc
allow=ulaw
nat=yes
canreinvite=no
callgroup=1
pickupgroup=1
[4040]
context=local
type=friend
username=4040
secret=4040
host=dynamic
dtmfmode=rfc2833
qualify=500
disallow=all
allow=ilbc
allow=ulaw
nat=yes
canreinvite=no
callgroup=1
pickupgroup=1
[5000]
context=ddd
type=friend
username=5000
secret=5000
host=dynamic
dtmfmode=rfc2833
qualify=500
allow=all
allow=ilbc
allow=ulaw
nat=yes
canreinvite=no
callgroup=1
pickupgroup=1
[5050]
context=ddd
type=friend
username=5050
secret=5050
host=dynamic
dtmfmode=rfc2833
qualify=500
disallow=all
allow=ilbc
allow=ulaw
nat=yes
canreinvite=no
callgroup=1
pickupgroup=1
[6000]
context=ddi
type=friend
username=6000
secret=6000
host=dynamic
dtmfmode=rfc2833
qualify=500
disallow=all
allow=ilbc
allow=ulaw
nat=yes
canreinvite=no
callgroup=1
pickupgroup=1
[6060]
context=ddi
type=friend
username=6060
secret=6060
host=dynamic
dtmfmode=rfc2833
qualify=500
disallow=all
allow=ilbc
allow=ulaw
nat=yes
canreinvite=no
callgroup=1
pickupgroup=1
#### voicemail.conf ###
[general]
format=gsm
serveremail=asterisk
attach=yes
maxmsg=30
maxmessage=60
minmessage=5
maxgreet=20
skipms=3000
maxsilence=10
silencethreshold=128
maxlogins=3
callback=fromvm
[zonemessages]
eastern=America/Sao_Paulo|'vm-received' Q 'digits/at' R
[ramais]
3000=12,Usuario1,danilo_vale em uol.com.br,,|tz=eastern
3030=25,Usuario2
[local]
4000=4000,Usuario3
4040=4040,Usuario4
[ddd]
5000 => 5,Usuario5
5050=5050,Usuario6,danilo_vale em uol.com.br,,|tz=eastern
[ddi]
6000 => 45,Usuario7
6060=6060,Usuario8
#### features.conf ####
;
; Sample Parking configuration
;
[general]
parkext => 700 ; What ext. to dial to park
parkpos => 701-720 ; What extensions to park calls on
context => parkedcalls ; Which context parked calls are in
;parkingtime => 45 ; Number of seconds a call can be parked for
; (default is 45 seconds)
;transferdigittimeout => 3 ; Number of seconds to wait between
digits when transfering a call
;courtesytone = beep ; Sound file to play to the parked caller
; when someone dials a parked call
;xfersound = beep ; to indicate an attended transfer is
complete
;xferfailsound = beeperr ; to indicate a failed transfer
;adsipark = yes ; if you want ADSI parking announcements
;findslot => next ; Continue to the 'next' parking space.
Defaults to 'first' available
pickupexten = *8 ; Configure the pickup extension.
Default is *8
;featuredigittimeout = 500 ; Max time (ms) between digits for
; feature activation. Default is 500
[featuremap]
blindxfer => *4 ; Blind transfer
;disconnect => *0 ; Disconnect
;automon => *1 ; One Touch Record
atxfer => *2 ; Attended transfer
[applicationmap]
;testfeature => #9,callee,Playback,tt-monkeys ;Play tt-monkes to
;callee if #9 was pressed
Muito Obrigado
Danilo do Vale
* *
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