[AsteriskBrasil] [Asteriskbrasil-biz] Asterisk com Caixa postal

Gilmar Cabral gilmarlinux em agrovale.com.br
Quarta Abril 30 14:12:47 BRT 2008


Segue abaixo o sip.conf e a linha referente a caixa postal.
; Give voicemail at extension 8500
;
exten => 8500,1,VoicemailMain
exten => 8500,n,Goto(s,6)


E abaixo segue o meu sip.conf completo esta em negrito.
;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your
; inbound and outbound calls in Asterisk.
;
; This configuration file is reloaded
; - With the "extensions reload" command in the CLI
; - With the "reload" command (that reloads everything) in the CLI

;
; The "General" category is for certain variables. 
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified.  Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=no
;
; If autofallthrough is set, then if an extension runs out of
; things to do, it will terminate the call with BUSY, CONGESTION
; or HANGUP depending on Asterisk's best guess (strongly recommended).
;
; If autofallthrough is not set, then if an extension runs out of
; things to do, asterisk will wait for a new extension to be dialed
; (this is the original behavior of Asterisk 1.0 and earlier).
;
autofallthrough=yes
;
; If clearglobalvars is set, global variables will be cleared
; and reparsed on an extensions reload, or Asterisk reload.
;
; If clearglobalvars is not set, then global variables will persist
; through reloads, and even if deleted from the extensions.conf or
; one of its included files, will remain set to the previous value.
;
clearglobalvars=no
;
; If priorityjumping is set to 'yes', then applications that support
; 'jumping' to a different priority based on the result of their operations
; will do so (this is backwards compatible behavior with pre-1.2 releases
; of Asterisk). Individual applications can also be requested to do this
; by passing a 'j' option in their arguments.
;
priorityjumping=no
; (without the ';'). Note that this is different from the "include" command
; that includes contexts within other contexts. The #include command works
; in all asterisk configuration files.
;#include "filename.conf"

; The "Globals" category contains global variables that can be referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental
; variables,
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
CONSOLE=Console/dsp                             ; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest                                   ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/g2                                    ; Trunk interface
;
; Note the 'g2' in the TRUNK variable above. It specifies which group 
(defined
; in zapata.conf) to dial, i.e. group 2, and how to choose a channel to 
use in
; the specified group. The four possible options are:
;
; g: select the lowest-numbered non-busy Zap channel
;    (aka. ascending sequential hunt group).
; G: select the highest-numbered non-busy Zap channel
;    (aka. descending sequential hunt group).
; r: use a round-robin search, starting at the next highest channel than 
last
;    time (aka. ascending rotary hunt group).
; R: use a round-robin search, starting at the next lowest channel than last
;    time (aka. descending rotary hunt group).
;
TRUNKMSD=1                                      ; MSD digits to strip 
(usually 1 or 0)
;TRUNK=IAX2/user:pass em provider

;
; Any category other than "General" and "Globals" represent
; extension contexts, which are collections of extensions. 
;
; Extension names may be numbers, letters, or combinations
; thereof. If an extension name is prefixed by a '_'
; character, it is interpreted as a pattern rather than a
; literal.  In patterns, some characters have special meanings:
;
;   X - any digit from 0-9
;   Z - any digit from 1-9
;   N - any digit from 2-9
;   [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
;   . - wildcard, matches anything remaining (e.g. _9011. matches
;       anything starting with 9011 excluding 9011 itself)
;   ! - wildcard, causes the matching process to complete as soon as
;       it can unambiguously determine that no other matches are possible

; Contexts contain several lines, one for each step of each
; extension, which can take one of two forms as listed below,
; with the first form being preferred.  One may include another
; context in the current one as well, optionally with a
; date and time.  Included contexts are included in the order
; they are listed.
;
;[context]
;exten => someexten,priority[+offset][(alias)],application(arg1,arg2,...)
;exten => someexten,priority[+offset][(alias)],application,arg1|arg2...
;
; Timing list for includes is
;
;   <time range>|<days of week>|<days of month>|<months>
;
;include => daytime|9:00-17:00|mon-fri|*|*
;
; ignorepat can be used to instruct drivers to not cancel dialtone upon
; receipt of a particular pattern.  The most commonly used example is
; of course '9' like this:
;
;ignorepat => 9
;
; so that dialtone remains even after dialing a 9.
;

;
; Sample entries for extensions.conf
;
;
[dundi-e164-canonical]
;
; List canonical entries here
;
;exten => 12564286000,1,Macro(stdexten,6000,IAX2/foo)
;exten => _125642860XX,1,Dial(IAX2/otherbox/${EXTEN:7})

[dundi-e164-customers]
;
; If you are an ITSP or Reseller, list your customers here.
;
;exten => _12564286000,1,Dial(SIP/customer1)
;exten => _12564286001,1,Dial(IAX2/customer2)

[dundi-e164-via-pstn]
;
; If you are freely delivering calls to the PSTN, list them here
;
;exten => _1256428XXXX,1,Dial(Zap/g2/${EXTEN:7}) ; Expose all of 256-428
;exten => _1256325XXXX,1,Dial(Zap/g2/${EXTEN:7}) ; Ditto for 256-325

[dundi-e164-local]
;
; Context to put your dundi IAX2 or SIP user in for
; full access
;
include => dundi-e164-canonical
include => dundi-e164-customers
include => dundi-e164-via-pstn

[dundi-e164-switch]
;
; Just a wrapper for the switch
;
switch => DUNDi/e164

[dundi-e164-lookup]
;
; Locally to lookup, try looking for a local E.164 solution
; then try DUNDi if we don't have one.
;
include => dundi-e164-local
include => dundi-e164-switch
;
; DUNDi can also be implemented as a Macro instead of using
; the Local channel driver.
;
[macro-dundi-e164]
;
; ARG1 is the extension to Dial
;
exten => s,1,Goto(${ARG1},1)
include => dundi-e164-lookup

;
; Here are the entries you need to participate in the IAXTEL
; call routing system.  Most IAXTEL numbers begin with 1-700, but
; there are exceptions.  For more information, and to sign
; up, please go to www.gnophone.com or www.iaxtel.com
;
[iaxtel700]
exten => _91700XXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel)

; The SWITCH statement permits a server to share the dialplan with
; another server. Use with care: Reciprocal switch statements are not
; allowed (e.g. both A -> B and B -> A), and the switched server needs
; to be on-line or else dialing can be severly delayed.
;
[iaxprovider]
;switch => IAX2/user:[key]@myserver/mycontext

[trunkint]
;
; International long distance through trunk
;
exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
exten => _9011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})

[trunkld]
;
; Long distance context accessed through trunk
;
exten => _91NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN:1})
exten => _91NXXNXXXXXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})

[trunklocal]
;
; Local seven-digit dialing accessed through trunk interface
;
exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})

[trunktollfree]
;
; Long distance context accessed through trunk interface
;
exten => _91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})

[international]
;
; Master context for international long distance
;
ignorepat => 9
include => longdistance
include => trunkint

[longdistance]
;
; Master context for long distance
;
ignorepat => 9
include => local
include => trunkld
[local]
;
; Master context for local, toll-free, and iaxtel calls only
;
ignorepat => 9
include => default
include => parkedcalls
include => trunklocal
include => iaxtel700
include => trunktollfree
include => iaxprovider
;
; You can use an alternative switch type as well, to resolve
; extensions that are not known here, for example with remote
; IAX switching you transparently get access to the remote
; Asterisk PBX
;
; switch => IAX2/user:password em bigserver/local
;
; An "lswitch" is like a switch but is literal, in that
; variable substitution is not performed at load time
; but is passed to the switch directly (presumably to
; be substituted in the switch routine itself)
;
; lswitch => Loopback/12${EXTEN}@othercontext
;
; An "eswitch" is like a switch but the evaluation of
; variable substitution is performed at runtime before
; being passed to the switch routine.
;
; eswitch => IAX2/context@${CURSERVER}

[macro-stdexten];
;
; Standard extension macro:
;   ${ARG1} - Extension  (we could have used ${MACRO_EXTEN} here as well
;   ${ARG2} - Device(s) to ring
;
exten => s,1,Dial(${ARG2},20)                                   ; Ring 
the interface, 20 seconds maximum
exten => s,2,Goto(s-${DIALSTATUS},1)                            ; Jump 
based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten => s-NOANSWER,1,Voicemail(u${ARG1})               ; If 
unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(default,s,1)                 ; If they press 
#, return to start

exten => s-BUSY,1,Voicemail(b${ARG1})                   ; If busy, send 
to voicemail w/ busy announce
exten => s-BUSY,2,Goto(default,s,1)                             ; If 
they press #, return to start

exten => _s-.,1,Goto(s-NOANSWER,1)                              ; Treat 
anything else as no answer

exten => a,1,VoicemailMain(${ARG1})                             ; If 
they press *, send the user into VoicemailMain

[macro-stdPrivacyexten];
; Standard extension macro:
;   ${ARG1} - Extension  (we could have used ${MACRO_EXTEN} here as well
;   ${ARG2} - Device(s) to ring
;   ${ARG3} - Optional DONTCALL context name to jump to (assumes the s,1 
extension-priority)
;   ${ARG4} - Optional TORTURE context name to jump to (assumes the s,1 
extension-priority)`
;
exten => s,1,Dial(${ARG2},20|p)                                 ; Ring 
the interface, 20 seconds maximum, call screening option (or use P for 
databased call screening)
exten => s,2,Goto(s-${DIALSTATUS},1)                            ; Jump 
based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten => s-NOANSWER,1,Voicemail(u${ARG1})               ; If 
unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(default,s,1)                 ; If they press 
#, return to start

exten => s-BUSY,1,Voicemail(b${ARG1})                   ; If busy, send 
to voicemail w/ busy announce
exten => s-BUSY,2,Goto(default,s,1)                             ; If 
they press #, return to start

exten => s-DONTCALL,1,Goto(${ARG3},s,1)               ; Callee chose to 
send this call to a polite "Don't call again" script.

exten => s-TORTURE,1,Goto(${ARG4},s,1)                ; Callee chose to 
send this call to a telemarketer torture script.

exten => _s-.,1,Goto(s-NOANSWER,1)                              ; Treat 
anything else as no answer

exten => a,1,VoicemailMain(${ARG1})                             ; If 
they press *, send the user into VoicemailMain

[demo]
;
; We start with what to do when a call first comes in.
;
exten => s,1,Wait,1                     ; Wait a second, just for fun
exten => s,n,Answer                     ; Answer the line
exten => s,n,Set(TIMEOUT(digit)=5)      ; Set Digit Timeout to 5 seconds
exten => s,n,Set(TIMEOUT(response)=10)  ; Set Response Timeout to 10 seconds
exten => s,n(restart),BackGround(demo-congrats) ; Play a congratulatory 
message
exten => s,n(instruct),BackGround(demo-instruct)        ; Play some 
instructions
exten => s,n,WaitExten          ; Wait for an extension to be dialed.

exten => 2,1,BackGround(demo-moreinfo)  ; Give some more information.
exten => 2,n,Goto(s,instruct)

exten => 3,1,Set(LANGUAGE()=fr)         ; Set language to french
exten => 3,n,Goto(s,restart)                    ; Start with the 
congratulations

exten => 1000,1,Goto(default,s,1)
;
; We also create an example user, 1234, who is on the console and has
; voicemail, etc.
exten => 1234,1,Playback(transfer,skip)         ; "Please hold while..."
                                        ; (but skip if channel is not up)
exten => 1234,n,Macro(stdexten,1234,${CONSOLE})

exten => 1235,1,Voicemail(u1234)                ; Right to voicemail

exten => 1236,1,Dial(Console/dsp)               ; Ring forever
exten => 1236,n,Voicemail(u1234)                ; Unless busy

;
; # for when they're done with the demo
;
exten => #,1,Playback(demo-thanks)              ; "Thanks for trying the 
demo"
exten => #,n,Hangup                     ; Hang them up.

;
; A timeout and "invalid extension rule"
;
exten => t,1,Goto(#,1)                  ; If they take too long, give up
exten => i,1,Playback(invalid)          ; "That's not valid, try again"

;
; Create an extension, 500, for dialing the
; Asterisk demo.
;
exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
exten => 500,n,Dial(IAX2/guest em misery.digium.com/s em default)     ; Call 
the Asterisk demo
exten => 500,n,Playback(demo-nogo)      ; Couldn't connect to the demo site
exten => 500,n,Goto(s,6)                ; Return to the start over message.

;
; Create an extension, 600, for evaluating echo latency.
;
exten => 600,1,Playback(demo-echotest)  ; Let them know what's going on
exten => 600,n,Echo                     ; Do the echo test
exten => 600,n,Playback(demo-echodone)  ; Let them know it's over
exten => 600,n,Goto(s,6)                ; Start over

;
; Give voicemail at extension 8500
;
*exten => 8500,1,VoicemailMain
exten => 8500,n,Goto(s,6)*
;
; Here's what a phone entry would look like (IXJ for example)
;
;exten => 1265,1,Dial(Phone/phone0,15)
;exten => 1265,n,Goto(s,5)

; Here's what a phone entry would look like (IXJ for example)
;
;exten => 1265,1,Dial(Phone/phone0,15)
;exten => 1265,n,Goto(s,5)

;[mainmenu]
;
; Example "main menu" context with submenu
;
;exten => s,1,Answer
;exten => s,n,Background(thanks)                ; "Thanks for calling 
press 1 for sales, 2 for support, ..."
;exten => s,n,WaitExten
;exten => 1,1,Goto(submenu,s,1)
;exten => 2,1,Hangup
;include => default
;
;[submenu]
;exten => s,1,Ringing                                   ; Make them 
comfortable with 2 seconds of ringback
;exten => s,n,Wait,2
;exten => s,n,Background(submenuopts)   ; "Thanks for calling the sales 
department.  Press 1 for steve, 2 for..."
;exten => s,n,WaitExten
;exten => 1,1,Goto(default,steve,1)
;exten => 2,1,Goto(default,mark,2)

[default]
;
; By default we include the demo.  In a production system, you
; probably don't want to have the demo there.
;
include => demo

;
; An extension like the one below can be used for FWD, Nikotel, sipgate etc.
; Note that you must have a [sipprovider] section in sip.conf
;
;exten => _41X.,1,Dial(SIP/${EXTEN:2}@sipprovider,,r)

; Real extensions would go here. Generally you want real extensions to be
; 4 or 5 digits long (although there is no such requirement) and start 
with a
; single digit that is fairly large (like 6 or 7) so that you have plenty of
; room to overlap extensions and menu options without conflict.  You can 
alias
; them with names, too, and use global variables

;exten => 6245,hint,SIP/Grandstream1&SIP/Xlite1,Joe Schmoe ; Channel 
hints for presence
;exten => 6245,1,Dial(SIP/Grandstream1,20,rt)   ; permit transfer
;exten => 6245,n(dial),Dial(${HINT},20,rtT)             ; Use hint as listed
;exten => 6245,n,Voicemail(u6245)               ; Voicemail (unavailable)
;exten => 6245,s+1,Hangup                       ; s+1, same as n
;exten => 6245,dial+101,Voicemail(b6245)        ; Voicemail (busy)
;exten => 6361,1,Dial(IAX2/JaneDoe,,rm)         ; ring without time limit
;exten => 6389,1,Dial(MGCP/aaln/1 em 192.168.0.14)
;exten => 6394,1,Dial(Local/6275/n)             ; this will dial ${MARK}
exten => 8501,1,VoiceMailMain()
[agrovale]
include => demo

;Ramais:
exten => 6600,1,Answer()
exten => 6600,2,Dial(SIP/gilmar,25,tT)
exten => 6600,3,Voicemail(u6600)
exten => 6600,s+1,Hangup
exten => 6600,dial_101,Voicemail(b6600)
;exten => 6600,3,MP3Player(/tmp/music/vida_boa.mp3)
;exten => 6600,4,Dial(SIP/voip6622,25)
;SIP - é o tipo da conta / teste - o nome do user / 25, é que vai chamar 
em segundos
;exten => 6600,5,Hangup

exten => 6602,1,Dial(SIP/voip6602,25)
exten => 6602,2,Hangup

exten => 6603,1,Dial(SIP/voip6603,25)
exten => 6603,2,Hangup


; Ramal Caixa postal
exten => 6604,1,Answer()
exten => 6604,2,Dial(SIP/voip6604,25)
exten => 6604,3,Voicemail(u6604)
exten => 6604,s+1,Hangup
exten => 6604,dial_101,Voicemail(b6604)


;exten => 6604,1,Dial(SIP/voip6604,25)
;exten => 6604,2,Hangup

exten => 6605,1,Dial(SIP/voip6605,25)
exten => 6605,2,Hangup

; chamada em espera
[general]

persistentagents=yes

[agents]

autologoff=15
ackcall=no
wrapuptime=5000
musiconhold => default
recordagentcalls=yes
recordformat=gsm














































Rodrigo Massao Umakoshi escreveu:
> Tem como voce copiar e colar seu dialplan pois no que voce colou nao 
> tem nada de " 8500 " ai fica mais facil para entender.
>
>
> [  ]'s
>
> Rodrigo Massao Umakoshi
>
>
> 2008/4/30 Gilmar Cabral <gilmarlinux em agrovale.com.br 
> <mailto:gilmarlinux em agrovale.com.br>>:
>
>     Fiz conforme voce me informou, digitei 8500 (numero da caixa
>     postal), ai
>     ouvi a gravação bem vindo ao correio de voz, e digitei o ramal que e
>     6604, logo apos ouvi a gravação senha, ai digitei a senha 123, e deu a
>     mensagem senha incorreta.
>     Ja troquei a senha e nda de funcionar.
>
>     Desde ja agradeço atençao e ajuda de todos.
>
>
>
>
>     Rodrigo Massao Umakoshi escreveu:
>     > Gilmar experimente digitar o numero da caixa postal quando vc
>     receber
>     > a mensagem "Bem vindo ao correio de voz", e em seguida vc ira
>     receber
>     > a mensagem "SENHA" ai sim voce digita a senha.
>     >
>     >
>     > [   ]'s
>     >
>     > Rodrigo Massao Umakoshi
>     >
>     > 2008/4/30 Gilmar Cabral <gilmarlinux em agrovale.com.br
>     <mailto:gilmarlinux em agrovale.com.br>
>     > <mailto:gilmarlinux em agrovale.com.br
>     <mailto:gilmarlinux em agrovale.com.br>>>:
>     >
>     >     Sidnei o agradeço ja sua atenção.
>     >     O modelo do ata e GKM2000t
>     >
>     >     Irei postar aki os tempos
>     >     Quando eu pego o telefone que esta ligado no ata e digito 8500
>     >     (correio
>     >     de voz) ele gasta 00:03:00 para falar: BEM VINDO AO CORREIO
>     DE VOZ.
>     >     Depois dele falar BEM VINDO AO CORREIO DE VOZ, gasta
>     00:00:08 para
>     >     falar: SENHA, ai digito a senha, e apois 00:00:04 ele fala SENHA
>     >     INCORRETA.
>     >     Depois ele acusar senha incorreta ele fala CORREIO DE VOZ e
>     SENHA, ai
>     >     digito a senha, e acusa senha incorreta, isto o corre 3
>     veses ai ele
>     >     sair da caixa postal.
>     >
>     >     Desde ja agradeço.
>     >
>     >
>     >     Sidnei -Telecom escreveu:
>     >     > ----- Original Message -----
>     >     > From: "Sidnei -Telecom" <sidnei em union.com.br
>     <mailto:sidnei em union.com.br>
>     >     <mailto:sidnei em union.com.br <mailto:sidnei em union.com.br>>>
>     >     > To: <asteriskbrasil-biz em listas.asteriskbrasil.org
>     <mailto:asteriskbrasil-biz em listas.asteriskbrasil.org>
>     >     <mailto:asteriskbrasil-biz em listas.asteriskbrasil.org
>     <mailto:asteriskbrasil-biz em listas.asteriskbrasil.org>>>
>     >     > Sent: Tuesday, April 29, 2008 6:07 PM
>     >     > Subject: Re: [Asteriskbrasil-biz] Asterisk com Caixa postal
>     >     >
>     >     >
>     >     >
>     >     >> Olá Gilmar.
>     >     >> Qual o modelo de ATA vc utiliza da Intelbras????
>     >     >> Quanto tempo depois que vc disca a senha leva para da a
>     >     mensagemde erro???
>     >     >>
>     >     >>
>     >     >>
>     >     >> ----- Original Message -----
>     >     >> From: "Gilmar Cabral" <gilmarlinux em agrovale.com.br
>     <mailto:gilmarlinux em agrovale.com.br>
>     >     <mailto:gilmarlinux em agrovale.com.br
>     <mailto:gilmarlinux em agrovale.com.br>>>
>     >     >> To: <asteriskbrasil-biz em listas.asteriskbrasil.org
>     <mailto:asteriskbrasil-biz em listas.asteriskbrasil.org>
>     >     <mailto:asteriskbrasil-biz em listas.asteriskbrasil.org
>     <mailto:asteriskbrasil-biz em listas.asteriskbrasil.org>>>
>     >     >> Sent: Tuesday, April 29, 2008 5:56 PM
>     >     >> Subject: Re: [Asteriskbrasil-biz] Asterisk com Caixa postal
>     >     >>
>     >     >>
>     >     >> Obrigado pela sua pasciencia em me responder.
>     >     >> Como estou começando agora no mundo asterisk, estou
>     estudando e
>     >     >> encontrando materiais de configuração do asterisk.
>     >     >> Assim acredito que seja alguma configuração no proprio ata da
>     >     intelbras,
>     >     >> pois nele configurei pouca coisa somente o sip do meu
>     servidor
>     >     asterisk
>     >     >> e o meu ramal e senha o restante ta tudo default.
>     >     >> Se souber qual configuração que posso auterar para fazer
>     o asterisk
>     >     >> aceitar senha digitado nele ficarei grato.
>     >     >>
>     >     >>
>     >     >>
>     >     >> Giov Bs escreveu:
>     >     >>
>     >     >>> Tá alguma coisa errada com o DTMF. ATA nao ta configurado
>     >     certo? Olha
>     >     >>> bem ou Teste alterar o DTMF e DTMFMODE no sip.conf.
>     >     >>>
>     >     >>> 2008/4/29, Gilmar Cabral <gilmarlinux em agrovale.com.br
>     <mailto:gilmarlinux em agrovale.com.br>
>     >     <mailto:gilmarlinux em agrovale.com.br
>     <mailto:gilmarlinux em agrovale.com.br>>>:
>     >     >>>
>     >     >>>
>     >     >>>> Olá a todos, hoje fiz mais aluns testes e descobri algo
>     curioso,
>     >     >>>> conforme eu tinha descrito abaixo que se eu tentar
>     acessar a
>     >     caixa
>     >     >>>> postal pelo telefone que esta conectado o ata simplesmente
>     >     ele da senha
>     >     >>>> incorreta, ai então fiz o seguinte teste, configurei o
>     xlite
>     >     com com o
>     >     >>>> ramal que no ata nao que da senha incorreta. e ja no xlite
>     >     funcionou de
>     >     >>>> boa a caixa postal.
>     >     >>>> Algora alguem sabe o que pode ser, o porque a conta
>     >     configurada pelo
>     >     >>>> xlite eu consigo acessar a caixa postal normalmente
>     digitando
>     >     a senha da
>     >     >>>> mesma e pelo ata nao consigo, so da senha incorreta?
>     >     >>>> Desde ja agradeço
>     >     >>>>
>     >     >>>>
>     >     >>>> gilmarlinux em agrovale.com.br
>     <mailto:gilmarlinux em agrovale.com.br>
>     >     <mailto:gilmarlinux em agrovale.com.br
>     <mailto:gilmarlinux em agrovale.com.br>> escreveu:
>     >     >>>>
>     >     >>>>
>     >     >>>>> Ola.
>     >     >>>>>
>     >     >>>>> Aki meu asterisk esta funcionando blz, so que estou com um
>     >     problema
>     >     >>>>> possuo o servico de correio de voz em 2 ramais, tenho um
>     >     ramal que
>     >     >>>>> esta configurado no xlite no computador que recebo e faco
>     >     ligacao
>     >     >>>>> atraves dele, e no computador usando o xlite eu
>     consigo ouvir as
>     >     >>>>> mensagem que estao no correio de voz. porem tenho um outro
>     >     ramal que
>     >     >>>>> fica ligado no ata, que quando eu digito o numero para
>     entrar no
>     >     >>>>> correio de voz ele entra, e chega a pedir senha, ai quando
>     >     eu coloco a
>     >     >>>>> senha simplesmente nao entra, da senha incorreta, ja
>     tentei
>     >     auterar a
>     >     >>>>> senha e nda, tudo qualquer senha ele da senha
>     incorreta, nao
>     >     consigo
>     >     >>>>> acessar a caixa postal deste ramal que esta o ata.
>     >     >>>>> Ja auterei a senha e dei o comando asterisk -r para entrar
>     >     em modo
>     >     >>>>> console e dei um reload, e mesmo assim nao passa a
>     senha, so
>     >     fala
>     >     >>>>> senha incorreta.
>     >     >>>>>
>     >     >>>>> O ramal que funciona e o 6600 que esta o programa xlite no
>     >     computador
>     >     >>>>> que funciona a caixa postal, e o ramal que esta ligado no
>     >     gatware ata
>     >     >>>>> e o ramal 6604 e pede a senha e nao passa.
>     >     >>>>>
>     >     >>>>> Segue abaixo arquivos de configuracao.
>     >     >>>>>
>     >     >>>>> *sip.conf*
>     >     >>>>>
>     >     >>>>> [gilmar]
>     >     >>>>> type=friend
>     >     >>>>> callerid = "Gilmar" <6600>
>     >     >>>>> username=gilmar ;usuario para login
>     >     >>>>> secret=123 ;senha
>     >     >>>>> host=dynamic ;se nao tem ip fixo
>     >     >>>>> nat=yes
>     >     >>>>> canreinvite=yes ;encaminhar chamadas
>     >     >>>>> context=agrovale
>     >     >>>>>
>     >     >>>>> [voip6602]
>     >     >>>>> type=friend
>     >     >>>>> callerid = "voip6602" <6602>
>     >     >>>>> username=voip6602 ;usuario para login
>     >     >>>>> secret=6602 ;senha
>     >     >>>>> host=dynamic ;se nao tem ip fixo
>     >     >>>>> nat=yes
>     >     >>>>> canreinvite=yes ;encaminhar chamadas
>     >     >>>>> context=agrovale
>     >     >>>>>
>     >     >>>>> [voip6603]
>     >     >>>>> type=friend
>     >     >>>>> callerid = "voip6603" <6603>
>     >     >>>>> username=voip6603 ;usuario para login
>     >     >>>>> secret=6603 ;senha
>     >     >>>>> host=dynamic ;se nao tem ip fixo
>     >     >>>>> nat=yes
>     >     >>>>> canreinvite=yes ;encaminhar chamadas
>     >     >>>>> context=agrovale
>     >     >>>>>
>     >     >>>>> [voip6604]
>     >     >>>>> type=friend
>     >     >>>>> callerid = "voip6604" <6604>
>     >     >>>>> username=voip6604 ;usuario para login
>     >     >>>>> secret=6604 ;senha
>     >     >>>>> host=dynamic ;se nao tem ip fixo
>     >     >>>>> nat=yes
>     >     >>>>> canreinvite=yes ;encaminhar chamadas
>     >     >>>>> context=agrovale
>     >     >>>>>
>     >     >>>>> [voip6605]
>     >     >>>>> type=friend
>     >     >>>>> callerid = "voip6605" <6605>
>     >     >>>>> username=voip6605 ;usuario para login
>     >     >>>>> secret=6605 ;senha
>     >     >>>>> host=dynamic ;se nao tem ip fixo
>     >     >>>>> nat=yes
>     >     >>>>> canreinvite=yes ;encaminhar chamadas
>     >     >>>>> context=agrovale
>     >     >>>>>
>     >     >>>>> *extensions.conf*
>     >     >>>>>
>     >     >>>>> ;Ramais:
>     >     >>>>> exten => 6600,1,Answer()
>     >     >>>>> exten => 6600,2,Dial(SIP/gilmar,25,tT)
>     >     >>>>> exten => 6600,3,Voicemail(u6600)
>     >     >>>>> exten => 6600,s+1,Hangup
>     >     >>>>> exten => 6600,5,dial+101,Voicemail(b6600)
>     >     >>>>> ;exten => 6600,3,MP3Player(/tmp/music/vida_boa.mp3)
>     >     >>>>> ;exten => 6600,4,Dial(SIP/voip6622,25)
>     >     >>>>> ;SIP - é o tipo da conta / teste - o nome do user / 25, é
>     >     que vai
>     >     >>>>> chamar em segundos
>     >     >>>>> ;exten => 6600,5,Hangup
>     >     >>>>>
>     >     >>>>> exten => 6602,1,Dial(SIP/voip6602,25)
>     >     >>>>> exten => 6602,2,Hangup
>     >     >>>>>
>     >     >>>>> exten => 6603,1,Dial(SIP/voip6603,25)
>     >     >>>>> exten => 6603,2,Hangup
>     >     >>>>>
>     >     >>>>>
>     >     >>>>> ; Ramal Caixa postal
>     >     >>>>> exten => 6604,1,Answer()
>     >     >>>>> exten => 6604,2,Dial(SIP/voip6604,25,tT)
>     >     >>>>> exten => 6604,3,Voicemail(u6604)
>     >     >>>>> exten => 6604,s+1,Hangup
>     >     >>>>> exten => 6604,5,dial+101,Voicemail(b6604)
>     >     >>>>>
>     >     >>>>> exten => 6605,1,Dial(SIP/voip6605,25)
>     >     >>>>> exten => 6605,2,Hangup
>     >     >>>>>
>     >     >>>>> ; chamada em espera
>     >     >>>>> [general]
>     >     >>>>>
>     >     >>>>> persistentagents=yes
>     >     >>>>>
>     >     >>>>> [agents]
>     >     >>>>>
>     >     >>>>> autologoff=15
>     >     >>>>> ackcall=no
>     >     >>>>> wrapuptime=5000
>     >     >>>>> musiconhold => default
>     >     >>>>> recordagentcalls=yes
>     >     >>>>> recordformat=gsm
>     >     >>>>>
>     >     >>>>> *voicemail.conf*
>     >     >>>>>
>     >     >>>>> *6600 => 1234,Gilmar
>     >     >>>>> Cabral,gilmar em agrovale.com.br
>     <mailto:gilmar em agrovale.com.br>
>     >     <mailto:gilmar em agrovale.com.br
>     <mailto:gilmar em agrovale.com.br>>,,|tz=central|attach=yes*
>     >     >>>>> 6602 => 6602,voip6602,gilmar em agrovale.com.br
>     <mailto:gilmar em agrovale.com.br>
>     >     <mailto:gilmar em agrovale.com.br
>     <mailto:gilmar em agrovale.com.br>>,,|tz=central|attach=yes
>     >     >>>>> 6603 => 6603,voip6603,gilmar em agrovale.com.br
>     <mailto:gilmar em agrovale.com.br>
>     >     <mailto:gilmar em agrovale.com.br
>     <mailto:gilmar em agrovale.com.br>>,,|tz=central|attach=yes
>     >     >>>>> *6604 => 123,voip6604,ctin em agrovale.com.br
>     <mailto:ctin em agrovale.com.br>
>     >     <mailto:ctin em agrovale.com.br
>     <mailto:ctin em agrovale.com.br>>,,|tz=central|attach=yes*
>     >     >>>>> 6605 => 6605,voip6605,gilmar em agrovale.com.br
>     <mailto:gilmar em agrovale.com.br>
>     >     <mailto:gilmar em agrovale.com.br
>     <mailto:gilmar em agrovale.com.br>>,,|tz=central|attach=yes
>     >     >>>>>
>     >     >>>>>
>     >     >>>>>
>     >     >>>>>
>     >     >>>>>
>     >     >>>>> Sao estes 2 ramais que estao em negrito que possuem
>     correio
>     >     de voz,
>     >     >>>>> porem o primeiro ai que e 6600 consigo entrar na caixa
>     >     postal, porem o
>     >     >>>>> 6604 nao consigo so da senha incorreta mesmo usando eta
>     >     senha ai 123.
>     >     >>>>>
>     >     >>>>> Se alguem puder me ajudar ficarei grato.
>     >     >>>>>
>     >     >>>>>
>     >     >>>>>
>     >     >>>>>
>     >     >>>>>
>     >     >>>
>     >     >>>
>     >     >
>     >
>     >     _______________________________________________
>     >
>



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