[asterisk-users] Catch all undefined numbers to play a nice message
and restart
pedro noticioso
cucnews at yahoo.com
Thu Apr 12 12:02:52 MST 2007
Hi there list!
I want to catch all numbers that don't exist, play a
nice message and restart operator, this is different
from dial i because that is for incorrect extensions,
an undefined number will give a busy signal, something
I don't like
You can search for the word irc to see my comments,
the line above is my latest unsuccessful test, thanks!
; #### #### #### #### #### #### #### #### ####
#### #### ####
;
;
;
;
; begin extensions
;
;
;
;
; #### #### #### #### #### #### #### #### ####
#### #### ####
;
[general] ;
language=es
; autofallthrough=yes
clearglobalvars=no
[globals]
; Definiendo variables para usarlas a traves de todo
el
; MINOMBRE=mailinator.net
; MITELEFONOFXO=55555555
; OPERADORA=
;
; Si static esta en no, u omitido, entonces pbx_config
va a sobreescribir
; a este archivo cuando se cambien las extensiones.
Recuerda que todos los
; comentarios de este archivo desapareceran si pasa
eso.
;
; XXX Todavia no ha sido implementado XXX
;
static=yes
;
;
; si stati=yes y writeprotect=no, tambien puedes
guardar al dialplan con
; linea de comandos ejecutando 'save dialplan' y
borrando estos comentarios
;
writeprotect=yes
CONSOLE=Zap/1 ; pendiente entender *
TRUNK=Zap/1 ; Trunk interface *
TRUNKMSD=1 ; MSD digits to strip (usually
1 or 0) *
; #### #### #### #### #### #### #### #### ####
#### #### ####
; Trunks
;[context] ;exten =>
someexten,priority[+offset][(alias)],application(arg1,arg2,...)
[trunkint] ; International long distance
through trunk
exten => _9001.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunkld] ; Long distance context
accessed through trunk
exten =>
_901ZXXXXXXXXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunklocal] ; Local eight-digit dialing
accessed through trunk interface
exten =>
_9ZXXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) ;
llamada local comun y corriente
exten => _90ZXS0,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
; 020, etc
exten => _9066,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) ;
066, etc
[trunktollfree] ; Long distance context
accessed through trunk interface
exten =>
_901800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunkpaypercall] ; Dangerous pay-per call!
exten =>
_901900.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunkcelular] ; Long distance context
accessed through trunk interface
exten =>
_9044ZZXXXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten =>
_9045ZZXXXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
; #### #### #### #### #### #### #### #### ####
#### #### ####
; Contexts
[international] ; Master context for
international long distance
ignorepat => 9
include => longdistance
include => trunkint
[longdistance] ; Master context for long
distance
ignorepat => 9
include => local
include => trunkld
include => trunktollfree
include => trunkpaypercall
[mercadotecnia]
ignorepat => 9
include => local
[local] ; Master context for local,
toll-free, and iaxtel calls only
ignorepat => 9
include => default
include => parkedcalls
include => trunklocal
[record]
exten => s,1,Answer
exten => s,2,Read(RECORD|enter4digits|4)
exten => s,3,Playback(record-instructions)
exten =>
s,4,Record(/var/lib/asterisk/sounds/recording/s-${RECORD}|wav)
exten => s,5,Wait(2)
exten =>
s,6,Playback(/var/lib/asterisk/sounds/recording/s-${RECORD})
exten => s,7,ResponseTimeout(10)
exten =>
s,8,Background(1toaccept2torerecord3torecordanother)
exten => 1,1,Hangup
exten => 2,1,Goto(s,3)
exten => 3,1,Goto(s,2)
[macro-stdexten];
;
; Macro de extensiones estandard:
; ${ARG1} - Extension (Pudimos haver usado
${MACRO_EXTEN} tambien aqui
; ${ARG2} - Aparato(s) a marcar
;
exten => s,1,Dial(${ARG2},20) ; Ring the interface,
20 seconds maximum
exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on
status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => s-NOANSWER,1,Voicemail(${ARG1},u) ; 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(${ARG1},b) ; If busy,
send to voicemail w/ busy announce
exten => s-BUSY,2,Goto(default,s,1) ; If they press
#, return to start
exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything
else as no answer
exten => a,1,VoicemailMain(${ARG1}) ; If they press
*, send the user into VoicemailMain
[macro-stdexten-viejo] ; Standard extension macro:
; ARG1 es el numero de la extension
; ARG2 es sip al cual voy a marcar
exten => s,1,Dial(${ARG2},20,rt) ; Ring the interface,
20 seconds maximum
exten => s,2,Goto(s-${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
; #### #### #### #### #### #### #### #### ####
#### #### ####
; Dial in
[default]
exten => s,1,Set(CHANNEL(language)=es)
exten => s,2,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,Answer ; Answer the
line
exten => s,n(restart),BackGround(enter-ext-of-person)
; Play the intoduction message
exten => s,n,WaitExten ; Wait for an extension to
be dialed.
; Si no marcan una extension, termina el tiepo de
timeout
; y marca a la operadora
exten => t,1,Macro(stdexten,603,SIP/sip603)
; Si marcan cero en cualquier momento, contesta la
operadora
exten => 0,1,Macro(stdexten,603,SIP/sip603)
exten => i,1,Playback(pbx-invalid) exten =>
i,n,Goto(s,restart,2)
exten => s,1,Set(CHANNEL(language)=es)
exten => 600,1,Macro(stdexten,600,SIP/sip600)
exten => 601,1,Macro(stdexten,601,SIP/sip601)
exten => 602,1,Macro(stdexten,602,SIP/sip602)
exten => 603,1,Macro(stdexten,603,SIP/sip603)
exten => 604,1,Macro(stdexten,604,SIP/sip604)
exten => 605,1,Macro(stdexten,605,SIP/sip605)
exten => 610,1,Macro(stdexten,610,SIP/sip610)
exten => 650,1,Macro(stdexten,650,SIP/sip650)
exten => 651,1,Macro(stdexten,651,SIP/sip651)
exten => 652,1,Macro(stdexten,652,SIP/sip652)
exten => 653,1,Macro(stdexten,653,SIP/sip653)
exten => 8500,1,VoicemailMain
; Para hacer prubas, ahora voy a llamarle al inicio de
las
; llamadas que entran por el FXS, aqui me esta
respondiendo
; Asterisk como si le estuviera llamando por telefono
desde
; otra linea normal
exten => 700,1,Goto(s,1)
exten => operadora,1,Goto(s,1)
exten => 8200,1,Goto(record,s,1)
exten => 8010,1,MusicOnHold(default)
;
; Esto va al final para no tapar a las anteriores
;
; reinicia el ciclo cuando se equivocan de extension
je
; el numero pelon
exten => _[1-580],1,Playback(pbx-invalid)exten =>
_1,n,Goto(s,restart,2)
;
;
; irc all this crap works on some extensions
;
;
;
;
;
;exten => _1,1,Playback(pbx-invalid)exten =>
_1,n,Goto(s,restart,2)
;exten => _2,1,Playback(pbx-invalid)exten =>
_2,n,Goto(s,restart,2)
;exten => _3,1,Playback(pbx-invalid)exten =>
_3,n,Goto(s,restart,2)
;exten => _4,1,Playback(pbx-invalid)exten =>
_4,n,Goto(s,restart,2)
;exten => _5,1,Playback(pbx-invalid)exten =>
_5,n,Goto(s,restart,2)
;exten => _7,1,Playback(pbx-invalid)exten =>
_7,n,Goto(s,restart,2)
;exten => _8,1,Playback(pbx-invalid)exten =>
_8,Goto(s,restart,2)
;exten => _0,1,Playback(pbx-invalid)exten =>
_0,n,Goto(s,restart,2)
;iniciando con y cualquier cosa mas
;exten => _1X.,1,Playback(pbx-invalid)exten =>
_1.,n,Goto(s,restart,2)
;exten => _2X.,1,Playback(pbx-invalid)exten =>
_2.,n,Goto(s,restart,2)
;exten => _3X.,1,Playback(pbx-invalid)exten =>
_3.,n,Goto(s,restart,2)
;exten => _4X.,1,Playback(pbx-invalid)exten =>
_4.,n,Goto(s,restart,2)
;exten => _5X.,1,Playback(pbx-invalid)exten =>
_5.,n,Goto(s,restart,2)
;exten => _7X.,1,Playback(pbx-invalid)exten =>
_7.,n,Goto(s,restart,2)
;exten => _8X.,1,Playback(pbx-invalid)exten =>
_8.,n,Goto(s,restart,2)
;exten => _0X.,1,Playback(pbx-invalid)exten =>
_0.,n,Goto(s,restart,2)
; Gracias por llamar, marque su extension 0 para que
le atienda la operadora
; busy zap?
;
; #### #### #### #### #### #### #### #### ####
#### #### ####
;
;
;
;
; begin features
;
;
;
;
; #### #### #### #### #### #### #### #### ####
#### #### ####
;
; Sample Call Features (parking, transfer, etc)
configuration
;
[general]
; parkext => 700 ; What extension to dial to park
; parkpos => 701-720 ; What extensions to park calls
on. These needs to be
; numeric, as Asterisk starts from the start
position
; and increments with one for the next parked
call.
; context => parkedcalls ; Which context parked calls
are in
;parkingtime => 45 ; Number of seconds a call can be
parked for
; (default is 45 seconds)
;courtesytone = beep ; Sound file to play to the
parked caller
; when someone dials a parked call
; or the Touch Monitor is activated/deactivated.
;parkedplay = caller ; Who to play the courtesy tone
to when picking up a parked call
; one of: parked, caller, both (default is
caller)
;adsipark = yes ; if you want ADSI parking
announcements
;findslot => next ; Continue to the 'next' free
parking space.
; Defaults to 'first' available
;parkedmusicclass=default ; This is the MOH class to
use for the parked channel
; as long as the class is not set on the channel
directly
; using Set(CHANNEL(musicclass)=whatever) in the
dialplan
;transferdigittimeout => 3 ; Number of seconds to wait
between digits when transferring a call
; (default is 3 seconds)
;xfersound = beep ; to indicate an attended transfer
is complete
;xferfailsound = beeperr ; to indicate a failed
transfer
;pickupexten = *8 ; Configure the pickup extension.
(default is *8)
;featuredigittimeout = 500 ; Max time (ms) between
digits for
; feature activation (default is 500 ms)
;atxfernoanswertimeout = 15 ; Timeout for answer on
attended transfer default is 15 seconds.
[featuremap]
;blindxfer => #1 ; Blind transfer (default is #)
;disconnect => *0 ; Disconnect (default is *)
;automon => *1 ; One Touch Record a.k.a. Touch
Monitor
;atxfer => *2 ; Attended transfer
;parkcall => #72 ; Park call (one step
parking)
atxfer => #
blindxfer => #
[applicationmap]
; Note that the DYNAMIC_FEATURES channel variable must
be set to use the features
; defined here. The value of DYNAMIC_FEATURES should
be the names of the features
; to allow the channel to use separated by '#'. For
example:
;
;
Set(DYNAMIC_FEATURES=myfeature1#myfeature2#myfeature3)
;
;
; The syntax for declaring a dynamic feature is the
following:
;
;<FeatureName> =>
<DTMF_sequence>,<ActivateOn>[/<ActivatedBy>],<Application>[,<AppArguments>[,MOH_Class]]
;
; FeatureName -> This is the name of the feature
used in when setting the
; DYNAMIC_FEATURES variable to
enable usage of this feature.
; DTMF_sequence -> This is the key sequence used to
activate this feature.
; ActivateOn -> This is the channel of the call
that the application will be executed
; on. Valid values are "self" and
"peer". "self" means run the
; application on the same channel
that activated the feature. "peer"
; means run the application on the
opposite channel from the one that
; has activated the feature.
; ActivatedBy -> This is which channel is allowed
to activate this feature. Valid
; values are "caller", "callee", and
"both". "both" is the default.
; The "caller" is the channel that
executed the Dial application, while
; the "callee" is the channel called
by the Dial application.
; Application -> This is the application to
execute.
; AppArguments -> These are the arguments to be
passed into the application.
; MOH_Class -> This is the music on hold class to
play while the idle
; channel waits for the feature to
complete. If left blank,
; no music will be played.
;
;
; IMPORTANT NOTE: The applicationmap is not intended
to be used for all Asterisk
; applications. When applications are used in
extensions.conf, they are executed
; by the PBX core. In this case, these applications
are executed outside of the
; PBX core, so it does *not* make sense to use any
application which has any
; concept of dialplan flow. Examples of this would
be things like Macro, Goto,
; Background, WaitExten, and many more.
;
; Enabling these features means that the PBX needs to
stay in the media flow and
; media will not be re-directed if DTMF is sent in the
media stream.
;
; Example Usage:
;
;testfeature => #9,peer,Playback,tt-monkeys ;Allow
both the caller and callee to play
;
;tt-monkeys to the opposite channel
;
;pauseMonitor => #1,self/callee,Pausemonitor
;Allow the callee to pause monitoring
; ;on
their channel
;unpauseMonitor => #3,self/callee,UnPauseMonitor
;Allow the callee to unpause monitoring
; ;on
their channel
;
;
;
;
;
; begin modules
;
;
;
;
;
; Asterisk configuration file
;
; Module Loader configuration file
;
[modules]
autoload=yes
;
; Any modules that need to be loaded before the
Asterisk core has been initialized
; (just after the logger has been initialized) can be
loaded using 'preload'. This
; will frequently be needed if you wish to map all
module configuration files into
; Realtime storage, since the Realtime driver will
need to be loaded before the
; modules using those configuration files are
initialized.
;
; An example of loading ODBC support would be:
;preload => res_odbc.so
;preload => res_config_odbc.so
;
; If you want, load the GTK console right away.
; Don't load the KDE console since
; it's not as sophisticated right now.
;
noload => pbx_gtkconsole.so
;load => pbx_gtkconsole.so
noload => pbx_kdeconsole.so
;
; Intercom application is obsoleted by
; chan_oss. Don't load it.
;
noload => app_intercom.so
;
; Explicitly load the chan_modem.so early on to be
sure
; it loads before any of the chan_modem_* 's afte rit
;
load => chan_modem.so
load => res_musiconhold.so
;
; Load either OSS or ALSA, not both
; By default, load OSS only (automatically) and do not
load ALSA
;
noload => chan_alsa.so
noload => chan_oss.so
;
; Module names listed in "global" section will have
symbols globally
; exported to modules loaded after them.
;
[global]
chan_modem.so=yes
;
;
;
;
; begin sip
;
;
;
;
; Contexto general
[general]
port = 5060 ; Puerto en el que empezamos
bindaddr = 0.0.0.0 ; dirección o direcciones ip
0.0.0.0 = todas
context=local ; Contexto default para todos
tos=lowdelay
dtmfmode=rfc2833 ; info ; Tonos DTMF
disallow=all ; Deshabilita todos los codecs
allow=ulaw ; Permite el codec ulaw (g711)
10kb/s
allow=ilbc ; Permite el codec ilbc 3kb/s
allow=gsm ; Permite el codec gsm 3kb/s
allow=g729 ; Permite el codec g729 2.5kb/s
(propietario)
; Hacemos login en FWD (registrando) para recibir
llamadas a nuestro numero y enviarlas
; A la extensión 21
; FWD number 77443 pointing to extension 21
; register => 77443:miclave at fwd.pulver.com/21
; Para poder sacar llamadas por FWD
; FWD account
;[fwd.pulver.com]
; type=peer
; host=fwd.pulver.com
; fromuser=77443
; fromdomain=fwd.pulver.com
; username=77443
; secret=miclave
; dtmfmode=rfc2833
; Extension 600
[sip600]
type=friend
secret=ext600
context=international
callerid="Nombre Apellido" <600>
host=dynamic
reinvite=no
canreinvite=yes
dtmfmode=info
transfer=yes
nat=no
disallow=all
allow=ulaw
allow=alaw
allow=g729
mailbox=600 at default,600
; Extension 601
[sip601]
type=friend
secret=ext601
context=international
callerid="Nombre Apellido" <601>
host=dynamic
reinvite=no
canreinvite=yes
dtmfmode=info
transfer=yes
nat=no
disallow=all
allow=ulaw
allow=alaw
allow=g729
mailbox=601 at default,601
; Extension 602
[sip602]
type=friend
secret=ext602
context=international
callerid="Nombre correo" <602>
host=dynamic
reinvite=no
canreinvite=yes
dtmfmode=info
transfer=yes
nat=no
disallow=all
allow=ulaw
allow=alaw
allow=g729
mailbox=602 at default,602
; Extension 603
[sip603]
type=friend
secret=ext603
context=international
callerid="Nombre correo" <603>
host=dynamic
reinvite=no
canreinvite=yes
dtmfmode=info
transfer=yes
nat=no
disallow=all
allow=ulaw
allow=alaw
allow=g729
mailbox=603 at default,603
; Extension 604
[sip604]
type=friend
secret=ext604
context=international
callerid="Nombre correo" <604>
host=dynamic
reinvite=no
canreinvite=yes
dtmfmode=info
transfer=yes
nat=no
disallow=all
allow=ulaw
allow=alaw
allow=g729
mailbox=604 at default,604
; Extension 605
[sip605]
type=friend
secret=ext605
context=international
callerid="Nombre correo" <605>
host=dynamic
reinvite=no
canreinvite=yes
dtmfmode=info
transfer=yes
nat=no
disallow=all
allow=ulaw
allow=alaw
allow=g729
mailbox=605 at default,605
[sip610]
type=friend
context=international
callerid="Nombre Apellido" <610>
username=sip610
secret=ext610
nat=no
canreinvite=yes
dtmfmode=info
mailbox=610 at default
disallow=all
allow=ulaw
allow=alaw
allow=g729
[sip650]
type=friend
callerid="XLite Apellido remote" <650>
host=dynamic ; This device needs to
register
username=sip650
secret=ext650
nat=yes ; X-Lite is behind a
NAT router
canreinvite=no ; Typically set to NO
if behind NAT
disallow=all
allow=gsm ; GSM consumes far less
bandwidth than ulaw
allow=ulaw
allow=alaw
[sip651]
type=friend
callerid="SoftPhone de Nombre Apellido" <651>
host=dynamic ; This device needs to
register
username=sip651
secret=ext651
nat=yes ; X-Lite is behind a
NAT router
canreinvite=no ; Typically set to NO
if behind NAT
disallow=all
allow=gsm ; GSM consumes far less
bandwidth than ulaw
allow=ulaw
[sip652]
type=friend
callerid="SoftPhone de Nombre correo Cabrera" <652>
host=dynamic ; This device needs to
register
username=sip652
secret=ext652
nat=yes ; X-Lite is behind a
NAT router
canreinvite=no ; Typically set to NO
if behind NAT
disallow=all
allow=gsm ; GSM consumes far less
bandwidth than ulaw
allow=ulaw
allow=alaw
[sip653]
type=friend
callerid="xlite de Nombre correo Cabrera" <653>
host=dynamic ; This device needs to
register
username=sip653
secret=ext653
nat=yes ; X-Lite is behind a
NAT router
canreinvite=no ; Typically set to NO
if behind NAT
disallow=all
allow=gsm ; GSM consumes far less
bandwidth than ulaw
allow=ulaw
allow=alaw
;
;
;
;
; begin voicemail
;
;
;
;
;
; Voicemail Configuration
;
[general]
; Default formats for writing Voicemail
format=wav ;format=g723sf|wav49|wav
serveremail=soporte at mailinator.net ; Who the e-mail
notification should appear to come from
attach=no ; Should the email contain the voicemail
as an attachment
;maxmessage=180 ; Maximum length of a voicemail
message in seconds
;minmessage=3 ; Minimum length of a voicemail
message in seconds
;maxgreet=60 ; Maximum length of greetings in
seconds
skipms=3000 ; How many miliseconds to skip
forward/back when rew/ff in message playback
maxsilence=2 ; How many seconds of silence before we
end the recording
silencethreshold=30 ; Silence threshold (what we
consider silence, the lower, the more sensitive)
maxlogins=3 ; Max number of failed login attempts
languaje=es
; If you need to have an external program, i.e.
/usr/bin/myapp
; called when a voicemail is left, delivered, or your
voicemailbox
; is checked, uncomment this:
;externnotify=/usr/bin/myapp
; If you need to have an external program, i.e.
/usr/bin/myapp
; called when a voicemail password is changed,
; uncomment this:
;externpass=/usr/bin/myapp
;directoryintro=dir-intro ; For the directory, you can
override the intro file if you want
;charset=ISO-8859-1 ; The character set for voicemail
messages can be specified here
;adsifdn=0000000F ; The ADSI feature descriptor
number to download to
;adsisec=9BDBF7AC ; The ADSI security lock code
;adsiver=1 ; The ADSI voicemail application version
number.
;pbxskip=yes ; Skip the "[PBX]:" string from the
message title
;fromstring=The Asterisk PBX ; Change the From: string
;usedirectory=yes ; Permit finding entries for
forward/compose from the directory
;pagerfromstring=The Asterisk PBX ;Change the From:
string for pager messages
; Change the email body and/or subject, variables:
; VM_NAME, VM_DUR, VM_MSGNUM, VM_MAILBOX,
VM_CALLERID, VM_CIDNUM, VM_CIDNAME, VM_DATE
;
; Note: The emailbody config row can be up to 512
characters due to a limitation in
; asterisk config files.
;emailsubject=[PBX]: New message ${VM_MSGNUM} in
mailbox ${VM_MAILBOX}
; The following definition is very close to the
default, but the default shows just
; the CIDNAME, if it is not null, else just the
CIDNUM, or "an unknown caller" if they are both null.
;emailbody=Dear ${VM_NAME}:\n\n\tjust wanted to let
you know you were just left a ${VM_DUR} long message
(number ${VM_MSGNUM})\nin mailbox ${VM_MAILBOX} from
${VM_CALLERID}, on ${VM_DATE}, so you might\nwant to
check it when you get a chance.
Thanks!\n\n\t\t\t\t--Asterisk\n
; You can override the default program to send e-mail
if you wish, too
;mailcmd=/usr/sbin/sendmail -t
; Advanced options example is extension 4069
; NOTE: All options can be expressed globally in the
general section, and overriden in the per-mailbox
; settings, unless listed otherwise.
;
; tz=central ; Timezone from zonemessages
above. Irrelevant if envelope=no.
attach=yes ; Attach the voicemail to the
notification email *NOT* the pager email
;saycid=yes ; Say the caller id information
before the message. If not described,
; or set to no, it will be
in the envelope
; cidinternalcontexts=intern ; Internal Context for
Name Playback instead of extension digits when saying
caller id.
; sayduration=no ; Turn on/off the duration
information before the message. [ON by default]
; saydurationm=2 ; Specify the minimum duration
to say. Default is 2 minutes
; dialout=fromvm ; Context to dial out from
[option 4 from the advanced menu]
; if not listed, dialing
out will not be permitted
sendvoicemail=yes ; Context to Send voicemail
from [option 5 from the advanced menu]
; if not listed, sending
messages from inside voicemail will not be
; permitted
; callback=fromvm ; Context to call back from
; if not listed, calling
the sender back will not be permitted
; review=yes ; Allow sender to
review/rerecord their message before saving it [OFF by
default
operator=yes ; Allow sender to hit 0
before/after/during leaving a voicemail to
; reach an operator [OFF
by default]
; envelope=no ; Turn on/off envelope
playback before message playback. [ON by default]
; This does NOT affect
option 3,3 from the advanced options menu
; delete=yes ; After notification, the
voicemail is deleted from the server. [per-mailbox
only]
; This is intended for use
with users who wish to receive their voicemail ONLY by
email.
; nextaftercmd=yes ; Skips to the next message
after hitting 7 or 9 to delete/save current message.
; [global option only at
this time]
; forcename=yes ; Forces a new user to record
their name. A new user is
; determined by the
password being the same as
; the mailbox number. The
default is "no".
; forcegreetings=no ; This is the same as
forcename, except for recording
; greetings. The default
is "no".
; hidefromdir=yes ; Hide this mailbox from the
directory produced by app_directory
; The default is "no".
[zonemessages]
eastern=America/New_York|'vm-received' Q 'digits/at'
IMp
central=America/Chicago|'vm-received' Q 'digits/at'
IMp
central24=America/Chicago|'vm-received' q 'digits/at'
H 'digits/hundred' M 'hours'
[default]
600 => 600,Nombre
Apellido,correo at mailinator.com,,operator=yes|attach=yes
601 => 601,Nombre
Apellido,correo at mailinator.com,,operator=yes|attach=yes
602 => 602,Nombre
correo,correo at mailinator.net,,operator=yes|attach=yes
603 => 603,Nombre
Apellido,correo at mailinator.com,,operator=yes|attach=yes
604 => 604,cuarto
sip,4 at sip.com,,operator=yes|attach=yes
605 => 605,quinto
sip,5 at sip.com,,operator=yes|attach=yes
610 => 610,Nombre
Apellido,correo at mailinator.com,,operator=yes|attach=yes
650 => 650,Nombre Apellido
remote,correo at mailinator.com,,operator=yes|attach=yes
650 => 650,Nombre correo
remote,correo at mailinator.net,,operator=yes|attach=yes
;4200 => 9855,Mark
Spencer,markster at linux-support.net,mypager at digium.com,attach=no|serveremail=myaddy at digium.com|tz=central
;4300 => 3456,Ben Rigas,ben at american-computer.net
;4310 => -5432,Sales,sales at marko.net
;4069 => 6522,Matt
Brooks,matt at marko.net,,|tz=central|attach=yes|saycid=yes|dialout=fromvm|callback=fromvm|review=yes|operator=yes|envelope=yes|sayduration=yes|saydurationm=1
;4073 => 1099,Bianca
Paige,bianca at biancapaige.com,,delete=1
;4110 => 3443,Rob Flynn,rflynn at blueridge.net
;
;
;
;
; begin zapata
;
;
;
;
[channels]
busydetect=no
busycount=5
relaxdtmf=yes
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
usecallerid=yes
hidecallerid=no
echotraining=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
immediate=no
context=default
signalling=fxs_ks
callerid=asreceived
callprogress=no
musiconhold=yes
channel => 1-2
____________________________________________________________________________________
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