;
; Asterisk Call Detail Record engine configuration
;
; CDR is Call Detail Record, which provides logging services via a variety of
; pluggable backend modules. Detailed call information can be recorded to
; databases, files, etc. Useful for billing, fraud prevention, compliance with
; Sarbanes-Oxley aka The Enron Act, QOS evaluations, and more.
;
[general]
; Define whether or not to use CDR logging. Setting this to "no" will override
; any loading of backend CDR modules. Default is "yes".
;enable=yes
; Define the CDR batch mode, where instead of posting the CDR at the end of
; every call, the data will be stored in a buffer to help alleviate load on the
; asterisk server. Default is "no".
;
; WARNING WARNING WARNING
; Use of batch mode may result in data loss after unsafe asterisk termination
; ie. software crash, power failure, kill -9, etc.
; WARNING WARNING WARNING
;
;batch=no
; Define the maximum number of CDRs to accumulate in the buffer before posting
; them to the backend engines. 'batch' must be set to 'yes'. Default is 100.
;size=100
; Define the maximum time to accumulate CDRs in the buffer before posting them
; to the backend engines. If this time limit is reached, then it will post the
; records, regardless of the value defined for 'size'. 'batch' must be set to
; 'yes'. Note that time is in seconds. Default is 300 (5 minutes).
;time=300
; The CDR engine uses the internal asterisk scheduler to determine when to post
; records. Posting can either occure inside the scheduler thread, or a new
; thread can be spawned for the submission of every batch. For small batches,
; it might be acceptable to just use the scheduler thread, so set this to "yes".
; For large batches, say anything over size=10, a new thread is recommended, so
; set this to "no". Default is "no".
;scheduleronly=no
; When shutting down asterisk, you can block until the CDRs are submitted. If
; you don't, then data will likely be lost. You can always check the size of
; the CDR batch buffer with the CLI "cdr status" command. To enable blocking on
; submission of CDR data during asterisk shutdown, set this to "yes". Default
; is "yes".
;safeshutdown=yes
; **** #### **** #### **** #### **** #### **** #### extensions.conf de 004
[general]
static=yes
writeprotect=no
clearglobalvars=no
languaje=es
[globals]
CONSOLE=Zap/1
TRUNK=Zap/1 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
;
; 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
;[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(std-exten,6000,IAX2/foo)
;exten => _125642860XX,1,Dial(IAX2/otherbox/${EXTEN:7})
[dundi-e164-via-pstn]
; If you are freely delivering calls to the PSTN, list them here
;exten => _1256428XXXX,1,Dial(Zap/g2/${EXTEN:7}) ; Expose all of 256-428
;exten => _1256325XXXX,1,Dial(Zap/g2/${EXTEN:7}) ; Ditto for 256-325
[dundi-e164-local]
; Context to put your dundi IAX2 or SIP user in for
; full access
include => dundi-e164-canonical
include => dundi-e164-via-pstn
[dundi-e164-switch]
; Just a wrapper for the switch
switch => DUNDi/e164
[dundi-e164-lookup]
; Locally to lookup, try looking for a local E.164 solution
; then try DUNDi if we don't have one.
include => dundi-e164-local
include => dundi-e164-switch
; DUNDi can also be implemented as a Macro instead of using
; the Local channel driver.
[macro-dundi-e164]
; ARG1 is the extension to Dial
exten => s,1,Goto(${ARG1},1)
include => dundi-e164-lookup
[iaxtel700]
ignorepat => 9
exten => _91700XXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel)
[trunkint] ; International long distance through trunk
ignorepat => 9
exten => _9001.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunkld] ; Long distance context accessed through trunk
ignorepat => 9
exten => _901ZXXXXXXXXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunklocal] ; Local eight-digit dialing accessed through trunk interface
ignorepat => 9
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:${TRUNKMSD}}) ; 066, etc
[trunktollfree] ; Long distance context accessed through trunk interface
ignorepat => 9
exten => _901800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunkpaypercall] ; Dangerous pay-per call!
ignorepat => 9
exten => _901900.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunkcelular] ; Long distance context accessed through trunk interface
ignorepat => 9
exten => _9044ZZXXXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9045ZZXXXXXXXX,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
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
; **** #### **** #### **** #### **** #### **** #### extensions.conf de 004
[macro-stdexten];
;
; Macro de extensiones estandard:
; ${ARG1} - Extension (Pudimos haver usado ${MACRO_EXTEN} tambien aqui
; ${ARG2} - Aparato(s) a marcar
;
exten => s,n,Set(CHANNEL(language)=es)
exten => s,1,Dial(${ARG2},20,rt) ; Ring the interface, 20 seconds maximum
exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => s-NOANSWER,1,Voicemail(${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
[testgoodbye] ; ext 701
exten => s,1,Answer()
exten => s,n,Wait,1
exten => s,2,Playback(vm-goodbye)
exten => s,n,Wait,1
;exten => s,3,Hangup()
exten => s,1,Goto(s,restart)
[testmovies1] ; ext 702
exten => s,1,Answer()
exten => s,2,Background(vm-options)
;exten => s,3,Hangup()
exten => s,1,Goto(s,restart)
exten => 1,1,Playback(phonetic/a_p)
exten => 1,2,Goto(incoming,s,1)
exten => 2,1,Playback(phonetic/z_p)
exten => 2,2,Goto(incoming,s,1)
[testmovies2]
exten => s,1,Answer() ; Answer the line
exten => s,2,Background(current-movies) ; Play back the "current movies" sound file
exten => s,3,Hangup() ; Now hangup the line if the caller doesn't press a key
exten => 1,1,Playback(movie1) ; Now hangup the line if the caller doesn't press a key
exten => 1,2,Goto(incoming,s,1) ; Now go back to the beginning
exten => 2,1,Playback(movie2) ; Now hangup the line if the caller doesn't press a key
exten => 2,2,Goto(incoming,s,1) ; Now go back to the beginning
exten => 0,1,Dial(Zap/1)
[default]
;*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*#*# AQUI ES DONDE ESTOY TRABAJANDO
exten => s,1,Wait,1 ; sirve para asegurarnos que los sip agarren bien
exten => s,n,Answer ; Contesta la llamada
exten => s,n,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten => s,n,ResponseTimeout,10 ; Set Response Timeout to 10 seconds
exten => s,n(restart),Set(CHANNEL(language)=es)
exten => s,n,BackGround(enter-ext-of-person) ; Play the intoduction message
exten => s,n,WaitExten ; Wait for an extension to be dialed.
exten => t,1,Dial(SIP/sip602,10,rt)
; reinicia si hay numeros incorrectos
exten => i,1,Goto(s,restart)
; Marca CERO si quieres OPERADORA, en este caso es un telefono fisico, o no sirve porque se va al inicio del dp
exten => 0,1,Macro(stdexten,503,SIP/sip503)
; SOPORTE
exten => 555,1,Macro(stdexten,501,SIP/sip501)
exten => s,n,Set(CHANNEL(language)=es)
exten => 500,1,Macro(stdexten,500,SIP/sip500)
exten => 501,1,Macro(stdexten,501,SIP/sip501)
exten => 502,1,Macro(stdexten,502,SIP/sip502)
exten => 503,1,Macro(stdexten,503,SIP/sip503)
exten => 504,1,Macro(stdexten,504,SIP/sip504)
exten => 505,1,Macro(stdexten,505,SIP/sip505)
exten => 510,1,Macro(stdexten,510,SIP/sip510)
exten => 550,1,Macro(stdexten,550,SIP/sip550)
exten => 551,1,Macro(stdexten,551,SIP/sip551)
exten => 552,1,Macro(stdexten,552,SIP/sip552)
exten => 553,1,Macro(stdexten,553,SIP/sip553)
# dumb little tests
exten => 701,1,Goto(testgoodbye,s,1)
exten => 702,1,Goto(vm-options,s,1)
exten => 703,1,Goto(testmovies1,s,1)
exten => 704,1,Goto(testmovies2,s,1)
;Dynamic conference room 123 with quiet entering. User foo becomes the marked
;user if he dials extension '610'. Conference lasts until foo leaves. When foo
;exits other users who dialed extension '611' will be disconnected.
exten => 400,1,MeetMe(400|xqdeD)
exten => 400,1,MeetMe(400|AqdeD)
exten => 401,1,MeetMe(401|xqdeD)
exten => 401,1,MeetMe(401|AqdeD)
exten => 402,1,MeetMe(402|xqdeD)
exten => 402,1,MeetMe(402|AqdeD)
; Marca 700 si quieres que te conteste el inicio
exten => 700,1,Goto(s,restart)
; correo de voz
exten => 8500,1,VoicemailMain
; Da un aviso amigable de que se equivocaron y reinicia el proceso
;pbx-invalid Im sorry, that is not an invalid extension, please try again
;Nombregab-invalid Lo siento, esa extension es incorrecta, por favor vuelva a intentar
exten => _.,1,Set(CHANNEL(language)=es)
exten => _.,n,Playback(pbx-invalid)
exten => _.,n,Goto(s,restart)
; #### #### #### #### #### #### #### #### #### #### #### ####
;
;
;
;
; 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)
; mis decisiones hasta ahora
blindxfer => #
atxfer => *2
[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
;
;
; 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
;
; Music on hold class definitions
;
[classes]
default => quietmp3:/var/lib/asterisk/mohmp3
;loud => mp3:/var/lib/asterisk/mohmp3
;random => quietmp3:/var/lib/asterisk/mohmp3,-z
;unbuffered => mp3nb:/var/lib/asterisk/mohmp3
;quietunbuf => quietmp3nb:/var/lib/asterisk/mohmp3
; Note that the custom mode cannot handle escaped parameters (specifically embedded spaces)
;manual => custom:/var/lib/asterisk/mohmp3,/usr/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s
;
; File-based (native) music on hold
;
; This plays files directly from the specified directory, no external
; processes are required. Files are played in normal sorting order
; (same as a sorted directory listing), and no volume or other
; sound adjustments are available. If the file is available in
; the same format as the channel's codec, then it will be played
; without transcoding (same as Playback would do in the dialplan).
; Files can be present in as many formats as you wish, and the
; 'best' format will be chosen at playback time.
;
; NOTE:
; If you are not using "autoload" in modules.conf, then you
; must ensure that the format modules for any formats you wish
; to use are loaded _before_ res_musiconhold. If you do not do
; this, res_musiconhold will skip the files it is not able to
; understand when it loads.
;
[moh_files]
;native => /var/lib/asterisk/moh-native
;native-random => /var/lib/asterisk/moh-native,r
;
;
;
;
; 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@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
[sip500]
type=friend
secret=ext500
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=500@default,500
;callgroup=1,3-4 ; we are member of the call groups 1, 3 and 4
;pickupgroup=1,3-4 ; we can do call pick-up with *8 for the callgroups 1, 3 and 4
callgroup=1 ; we are member of the call groups 1, 3 and 4
pickupgroup=1 ; we can do call pick-up with *8 for the callgroups 1, 3 and 4
; Extension 601
[sip501]
type=friend
secret=ext501
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=501@default,501
callgroup=1 ; we are member of the call group 1
pickupgroup=1 ; we can do call pick-up with *8 for the callgroup 1
; Extension 602
[sip502]
type=friend
secret=ext502
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=502@default,502
callgroup=1 ; we are member of the call group 1
pickupgroup=1 ; we can do call pick-up with *8 for the callgroup 1
; Extension 603
[sip503]
type=friend
secret=ext503
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=503@default,503
callgroup=1 ; we are member of the call group 1
pickupgroup=1 ; we can do call pick-up with *8 for the callgroup 1
; Extension 604
[sip504]
type=friend
secret=ext504
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=504@default,504
callgroup=1 ; we are member of the call group 1
pickupgroup=1 ; we can do call pick-up with *8 for the callgroup 1
; Extension 605
[sip505]
type=friend
secret=ext505
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=505@default,505
callgroup=1 ; we are member of the call group 1
pickupgroup=1 ; we can do call pick-up with *8 for the callgroup 1
[sip510]
type=friend
context=international
callerid="Nombre Apellido" <610>
username=sip610
secret=ext510
nat=no
canreinvite=yes
dtmfmode=info
mailbox=510@default
disallow=all
allow=ulaw
allow=alaw
allow=g729
callgroup=1 ; we are member of the call group 1
pickupgroup=1 ; we can do call pick-up with *8 for the callgroup 1
[sip550]
type=friend
callerid="XLite Apellido remote" <650>
host=dynamic ; This device needs to register
username=sip650
secret=ext550
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
callgroup=1 ; we are member of the call group 1
pickupgroup=1 ; we can do call pick-up with *8 for the callgroup 1
[sip551]
type=friend
callerid="SoftPhone de Nombre Apellido" <651>
host=dynamic ; This device needs to register
username=sip651
secret=ext551
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
callgroup=1 ; we are member of the call group 1
pickupgroup=1 ; we can do call pick-up with *8 for the callgroup 1
[sip552]
type=friend
callerid="SoftPhone de Nombre correo Cabrera" <652>
host=dynamic ; This device needs to register
username=sip652
secret=ext552
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
callgroup=1 ; we are member of the call group 1
pickupgroup=1 ; we can do call pick-up with *8 for the callgroup 1
[sip553]
type=friend
callerid="xlite de Nombre correo Cabrera" <653>
host=dynamic ; This device needs to register
username=sip653
secret=ext553
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
callgroup=1 ; we are member of the call group 1
pickupgroup=1 ; we can do call pick-up with *8 for the callgroup 1
;
;
;
;
; begin voicemail
;
;
;
;
;
; Voicemail Configuration
;
[general]
; Default formats for writing Voicemail
format=wav ;format=g723sf|wav49|wav
serveremail=soporte@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]
500 => 500,Nombre Apellido,correo@mailinator.com,,operator=yes|attach=yes
501 => 501,Nombre Apellido,correo@mailinator.com,,operator=yes|attach=yes
502 => 502,Nombre correo,correo@mailinator.net,,operator=yes|attach=yes
503 => 503,Nombre Apellido,correo@mailinator.com,,operator=yes|attach=yes
504 => 504,cuarto sip,4@sip.com,,operator=yes|attach=yes
505 => 505,quinto sip,5@sip.com,,operator=yes|attach=yes
510 => 510,Nombre Apellido,correo@mailinator.com,,operator=yes|attach=yes
550 => 550,Nombre Apellido remote,correo@mailinator.com,,operator=yes|attach=yes
550 => 550,Nombre correo remote,correo@mailinator.net,,operator=yes|attach=yes
[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=2.0
txgain=2.0
immediate=no
context=default
signalling=fxs_ks
callerid=asreceived
callprogress=no
musiconhold=yes
channel => 1