;
; 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