configuracion de asterisk 1.6.0.1

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Buenas noches. Estoy instalando asterisk 1.6.0.1 en linux fedora core 9. El servidor esta corriendo pero el problema los tengo en los archivos sip.conf y extensions.conf pues son diferentes a las otras versiones de asterisk, es decir, extensions.conf.sample y sip.conf.sample. Ademas cambia la ruta en donde se encuebtran esos archivos, pues ya no estan en etc/extensions.conf y sip.conf. Necesito ayuda para configurar estos archivos gracias

La ruta es

Imagen de RazaMetaL

La ruta es /etc/asterisk/sip.conf y /etc/asterisk/extensions.conf

Debes mirar el wiki de www.voip-info.org ahí encontrarás toda la ayuda que necesitas.

-----

Visita este [url=http://www.sindominio.net/ayuda/preguntas-inteligentes.html]link[/url] :evil:

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Configuracion de asterisk 1.6.0.1

Gracias amigo por ayudarme pero esa es la ruta de la version 1.4.xx y yo estoy con 1.6.0.1 y en esa ruta no estan ni el extensions.conf y sip.conf. La ruta en la version 1.6.0.1 es la siguiente /etc/asterisk-1.6.0.1/configs y ahora se llaman extensions.conf.sample y sip.conf.sample. Pregunto sera que editandolos dara el mismo resultado que con las versiones viejas de asterisk??. Gracias

que raro

Imagen de eliche

Hola que tal? bueno yo he trabajado con el asterisk 1.6 y la verdad la ubicación de los archivos sip.conf y extensions.conf no cambia, sigue siendo: /etc/asterisk/*.conf.

La verdad te recomiendo que observes bien si están en ese directorio, y además que los edites pues si funciona...solo renombra o crea un nuevo sip.conf en ese directorio y ya estaría todo listo...Una cosa más, el asterisk lo compilaste o lo instalaste desde paquetes? Pues el asterisk compilado si te crea el /etc/asterisk/

Saludos!

Ing. Eliécer Tatés Montenegro
Neovoice
VoIP & IT Consulting
e-mail: eliecer.tates@neovoice.co / elitatmon@gmail.com

Configuracion de asterisk 1.6.0.1

Si amigo correcto lo compile, te explico lo que hice como root:

1) lo descomprimi con tar -zxvf asterisk-1.6.0.1.tar

2) borro el tar con rm -f asterisk-1.6.0.1.tar

3) Ejecute el ./configure segun lo que dice el archivo README

4) ejecute el make

5) Make install

6) Ejecuto el servidor asterisk y todo bien, pero no en el directorio /etc/asterisk sino en /etc/asterisk-1.6.0.1/configs y no se llaman extensions.conf y sip.conf sino extensions.conf.samples y sip.conf.sample. A mi tamboien me parece raro pero por eso les pregunto a ver si es normal y si me pueden ayudar. Gracias a todos

La instalación por default de

Imagen de acl

La instalación por default de asterisk creó esos archivos *.sample porque son muestras de cómo escribir el archivo. Puedes cambiarles de nombre a extensions.conf y sip.conf sin problema (y luego editarles para darles la forma que quieras). Las extensiones en unix no significan nada, son parte del nombre.

A veces los paquetes no vienen con archivos por defecto que puedes usar a la primera, sino que tienes que ajustarlos a tu ambiente. Tal es el caso de asterisk y te dan las muestras para obligarte a pensar cuando escribas tu archivo.

extension.conf en asterisk 1.6.0.1

Buenas estoy haciendo un prototipo de asterisk para mi defensa de tesis. Ya instale el asterisk 1.6.0.1 pero no se en que parte del archivo extensions.conf configurar las extensiones pues existen varias secciones y no se donde es. Gracias por la ayuda

extensions.conf en asterisk 1.6.0.1

Esta es una copia del archivo extensions.conf y la pregunta es: ¿en que parte especificamente agrego las extensiones??. Gracias nuevamente

; extensions.conf - the Asterisk dial plan
;
; 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 "dialplan 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 "dialplan save" 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. This is the default.
;
; 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=no
;
;
;
; If extenpatternmatchnew is set (true, yes, etc), then a new algorithm that uses
; a Trie to find the best matching pattern is used. In dialplans
; with more than about 20-40 extensions in a single context, this
; new algorithm can provide a noticeable speedup.
; With 50 extensions, the speedup is 1.32x
; with 88 extensions, the speedup is 2.23x
; with 138 extensions, the speedup is 3.44x
; with 238 extensions, the speedup is 5.8x
; with 438 extensions, the speedup is 10.4x
; With 1000 extensions, the speedup is ~25x
; with 10,000 extensions, the speedup is 374x
; Basically, the new algorithm provides a flat response
; time, no matter the number of extensions.
;
; By default, the old pattern matcher is used.
;
; ****This is a new feature! *********************
; The new pattern matcher is for the brave, the bold, and
; the desperate. If you have large dialplans (more than about 50 extensions
; in a context), and/or high call volume, you might consider setting
; this value to "yes" !!
; Please, if you try this out, and are forced to return to the
; old pattern matcher, please report your reasons in a bug report
; on bugs.digium.com. We have made good progress in providing something
; compatible with the old matcher; help us finish the job!
;
; This value can be switched at runtime using the cli command "dialplan set extenpatternmatchnew true"
; or "dialplan set extenpatternmatchnew false", so you can experiment to your hearts content.
;
;extenpatternmatchnew=no
;
; If clearglobalvars is set, global variables will be cleared
; and reparsed on a dialplan 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.
;
; NOTE: A complication sets in, if you put your global variables into
; the AEL file, instead of the extensions.conf file. With clearglobalvars
; set, a "reload" will often leave the globals vars cleared, because it
; is not unusual to have extensions.conf (which will have no globals)
; load after the extensions.ael file (where the global vars are stored).
; So, with "reload" in this particular situation, first the AEL file will
; clear and then set all the global vars, then, later, when the extensions.conf
; file is loaded, the global vars are all cleared, and then not set, because
; they are not stored in the extensions.conf file.
;
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=yes
;
; User context is where entries from users.conf are registered. The
; default value is 'default'
;
;userscontext=default
;
; You can include other config files, use the #include command
; (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 the GLOBAL dialplan function:
; ${GLOBAL(VARIABLE)}
; ${${GLOBAL(VARIABLE)}} or ${text${GLOBAL(VARIABLE)}} or any hybrid
; Unix/Linux environmental variables can be reached with the ENV dialplan
; function: ${ENV(VARIABLE)}
;
[globals]
CONSOLE=Console/dsp ; Console interface for demo
;CONSOLE=DAHDI/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=DAHDI/G2 ; Trunk interface
;
; Note the 'G2' in the TRUNK variable above. It specifies which group (defined
; in dahdi.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 DAHDI channel
; (aka. ascending sequential hunt group).
; G: select the highest-numbered non-busy DAHDI 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@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
;
; For example the extension _NXXXXXX would match normal 7 digit dialings,
; while _1NXXNXXXXXX would represent an area code plus phone number
; preceded by a one.
;
; Each step of an extension is ordered by priority, which must
; always start with 1 to be considered a valid extension. The priority
; "next" or "n" means the previous priority plus one, regardless of whether
; the previous priority was associated with the current extension or not.
; The priority "same" or "s" means the same as the previously specified
; priority, again regardless of whether the previous entry was for the
; same extension. Priorities may be immediately followed by a plus sign
; and another integer to add that amount (most useful with 's' or 'n').
; Priorities may then also have an alias, or label, in
; parenthesis after their name which can be used in goto situations
;
; 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|label{+|-}offset}[(alias)],application(arg1,arg2,...)
;
; Timing list for includes is
;
; ,,,
;
; Note that ranges may be specified to wrap around the ends. Also, minutes are
; fine-grained only down to the closest even minute.
;
;include => daytime,9:00-17:00,mon-fri,*,*
;include => weekend,*,sat-sun,*,*
;include => weeknights,17:02-8:58,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,Gosub(stdexten,s,1(6000,IAX2/foo))
;exten => 12564286000,n,Goto(default,s,1) ; exited Voicemail
;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(DAHDI/G2/${EXTEN:7}) ; Expose all of 256-428
;exten => _1256325XXXX,1,Dial(DAHDI/G2/${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
;
; Extension "s" is not a wildcard extension that matches "anything".
; In macros, it is the start extension. In most other cases,
; you have to goto "s" to execute that extension.
;
; For wildcard matches, see above - all pattern matches start with
; an underscore.
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/${GLOBAL(IAXINFO)}@iaxtel.com/${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,${EXTEN:4})
exten => _9011.,n,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})

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

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

[trunktollfree]
;
; Long distance context accessed through trunk interface
;
exten => _91800NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten => _91888NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten => _91877NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten => _91866NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(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 => trunklocal
include => iaxtel700
include => trunktollfree
include => iaxprovider

;Include parkedcalls (or the context you define in features conf)
;to enable call parking.
include => parkedcalls
;
; 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@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-trunkdial]
;
; Standard trunk dial macro (hangs up on a dialstatus that should
; terminate call)
; ${ARG1} - What to dial
;
exten => s,1,Dial(${ARG1})
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Hangup
exten => s-BUSY,1,Hangup
exten => _s-.,1,NoOp

[stdexten]
;
; Standard extension subroutine:
; ${ARG1} - Extension
; ${ARG2} - Device(s) to ring
; ${ARG3} - Optional context in Voicemail (if empty, then "default")
;
; Note that the current version will drop through to the next priority in the
; case of their pressing '#'. This gives more flexibility in what do to next:
; you can prompt for a new extension, or drop the call, or send them to a
; general delivery mailbox, or...
;
; The use of the LOCAL() function is purely for convenience. Any variable
; initially declared as LOCAL() will disappear when the innermost Gosub context
; in which it was declared returns. Note also that you can declare a LOCAL()
; variable on top of an existing variable, and its value will revert to its
; previous value (before being declared as LOCAL()) upon Return.
;
exten => s,1,NoOp(Start stdexten)
exten => s,n,Set(LOCAL(ext)=${ARG1})
exten => s,n,Set(LOCAL(dev)=${ARG2})
exten => s,n,Set(LOCAL(cntx)=${ARG3})

exten => s,n,Set(LOCAL(mbx)="${ext}"$["${cntx}" ? "@${cntx}" :: ""])
exten => s,n,Dial(${dev},20) ; Ring the interface, 20 seconds maximum
exten => s,n,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten => s-NOANSWER,1,Voicemail(${mbx},u) ; If unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,n,NoOp(Finish stdexten NOANSWER)
exten => s-NOANSWER,n,Return() ; If they press #, return to start

exten => s-BUSY,1,Voicemail(${mbx},b)
; If busy, send to voicemail w/ busy announce
exten => s-BUSY,n,NoOp(Finish stdexten BUSY)
exten => s-BUSY,n,Return() ; If they press #, return to start

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

exten => a,1,VoicemailMain(${mbx}) ; If they press *, send the user into VoicemailMain
; does this ever return?

[stdPrivacyexten]
;
; Standard extension subroutine:
; ${ARG1} - Extension
; ${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)`
; ${ARG5} - Context in voicemail (if empty, then "default")
;
; See above note in stdexten about priority handling on exit.
;
exten => s,1,NoOp(Start stdPrivacyexten)
exten => s,n,Set(LOCAL(ext)=${ARG1})
exten => s,n,Set(LOCAL(dev)=${ARG2})
exten => s,n,Set(LOCAL(dontcntx)=${ARG3})
exten => s,n,Set(LOCAL(tortcntx)=${ARG4})
exten => s,n,Set(LOCAL(cntx)=${ARG5})

exten => s,n,Set(LOCAL(mbx)="${ext}"$["${cntx}" ? "@${cntx}" :: ""])
exten => s,n,Dial(${dev},20,p) ; Ring the interface, 20 seconds maximum, call screening
; option (or use P for databased call screening)
exten => s,n,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten => s-NOANSWER,1,Voicemail(${mbx},u) ; If unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,n,NoOp(Finish stdPrivacyexten NOANSWER)
exten => s-NOANSWER,n,Return() ; If they press #, return to start

exten => s-BUSY,1,Voicemail(${mbx},b) ; If busy, send to voicemail w/ busy announce
exten => s-BUSY,n,NoOp(Finish stdPrivacyexten BUSY)
exten => s-BUSY,n,Return() ; If they press #, return to start

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

exten => s-TORTURE,1,Goto(${tortcntx},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(${mbx}) ; If they press *, send the user into VoicemailMain
; does this ever return?

[macro-page];
;
; Paging macro:
;
; Check to see if SIP device is in use and DO NOT PAGE if they are
;
; ${ARG1} - Device to page

exten => s,1,ChanIsAvail(${ARG1},js) ; j is for Jump and s is for ANY call
exten => s,n,GoToIf([${AVAILSTATUS} = "1"]?autoanswer:fail)
exten => s,n(autoanswer),Set(_ALERT_INFO="RA") ; This is for the PolyComs
exten => s,n,SIPAddHeader(Call-Info: Answer-After=0) ; This is for the Grandstream, Snoms, and Others
exten => s,n,NoOp() ; Add others here and Post on the Wiki!!!!
exten => s,n,Dial(${ARG1})
exten => s,n(fail),Hangup

[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,Gosub(stdexten,s,1(1234,${GLOBAL(CONSOLE)}))
exten => 1234,n,Goto(default,s,1) ; exited Voicemail

exten => 1235,1,Voicemail(1234,u) ; Right to voicemail

exten => 1236,1,Dial(Console/dsp) ; Ring forever
exten => 1236,n,Voicemail(1234,b) ; 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@pbx.digium.com/s@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

;
; You can use the Macro Page to intercom a individual user
exten => 76245,1,Macro(page,SIP/Grandstream1)
; or if your peernames are the same as extensions
exten => _7XXX,1,Macro(page,SIP/${EXTEN})
;
;
; System Wide Page at extension 7999
;
exten => 7999,1,Set(TIMEOUT(absolute)=60)
exten => 7999,2,Page(Local/Grandstream1@page&Local/Xlite1@page&Local/1234@page/n,d)

; 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)

;
; The page context calls up the page macro that sets variables needed for auto-answer
; It is in is own context to make calling it from the Page() application as simple as
; Local/{peername}@page
;
[page]
exten => _X.,1,Macro(page,SIP/${EXTEN})

;[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(6245,u) ; Voicemail (unavailable)
;exten => 6245,s+1,Hangup ; s+1, same as n
;exten => 6245,dial+101,Voicemail(6245,b) ; Voicemail (busy)
;exten => 6361,1,Dial(IAX2/JaneDoe,,rm) ; ring without time limit
;exten => 6389,1,Dial(MGCP/aaln/1@192.168.0.14)
;exten => 6390,1,Dial(JINGLE/caller/callee) ; Dial via jingle using labels
;exten => 6391,1,Dial(JINGLE/asterisk@digium.com/mogorman@astjab.org) ;Dial via jingle using asterisk as the transport and calling mogorman.
;exten => 6394,1,Dial(Local/6275/n) ; this will dial ${MARK}

;exten => 6275,1,Gosub(stdexten,s,1(6275,${MARK}))
; assuming ${MARK} is something like DAHDI/2
;exten => 6275,n,Goto(default,s,1) ; exited Voicemail
;exten => mark,1,Goto(6275,1) ; alias mark to 6275
;exten => 6536,1,Gosub(stdexten,s,1(6236,${WIL}))
; Ditto for wil
;exten => 6536,n,Goto(default,s,1) ; exited Voicemail
;exten => wil,1,Goto(6236,1)

;If you want to subscribe to the status of a parking space, this is
;how you do it. Subscribe to extension 6600 in sip, and you will see
;the status of the first parking lot with this extensions' help
;exten => 6600,hint,park:701@parkedcalls
;exten => 6600,1,noop
;
; Some other handy things are an extension for checking voicemail via
; voicemailmain
;
;exten => 8500,1,VoicemailMain
;exten => 8500,n,Hangup
;
; Or a conference room (you'll need to edit meetme.conf to enable this room)
;
;exten => 8600,1,Meetme(1234)
;
; Or playing an announcement to the called party, as soon it answers
;
;exten = 8700,1,Dial(${MARK},30,A(/path/to/my/announcemsg))
;

; example of a compartmentalized company called "acme"
;
; this is the context that your incoming IAX/SIP trunk dumps you in...
;[acme-incoming]
;exten => s,1,Wait(1)
;exten => s,n,Answer()
;exten => s,n(menu),Playback(acme/vm-brief-menu)
;exten => s,n(exten),Background(vm-enter-num-to-call)
;exten => s,n,WaitExten(5)
;exten => s,n(goodbye),Playback(vm-goodbye)
;exten => s,n(end),Hangup()
;
;include => acme-extens
;
;exten => i,1,Playback(vm-invalid)
;exten => i,n,Goto(s,exten) ; optionally, transfer to operator
;
;exten => t,1,Goto(s,goodbye)
;
; this is the context our internal SIP hardphones use (see sip.conf)
;
;[acme-internal]
;exten => s,1,Answer()
;exten => s,n(exten),Background(vm-enter-num-to-call)
;exten => s,n,WaitExten(5)
;exten => s,n(goodbye),Playback(vm-goodbye)
;exten => s,n(end),Hangup()
;
;include => trunkint
;include => trunkld
;include => trunklocal
;
;include => acme-extens
;
; you can test what your system sounds like to outside callers by dialing this
;exten => 777,1,DISA(no-password,acme-incoming)
;
; grouping of acme's extensions... never used directly, always included.
;
;[acme-extens]
;exten => 111,1,Gosub(stdexten,s,1(111,SIP/pete_1,acme))
;exten => 111,n,Goto(s,exten)
;
;exten => 112,1,Gosub(stdexten,s,1(112,SIP/nancy_1,acme))
;exten => 112,n,Goto(s,end)
;
; end of acme example

; For more information on applications, just type "core show applications" at your
; friendly Asterisk CLI prompt.
;
; "core show application " will show details of how you
; use that particular application in this file, the dial plan.
; "core show functions" will list all dialplan functions
; "core show function " will show you more information about
; one function. Remember that function names are UPPER CASE.

Cita: Esta es una copia del

Imagen de RazaMetaL

[quote]Esta es una copia del archivo extensions.conf y la pregunta es: ¿en que parte especificamente agrego las extensiones??. Gracias nuevamente[/quote]

En ningún lado de ese archivo. Las extensiones las creas en el archivo de configuración de cada canal (sip.conf/iax.conf/h323.conf/etc). El archivo extensions.conf contiene el dialplan.

Recomiendo léas el siguiente link:
[url]http://www.voip-info.org/wiki/view/Asterisk#Introduction[/url]

-----

Visita este [url=http://www.sindominio.net/ayuda/preguntas-inteligentes.html]link[/url] :evil:

[img]http://www.bbspot.com/Images/News_Features/2003/01/os_quiz/debian.jpg[/img]

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extension.conf en asterisk 1.6.0.1

OK gracias amigo entonces para nada toco el extensions.conf??. ya las configuro entonces. solo en este caso el sip.conf, pues solo necesito mediante un gateway voip y una central asterisk hacer sonar un telefono ip y del telefono al sofphone. Lo que pasa es que lei una tesis donde en el extensions.conf estan las extensiones y por eso pregunto. Tambien habia leido que en default y en el contextera donde se definian las extensiones. Gracias de verdad. Cualquier cosa vuelvo y pregunto

asterisk GUI

Buenas estoy tratandop de instalar asterisk GUI voy a un link donde supuestamente esta y hay varios directorios y archivos, es decir, no entiendo porque no esta el asterisk-gui. Me podrian ayudar. Gracias