Router Cisco FXO con Asterisk

Comparto la configuración de un router cisco con puertos FXO usando el protocolo SIP para ser usado como gateway de un servidor asterisk.

Saludos,

version 12.4
service timestamps debug datetime msec localtime show-timezone
service timestamps log datetime msec localtime show-timezone
no service password-encryption
!
hostname gateway-asterisk
!
boot-start-marker
boot-end-marker
!
logging buffered 16000
enable password cisco
!
no aaa new-model
clock timezone GMT -5
ip telnet tos 0
!
!
ip cef
!
!
multilink bundle-name authenticated
!
!
voice-card 0
no dspfarm
!
!
voice call send-alert
no voice call carrier capacity active
voice rtp send-recv
!
voice service pots
!
voice service voip
allow-connections sip to sip
sip
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
registrar server
!
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
!
!
!
!
!
!
application
service dsapp
param callWaiting TRUE
!
global
service default dsapp
!
!
!
!
!
!
interface GigabitEthernet0/0
ip address 192.168.1.15 255.255.255.0 => ip del router
ip route-cache policy
ip policy route-map VOZ
duplex auto
speed auto
no keepalive
!
interface GigabitEthernet0/1
no ip address
shutdown
duplex auto
speed auto
!
ip route 0.0.0.0 0.0.0.0 192.168.1.1
!
!
ip http server
no ip http secure-server
!
access-list 100 permit ip any any
route-map VOZ permit 10
match ip address 100
set ip precedence critical
!
!
!
!
control-plane
!
!
!
voice-port 0/0/0
supervisory disconnect dualtone mid-call
cptone PE
timing hookflash-out 500
connection plar 1100 => envio al ivr, cola 1100
caller-id enable
!
voice-port 0/0/1
supervisory disconnect dualtone mid-call
cptone PE
timing hookflash-out 500
connection plar 1100 => envio al ivr, cola 1100
caller-id enable
!
voice-port 0/0/2
supervisory disconnect dualtone mid-call
cptone PE
timing hookflash-out 500
connection plar 1100 => envio al ivr, cola 1100
caller-id enable
!
voice-port 0/0/3
supervisory disconnect dualtone mid-call
cptone PE
timing hookflash-out 500
connection plar 1100 => envio al ivr, cola 1100
caller-id enable
!
voice-port 0/1/0
supervisory disconnect dualtone mid-call
cptone PE
timing hookflash-out 500
connection plar 1112 => envio directo a un anexo 1112
caller-id enable
!
voice-port 0/1/1
supervisory disconnect dualtone mid-call
cptone PE
timing hookflash-out 500
connection plar 1113 => envio directo a un anexo 1113
caller-id enable
!
voice-port 0/1/2
supervisory disconnect dualtone mid-call
cptone PE
timing hookflash-out 500
connection plar 3114 => envio directo a un anexo 1114
caller-id enable
!
voice-port 0/1/3
supervisory disconnect dualtone mid-call
cptone PE
timing hookflash-out 500
connection plar 3115 => envio directo a un anexo 1115
caller-id enable
!
!
no ccm-manager fax protocol cisco
!
no mgcp package-capability res-package
no mgcp package-capability fxr-package
no mgcp timer receive-rtcp
!
!
!
dial-peer voice 10 pots
description LLAMADAS FIJOS
incoming called-number .
destination-pattern [2-8]......
no digit-strip
direct-inward-dial
port 0/0/1
forward-digits all
!
dial-peer voice 11 pots
description LLAMADAS A MOVILES
incoming called-number .
destination-pattern 9........
no digit-strip
direct-inward-dial
port 0/0/1
forward-digits all
!
dial-peer voice 12 pots
description LLAMADAS FIJOS
incoming called-number .
destination-pattern [2-8]......
no digit-strip
direct-inward-dial
port 0/0/2
forward-digits all
!
dial-peer voice 13 pots
description LLAMADAS A MOVILES
incoming called-number .
destination-pattern 9........
no digit-strip
direct-inward-dial
port 0/0/2
forward-digits all
!
dial-peer voice 14 pots
description LLAMADAS FIJOS
incoming called-number .
destination-pattern [2-8]......
no digit-strip
direct-inward-dial
port 0/0/3
forward-digits all
!
dial-peer voice 15 pots
description LLAMADAS A MOVILES
incoming called-number .
destination-pattern 9........
no digit-strip
direct-inward-dial
port 0/0/3
forward-digits all
!
dial-peer voice 100 voip
description LLAMADAS ANEXOS SIP
incoming called-number .
destination-pattern 11.. => todos mis anexos son 11XX
voice-class codec 1
session protocol sipv2
session target ipv4:192.168.1.17 => ip del servidor asterisk
session transport udp
dtmf-relay sip-notify rtp-nte
no vad
!
!
gateway
timer receive-rtp 1200
!
sip-ua
authentication username cisco password asterisk
retry invite 3
retry response 3
retry bye 3
retry cancel 3
retry options 0
timers trying 1000
mwi-server ipv4:192.168.1.17 expires 3600 port 5060 transport udp
registrar ipv4:192.168.1.17 expires 3600
sip-server ipv4:192.168.1.17
!
line con 0
password cisco
login
line aux 0
line vty 0 4
password cisco
login
!
scheduler allocate 20000 1000
!
end

Ahora creamos un Trunk SIP en nuestro asterisk

PEER
allow=ulaw
canreinvite=no
context=from-internal
disallow=all
dtmfmode=rfc2833
host=192.168.1.15
insecure=very
qualify=yes
secret=asterisk
type=peer
username=cisco

USER
allow=ulaw
canreinvite=no
context=from-internal
disallow=all
dtmfmode=rfc2833
host=192.168.1.15
qualify=yes
type=user

Creo que con esto, mas la configuracion de un cisco VG224 se demuestra que se puede reutilizar toda la infraestructura cisco para que funcione con asterisk, tal como lo hicieron en la universidad del sur de texas ...... no es tan dificil :)