Cisco VG224 con Asterisk

Comparto la configuración de un cisco vg224 usando el protocolo SIP para comunicarse con un servidor asterisk, no se si con MGCP se puede tener una mayor integracion, agradecere mayor informacion si alguien ya realizo alguna configuracion de esta manera

Saludos,

version 12.4
no service pad
service timestamps debug datetime msec localtime show-timezone
service timestamps log datetime msec localtime show-timezone
no service password-encryption
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hostname VG224
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boot-start-marker
boot-end-marker
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logging message-counter syslog
logging buffered 16000
enable password cisco
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no aaa new-model
clock timezone GMT -5
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ip source-route
ip cef
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voice call send-alert
voice rtp send-recv
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voice service pots
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voice service voip
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voice class codec 1
codec preference 1 g711ulaw
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voice-card 0
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application
service dsapp
param callWaiting TRUE
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global
service default dsapp
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archive
log config
hidekeys
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interface FastEthernet0/0
ip address 192.168.202.33 255.255.255.0 ===> IP del VG224
duplex auto
speed auto
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interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
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ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 192.168.202.1 ===> IP del mi router
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ip http server
no ip http secure-server
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control-plane
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voice-port 2/0
mwi
timeouts call-disconnect 5
station-id name Charly
station-id number 1980
caller-id enable
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voice-port 2/1
mwi
timeouts call-disconnect 5
station-id name Nito
station-id number 1981
caller-id enable
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voice-port 2/2
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voice-port 2/3
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voice-port 2/4
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voice-port 2/5
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voice-port 2/6
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voice-port 2/7
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voice-port 2/8
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voice-port 2/9
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voice-port 2/10
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voice-port 2/11
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voice-port 2/12
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voice-port 2/13
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voice-port 2/14
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voice-port 2/15
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voice-port 2/16
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voice-port 2/17
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voice-port 2/18
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voice-port 2/19
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voice-port 2/20
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voice-port 2/21
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voice-port 2/22
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voice-port 2/23
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dial-peer voice 10 pots
destination-pattern 1980
port 2/0
authentication username 1980 password 12345 ===> Extensión 1980 creada en mi asterisk
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dial-peer voice 11 pots
destination-pattern 1981
port 2/1
authentication username 1981 password 12345 ===> Extensión 1981 creada en mi asterisk
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dial-peer voice 12 pots
port 2/2
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dial-peer voice 13 pots
port 2/3
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dial-peer voice 14 pots
port 2/4
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dial-peer voice 15 pots
port 2/5
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dial-peer voice 16 pots
port 2/6
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dial-peer voice 17 pots
port 2/7
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dial-peer voice 18 pots
port 2/8
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dial-peer voice 19 pots
port 2/9
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dial-peer voice 20 pots
port 2/10
!
dial-peer voice 21 pots
port 2/11
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dial-peer voice 22 pots
port 2/12
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dial-peer voice 23 pots
port 2/13
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dial-peer voice 24 pots
port 2/14
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dial-peer voice 25 pots
port 2/15
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dial-peer voice 26 pots
port 2/16
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dial-peer voice 27 pots
port 2/17
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dial-peer voice 28 pots
port 2/18
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dial-peer voice 29 pots
port 2/19
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dial-peer voice 30 pots
port 2/20
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dial-peer voice 31 pots
port 2/21
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dial-peer voice 32 pots
port 2/22
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dial-peer voice 33 pots
port 2/23
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dial-peer voice 100 voip
destination-pattern 1...
voice-class codec 1
session protocol sipv2
session target sip-server
dtmf-relay sip-notify rtp-nte
no vad
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sip-ua
authentication username cisco password cisco123
retry invite 3
retry response 3
retry bye 3
retry cancel 3
timers trying 1000
mwi-server ipv4:192.168.202.5 expires 3600 port 5060 transport udp
registrar ipv4:192.168.202.5:5060 expires 3600
sip-server ipv4:192.168.202.5 ==> IP Server Asterisk
!
.

Ahora creamos un Trunk SIP en nuestro asterisk
PEER
allow=ulaw
canreinvite=no
context=from-internal
disallow=all
dtmfmode=rfc2833
host=192.168.202.33
insecure=very
qualify=yes
secret=cisco123
type=peer
username=cisco

USER
allow=ulaw
canreinvite=no
context=from-internal
disallow=all
dtmfmode=rfc2833
host=192.168.202.33
qualify=yes
type=user

En el VG224 verificamos que los anexos creados esten registrados en nuestro asterisk:
VG224#show sip register status
Line peer expires(sec) registered
============ ============= ============ ===========
1980 10 33 yes
1981 11 33 yes