OK thank you I have upgraded my config as you are saying, I have followed the steps you have said and the steps in the link and now this is my config:
!
version 12.4
service config
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname Router
!
boot-start-marker
boot-end-marker
!
!
!
resource policy
!
no aaa new-model
no network-clock-participate slot 1
no network-clock-participate wic 0
ip subnet-zero
!
!
ip cef
ip name-server 80.58.0.33
no ip dhcp use vrf connected
!
ip dhcp pool ITS
network 192.168.3.0 255.255.255.0
option 150 ip 192.168.3.1
default-router 192.168.3.1
!
!
!
!
!
voice service voip
sip
!
!
!
!
!
!
!
!
!
voice translation-rule 1
rule 1 /2../ /964812530/
!
voice translation-rule 3
rule 1 /964812530/ /200/
!
!
voice translation-profile SIPout
translate calling 1
!
voice translation-profile incoming
translate called 3
!
!
voip-incoming translation-profile incoming
!
application
service autoatt flash:app-b-acd-aa-2.1.0.0.tcl
param operator 6001
paramspace english language en
paramspace english index 0
paramspace english location flash:
param aa-pilot 6060
!
!
!
!
!
!
interface FastEthernet0/0
ip address 192.168.3.1 255.255.255.0
ip nat inside
ip virtual-reassembly
duplex auto
speed auto
!
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
!
interface Ethernet1/0
ip address 192.168.2.200 255.255.255.0
ip nat outside
ip virtual-reassembly
half-duplex
!
ip http server
ip classless
ip route 0.0.0.0 0.0.0.0 192.168.2.1
!
!
ip nat inside source list 1 interface Ethernet1/0 overload
!
access-list 1 permit 192.168.3.0 0.0.0.255
no cdp log mismatch duplex
!
!
!
control-plane
!
!
!
!
!
!
!
!
dial-peer voice 100 voip
translation-profile outgoing SIPout
destination-pattern 9........
session protocol sipv2
session target sip-server
codec g711ulaw
!
dial-peer voice 101 voip
translation-profile outgoing SIPout
destination-pattern 6........
session protocol sipv2
session target sip-server
codec g711ulaw
!
dial-peer voice 1001 voip
service autoatt
session protocol sipv2
session target sip-server
incoming called-number 1000
dtmf-relay rtp-nte
codec g711ulaw
!
sip-ua
authentication username xxx password xxx
no remote-party-id
retry invite 4
retry response 3
retry bye 2
retry cancel 2
retry register 5
timers register 250
registrar ipv4:213.162.201.146 expires 60
sip-server ipv4:213.162.201.146
!
!
gatekeeper
shutdown
!
!
telephony-service
max-ephones 3
max-dn 4
ip source-address 192.168.3.1 port 2000
auto assign 1 to 3
user-locale ES
network-locale ES
create cnf-files version-stamp Jan 01 2002 00:00:00
max-conferences 4 gain -6
transfer-system full-consult
!
!
ephone-dn 1 dual-line
number 200 no-reg primary
!
!
ephone-dn 2 dual-line
number 201 no-reg primary
!
!
ephone-dn 3 dual-line
number 964812530
!
!
ephone-dn 4
number 1000
!
!
ephone 1
mac-address 0012.431E.BF3E
type 7902
button 1:1
!
!
!
ephone 2
mac-address 0013.60C3.CE02
type 7902
button 1:2
!
!
!
ephone 3
!
!
!
line con 0
line aux 0
line vty 0 4
password 3Dx25.8*
login
!
!
end
But autoatt it's not running, when I have put this commands I have had warnings:
Router(config-app-param)#param operator 6001
Warning: parameter operator has not been registered under autoatt namespace
Router(config-app-param)#param aa-pilot 6060
Warning: parameter aa-pilot has not been registered under autoatt namespace
Is necessary to create a new ephone-dn with number 1000? Why I have this warnings?
Please you help will be great.
Thanks in advanced.
