ITC-2300 VoIP Labs Switch and Router Configurations

hostname PodX-Router
!
ip dhcp excluded-address 192.168.10.1 192.168.10.20
!
ip dhcp pool internal-voip
   network 192.168.10.0 255.255.255.0
   dns-server 172.17.139.11
   default-router 192.168.10.1
   ! Option 150 is used by Cisco phones to find the TFTP server where they get their configuration.
   ! We are running the TFTP server on our Asterisk server so we can use the IP of that server.
   option 150 ip 192.168.10.3
!
interface FastEthernet0/0
 ! Rememeber to change the IP address on this interface to match the IP address assigned to your pod.
 ip address 172.17.144.XX 255.255.255.0
 ip nat outside
 no shutdown
!
interface FastEthernet0/1
 no shutdown
!
interface FastEthernet0/1.10
 encapsulation dot1Q 10
 ip address 192.168.10.1 255.255.255.0
 ip nat inside
 no shutdown
!
ip route 0.0.0.0 0.0.0.0 172.17.144.1
!
ip nat inside source list 1 interface FastEthernet0/0 overload
!
access-list 1 permit 192.168.10.0 0.0.0.255

hostname PodX-Switch
!
vtp mode transparent
!
vlan 10
!
spanning-tree portfast default
!
!
interface GigabitEthernet1/0/1
 description Lab PC
 switchport access vlan 10
 switchport mode access
 spanning-tree portfast
!
interface GigabitEthernet1/0/2
 description VoIP Phone 1
 switchport access vlan 10
 switchport mode access
 switchport voice vlan 10
 spanning-tree portfast
!
interface GigabitEthernet1/0/3
 description VoIP Phone 2
 switchport access vlan 10
 switchport mode access
 switchport voice vlan 10
 spanning-tree portfast
!
interface GigabitEthernet1/0/23
 description Layer 2 connection to Asterisk server ens224 interface on internal 192.168.10.0/24 subnet
 switchport access vlan 10
 switchport mode access
 no shutdown
!
interface GigabitEthernet1/0/24
 description Trunk Connection to Router
 switchport mode trunk
 no shutdown
!
interface Vlan10
 ip address 192.168.10.2 255.255.255.0
!
ip route 0.0.0.0 0.0.0.0 192.168.10.1
!

! Put the T1/E1 card found in slot 0 , subslot 3 into T1 mode
card type t1 0 3
!
! Setup ISDN on the router to use signaling for PRI connections to a National ISDN switch
isdn switch-type primary-ni
!
! Tell the router to synchronize the DSP clock to the card in WIC slot 3
network-clock-participate wic 3
!
! Enable SIP VoIP capabilities on the router and set the router to use the IP outside network for SIP signalling because that is where Asterisk is
voice service voip
 sip
  bind control source-interface FastEthernet0/1.10
  bind media source-interface FastEthernet0/1.10
!
! Configure the T1 settings on port 0 in subslot 3, slot 0 to use the correct modes for the ISDN switch and to use only the first 6 channels on the T1 PRI (due to limited DSP resources on our routers)
controller T1 0/3/0
 framing esf
 linecode b8zs
 clock source line
 pri-group timeslots 1-6
!
! Prevent any connections to the router's SIP port from the outside network
ip access-list extended block-sip-outside
 ! change the next two lines to have the correct outside IP address for your router
 deny   tcp any host 172.17.144.XX eq 5060
 deny   udp any host 172.17.144.XX eq 5060
 permit ip any any
!
! Prevent any connections to the router's SIP port from the inside EXCEPT from the Asterisk server
ip access-list extended block-sip-inside
 permit udp host 192.168.10.3 host 192.168.10.1 eq 5060
 deny   tcp any host 192.168.10.1 eq 5060
 deny   udp any host 192.168.10.1 eq 5060
 permit ip any any
!
! Apply the outside ACL
interface FastEthernet0/0
 ip access-group block-sip-outside in
!
! Apply the inside ACL
interface FastEthernet0/1.10
 ip access-group block-sip-inside in
!
! Send calls to any phone number received by the router (from Asterisk) to the PSTN over the T1.
!   Note this is why we need the ACLs to only allow access from Asterisk, otherwise anyone could send a SIP call to the router out over our T1 which could cause us to be charged!
dial-peer voice 1 pots
 description Calls from Asterisk to PSTN
 destination-pattern .T
 port 0/3/0:23
!
! Send calls to 510555ZZ.. numbers to the Asterisk server using SIP to route further
dial-peer voice 2 voip
 description Calls from PSTN to Asterisk
 ! change the two ZZs in the next line to match the incoming numbers for your pod (the last two characters should be periods to match any number)
 destination-pattern 510555ZZ..
 session protocol sipv2
 ! The IP in the next line should be the internal network IP of your Asterisk server
 session target ipv4:192.168.10.3:5060
 incoming called-number .T
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
!
dial-peer voice 3 pots
 description Passes inbound calls from PSTN on to dial peer 2 which will send them to Asterisk
 ! change the two ZZs in the next line to match the incoming numbers for your pod (the last two characters should be periods to match any number)
 incoming called-number 510555ZZ..
 direct-inward-dial
 port 0/3/0:23
!