ITC-2300 VoIP Labs Switch and Router Configurations

Notes

Don’t forget to modify for the information specific to your Pod:

  • Change the hostname of both the Router and Switch to include your Pod ID

  • Change the extension numbers/phone numbers to those appropriate for your pod

  • Read through all lines in the configurations, especially comment lines starting with a !

You may want to make some additional changes for administration ease:

  • Set an enable secret

  • Setup a username and secret on both devices

  • Setup SSH on both devices

Basic Router Configuration

hostname PodX-Router
!
interface GigabitEthernet0/0/0
 ip address 192.168.1.5 255.255.255.0
 no shutdown
!
ip route 0.0.0.0 0.0.0.0 192.168.1.1
!

Basic Switch Configuration

hostname PodX-Switch
!
vtp mode transparent
!
spanning-tree portfast default
!

Adding ISDN <→ SIP Gateway Support to the Router

In order to setup VoIP services on your router you will need to verify the uck9 features are enabled. In our lab environment you want to see that RightToUse is yes for uck9.

  1. Verify uck9 is yes for RightToUse

    Router#show license feature
Feature name             Enforcement  Evaluation  Subscription   Enabled  RightToUse
appxk9                   yes          yes         no             yes      yes
uck9                     yes          yes         no             yes      yes
securityk9               yes          yes         no             yes      yes
ipbasek9                 no           no          no             yes      no
FoundationSuiteK9        yes          yes         no             yes      yes
AdvUCSuiteK9             yes          yes         no             no       yes
cme-srst                 yes          yes         no             no       yes
hseck9                   yes          no          no             no       no
throughput               yes          yes         no             no       yes
internal_service         yes          no          no             no       no

Router#

  2. If uck9 is not yes for RightToUse set it up

    Router#configure terminal
Enter configuration commands, one per line.  End with CNTL/Z.
Router(config)#license boot level uck9
% use 'write' command to make license boot config take effect on next boot
Router(config)#exit
Router#write memory
Router#reload

Note that you may need to accept a EULA after running the license boot level uck9 command. Also you must save the configuation write memory or copy running-config startup-config and reload the router in order to activate the license.

These additions to the configuration of your router allow it to act as a VoIP gateway which converts a digital ISDN PRI trunk from the Public Switched Telephone Network (PSTN) to a SIP trunk which Asterisk can send and receive calls on. This means that calls sent to the router (only from the IP of our Asterisk server thanks to the ACLs) are able to be sent to the PSTN over a T1 PRI voice trunk with guaranteed quality and calls coming from the PSTN can be converted into VoIP by the router and sent on to Asterisk.

! Put the T1/E1 card found in slot 0 , subslot 1 into T1 mode
card type t1 0 1
!
! 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 T1 card
network-clock synchronization automatic
!
! Enable SIP VoIP capabilities on the router and set the router to use the correct interface for SIP signalling
voice service voip
 sip
  bind control source-interface GigabitEthernet 0/0/0
  bind media source-interface GigabitEthernet 0/0/0
!
! Configure the T1 settings for slot 0, in subslot 1, on port 0 to use the correct modes for the ISDN switch and to use all 24 timeslots in the ISDN PRI T1 connection
controller T1 0/1/0
 framing esf
 clock source line primary
 linecode b8zs
 cablelength short 110
 pri-group timeslots 1-24
!
! Send calls to any phone number received by the router (from Asterisk) to the PSTN over the T1.
dial-peer voice 1 pots
 description Calls from Asterisk to PSTN
 destination-pattern .T
 port 0/1/0:23
!
! 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 ipv4:192.168.1.10:5060
 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 0/1/0:23
!

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