cube_auto_deploy

Program to auto-deploy CUBE. This will capture all calls from the CUCM/ITSP and pass them out a necessary outbound dial-peer. This is achieved by matching on the URI of the CUCM and ITSP. The benefit of matching on URI rather than destination patterns, incoming called/calling, etc is that it allows this solution to be a almost a one size fits all solution. It doesn't matter what country the code is ran in so long as the IPs/DNS are provide this code will accept any type of dialling patterns e.g if it's +44, 900353, 0061, they'll all be accepted and depending on who they're sent from will determine the path the call will take. If you're following Cisco’s best practise of globalising the number to e.164 prior to arriving to the CUBE and the ITSP supports e.164 dialling then this program is the right fit for you. Yes there will be exceptions were certain providers will need SIP profiles and others registrar information, something that I will look to further add as I develop into the future but if you just require a standard CUBE stood up, this will achieve that goal. Codecs supported are g711u/alaw

Software Requirements

Python 3.8
NAPALM
Netmiko
requirements.txt

Running the code

git clone https://github.com/b-git-hub/cube_auto_deploy/cube_deploy.py
pip install requirements.txt 
python cube_deploy.py

Information required to run

The program assumes that you already have the below information 
IP Address of CUBE
Username/Password
ITSP media, signalling and gateway IP to provider
Signalling protocols being used

Operation

This program will configure CUBE and deploy 4 dial-peers, using server groups and dpgs any calls destined from the CUCM to the ITSP will automatically be sent to the ITSP and visa versa.
There isn't any manipulation of the calls being used, the program assumes that none is required.
There isn't any manipulation of SIP headers involved. If these are required, configure them seperately.
The program is very much still in beta. I've not implemented error handling to capture in the event users put in IPs instead of a signalling protocol.
This will come in future iterations, this for the moment will deploy your CUBE provided you follow the prompts correctly.

The operator of this code will need access to the CUBE via SSH

Call Flow

The call flow is very simple, it takes the call from CUCM via any means and passes it through the CUBE to the ITSP without any manipulation other than binding signalling and media which is required to send/recieve calls. The program will also deploy dial-peers in the opposite direction to accept calls from the ISP and pass them to the CUCM. Matching from a CUBE perspective is done via the URI headers.

Assumptions - Hardware

This code was tested on CSR and ISR4331. It should function on any Cisco device that supports CUBE as it is using it's standard commands

Enviornment Variables

input_ip_addr = Hostname (must be resolvable) or IP address of CSR/ISR
input_username = User with priviledge to make configuration changes
input_password = Password for User
internal_cube_interface = internal interface for binding signalling/media. This is the interface that the CUCM SIP trunk will send traffic towards and also recieve
external_cube_interface = external interface for binding signalling/media. This is the interface that the ITSP SIP trunk will send traffic towards and also recieve
verify_ip_change = Prompt to register response to external/internal IP address changes requirements
external_ip = Verify the IP of the external interface
internal_ip = Verifying the IP of the external interface
subnet_mask = Used in methods to verify subnet during IP address change on internal/external interface
cucm_signalling_protocol = CUCM signalling protocol for SIP trunks (tcp/udp)
number_of_cucm = Number of CUCM in the cluster
cucm_ip = IP/Hostnames of each CUCM held in a list
itsp_signalling_ip = ITSP signaling IP
itsp_media_ip = ITSP media IP
itsp_route_ip = Route to use to forward traffic to ITSP SIP gateway
itsp_signalling_protocol = ITSP signalling protocol(udp/tcp)
dial_peer_range = Dial-Peer numbering . It will ask for a single number and create 4 dialpeer 1 after another a.g 1000 will result in 1000,1001,1002 and 1003   being created

Successful Output

When the program has ran, on success the output will look like the below. The program will produce this for you. I've excluded any irrelevant configuration and my comments on the code can be seen with ()

voice service voip
ip address trusted list
ipv4 1.1.1.2 255.255.255.255
ipv4 1.1.1.3 255.255.255.255
ipv4 2.2.2.3 255.255.255.255    (IPs of CUCM/ITSP for signalling any IPs not listed will get rejected)
address-hiding
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
supplementary-service media-renegotiate
redirect ip2ip
fax protocol pass-through g711ulaw
sip
session refresh
asserted-id pai
early-offer forced
midcall-signaling passthru
pass-thru subscribe-notify-events all
pass-thru content unsupp
sip-profiles inbound
!
!
voice class uri FromCUCM sip (IPs/DNS of all CUCM nodes inputted during run time)
host ipv4:1.1.1.2
host ipv4:1.1.1.3
!
voice class uri FromITSP sip (ITSP IP address)
host ipv4:2.2.2.3
voice class codec 1
 codec preference 1 g711alaw
codec preference 2 g711ulaw
!
!
!
voice class dpg 1000
dial-peer 1001
!
voice class dpg 1002
dial-peer 1003
!
voice class server-group 1001
ipv4 2.2.2.3 preference 1
!
voice class server-group 1003
ipv4 1.1.1.2 preference 1
ipv4 1.1.1.3 preference 2

interface GigabitEthernet2
ip address 1.1.1.1 255.255.255.0
negotiation auto
no mop enabled
no mop sysid
!
interface GigabitEthernet3
ip address 2.2.2.2 255.255.255.0
negotiation auto
no mop enabled
no mop sysid
!

ip route 2.2.2.3 255.255.255.255 GigabitEthernet3 2.2.2.3  (Static routes to providers IPs via the gateway provided during runtime)
ip route 2.2.2.4 255.255.255.255 GigabitEthernet3 2.2.2.3
dial-peer voice 1000 voip
description Incoming from CUCM
session protocol sipv2
session transport tcp
destination dpg 1000
incoming uri via FromCUCM
voice-class codec 1  
voice-class sip options-keepalive
voice-class sip bind control source-interface GigabitEthernet2
voice-class sip bind media source-interface GigabitEthernet2
dtmf-relay rtp-nte
fax protocol pass-through g711ulaw
no vad
!
dial-peer voice 1001 voip
description Outbound to ITSP
destination-pattern 99999999$
session protocol sipv2
session transport udp
session server-group 1001
voice-class codec 1  
voice-class sip options-keepalive
voice-class sip bind control source-interface GigabitEthernet3
voice-class sip bind media source-interface GigabitEthernet3
dtmf-relay rtp-nte
fax protocol pass-through g711ulaw
no vad
!
dial-peer voice 1002 voip
description Incoming from ITSP
session protocol sipv2
session transport udp
destination dpg 1002
incoming uri via FromITSP
voice-class codec 1  
voice-class sip options-keepalive
voice-class sip bind control source-interface GigabitEthernet3
voice-class sip bind media source-interface GigabitEthernet3
dtmf-relay rtp-nte
fax protocol pass-through g711ulaw
no vad

dial-peer voice 1003 voip
description Outbound to ITSP
destination-pattern 99999999$
session protocol sipv2
session transport tcp
session server-group 1003
voice-class codec 1  
voice-class sip options-keepalive
voice-class sip bind control source-interface GigabitEthernet2
voice-class sip bind media source-interface GigabitEthernet2
dtmf-relay rtp-nte
fax protocol pass-through g711ulaw
no vad