Access the documentation to integrate VerbumCall with Cisco CUCM/CUBE platform.
Follow the next steps to integrate VerbumCall with Cisco CUCM/CUBE :
1. Introduction
This document is intended for Nexmo SIP trunk customer’s technical staff and Value Added Retailer (VAR) having installation and operational responsibilities. This configuration guide provides steps for configuring Cisco Unified Communications Manager (Cisco UCM) 11.5.1.12900-21 and Cisco Unified Boarder Element (Cisco UBE) 16.05.01b to Nexmo SIP Trunking services.
2. SIP Trunking Network Components
The network for the SIP trunk reference configuration is illustrated below and is representative of a Cisco UCM and Cisco UBE configuration to Nexmo SIP trunking.
2.1 Hardware Components
Cisco UCS-C240-M3S VMWare host running ESXi 5.5.0 Standard
Cisco ISR4321/K9 router as CUBE
Cisco ISR4321/K9 (1RU) processor with 1684579K/6147K bytes of memory with 3 Gigabit Ethernet interfaces
Incoming and outgoing off-net calls using G711ULAW & G711ALAW voice codecs
Calling Line (number) Identification Presentation
Calling Line (number) Identification Restriction
Call hold and resume
Call transfer (unattended and attended )
Call Conference
Call forward (all, no answer)
DTMF relay both directions (RFC2833)
Media flow-through on Cisco UBE
3.1.2 Features Not Supported by PBX
None
3.1.3 Caveats and Limitations
Caller ID is not updated after attended or semi-attended transfers to off-net phones. This is due to a limitation on Cisco UBE. The issue does not impact the calls.
4. Configuration.
4.1 IP Address Worksheet
The specific values listed in the table below and in subsequent sections are used in the lab configuration described in this document, and are for illustrative purposes only. The customer must obtain and use the values for your deployment.
4.2.2 Cisco Unified Call Manager Service Parameters.
Navigation: System → Service Parameters
Select Server : clus21pub--CUCM Voice/Video (Active)
Select Service : Cisco Call Manager (Active)
All other fields are set to default values
4.2.3 Off-Net Calls via Nexmo SIP Trunk
Off-net calls are served by SIP trunks configured between Cisco UCM and Nexmo Network and calls are routed via Cisco UBE. From Cisco UBE, we have pointed the trunk to sip.nexmo.com and opened the firewall for the list of IP addresses in the portal provided by Nexmo.
4.2.3.1 SIP Trunk Security Profile
Navigation: System → Security → SIP Trunk Security Profile
Set Name : Non Secure SIP Trunk Profile is used as an example
Set Outgoing Transport Type : UDP in this example
SIP trunks to Nexmo should use UDP as a transport protocol for SIP. This is configured using SIP Trunk Security profile, which is later assigned to the SIP trunk itself.
Cisco IP phone dial "8"+11 digit number to access PSTN via Cisco UBE. "8" is removed before sending to Cisco UBE.
4.3 Configuring Cisco Unified Border Element
4.3.1 Network Interface
Configure Ethernet IP address and sub interface. The IP address and VLAN encapsulation used are for illustration only, the actual IP address can vary. For SIP trunks two IP addresses must be configured—LAN and WAN.
interface GigabitEthernet0/0/ ip address 192.65.79.XXX 255.255.255. negotiation auto interface GigabitEthernet0/0/ ip address 10.80.11.15 255.255.255. negotiation auto
4.3.2 Global Cisco UBE Settings
In order to enable Cisco UBE IP2IP gateway functionality, enter the following:
voice service voip ip address trusted list ipv4 173.193.199. ipv4 174.37.245. ipv4 5.10.112. ipv4 5.10.112. ipv4 119.81.44. ipv4 119.81.44.
Cisco UBE uses dial-peer to route the call based on the digit to route the call accordingly.
dial-peer voice 1 voip description incoming dial-peer from CUCM to CUBE session protocol sipv session transport udp incoming called-number .T voice-class codec 1 dtmf-relay rtp-nte no vad dial-peer voice 2 voip description outgoing dial-peer from CUBE to CUCM destination-pattern 120 session protocol sipv session target ipv4:10.80.11. session transport udp voice-class codec 1 voice-class sip options-keepalive
dtmf-relay rtp-nte no vad dial-peer voice 3 voip description incoming dial-peer from NEXMO to CUBE session protocol sipv session transport udp incoming called-number 120 voice-class codec 1 dtmf-relay rtp-nte no vad dial-peer voice 4 voip description outgoing dial-peer from CUBE to NEXMO destination-pattern .T session protocol sipv session target sip-server session transport udp voice-class codec 1 voice-class sip asserted-id pai voice-class sip options-keepalive dtmf-relay rtp-nte no vad
4.3.5 Configuration Example
User Access Verification
Username: cisco Password: nexmo# nexmo#sh run Building configuration. Current configuration : 5992 bytes version 16. service timestamps debug datetime msec service timestamps log datetime msec service password-encryption platform qfp utilization monitor load 80 no platform punt-keepalive disable-kernel-core hostname nexmo boot-start-marker
boot system flash isr4300-universalk9.16.05.01b.SPA.bin boot-end-marker vrf definition Mgmt-intf
voice service voip ip address trusted list ipv4 173.193.199. ipv4 174.37.245. ipv4 5.10.112. ipv4 5.10.112. ipv4 119.81.44. ipv4 119.81.44. address-hiding mode border-element license capacity 20 allow-connections sip to sip sip session refresh asserted-id pai early-offer forced midcall-signaling passthru g729 annexb-all voice class codec 1 codec preference 1 g711ulaw codec preference 2 g711alaw license udi pid ISR4321/K9 sn FDO19220MQ license boot suite AdvUCSuiteK license boot level uck diagnostic bootup level minimal spanning-tree extend system-id username cisco privilege 15 password 7 redundancy mode none interface GigabitEthernet0/0/ ip address 192.65.79.160 255.255.255. negotiation auto interface GigabitEthernet0/0/
ip address 10.80.11.15 255.255.255. negotiation auto interface GigabitEthernet0/1/ no ip address negotiation auto interface GigabitEthernet vrf forwarding Mgmt-intf no ip address negotiation auto threat-visibility ip forward-protocol nd ip http server ip http authentication local ip http secure-server ip route 0.0.0.0 0.0.0.0 192.65.79. ip route 10.64.0.0 255.255.0.0 10.80.11. ip route 172.16.24.0 255.255.248.0 10.80.11. ip ssh server algorithm encryption aes128-ctr aes192-ctr aes256-ctr ip ssh client algorithm encryption aes128-ctr aes192-ctr aes256-ctr control-plane mgcp behavior rsip-range tgcp-only mgcp behavior comedia-role none mgcp behavior comedia-check-media-src disable mgcp behavior comedia-sdp-force disable mgcp profile default dial-peer voice 4 voip description outgoing dial-peer from CUBE to NEXMO destination-pattern .T session protocol sipv session target sip-server session transport udp voice-class codec 1 voice-class sip asserted-id pai voice-class sip options-keepalive dtmf-relay rtp-nte no vad dial-peer voice 1 voip description incoming dial-peer from CUCM to CUBE session protocol sipv
session transport udp incoming called-number .T voice-class codec 1 dtmf-relay rtp-nte no vad dial-peer voice 2 voip description outgoing dial-peer from CUBE to CUCM destination-pattern 120 session protocol sipv session target ipv4:10.80.11. session transport udp voice-class codec 1 voice-class sip options-keepalive dtmf-relay rtp-nte no vad dial-peer voice 3 voip description incoming dial-peer from NEXMO to CUBE session protocol sipv session transport udp incoming called-number 120 voice-class codec 1 dtmf-relay rtp-nte no vad sip-ua credentials number 12014647035 username 911236e3 password 7 realm sip.nexmo.com authentication username 911236e3 password 7 sip-server dns:sip.nexmo.com: line con 0 transport input none stopbits 1 line aux 0 stopbits 1 line vty 0 4 login local no network-clock synchronization automatic
4.4 Configure Numbers in Nexmo Account.
Login to the Nexmo account using the credentials provided at the time of registration. A Key and Secret will be displayed on the dashboard and this can be used as the username and password for Registration SIP Trunks.
In order to provide the URL to which the call has to be routed from Nexmo, navigate to the Numbers tab
Click Edit against each number as shown below
A pop-up will be displayed
Select the "Forward to" and provide the URL to which the calls route
