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Page 1: Cisco UCM 11.0.1 to CenturyLink SIP Trunk via Cisco ... · voice-class codec 2 voice-class sip bind control source-interface GigabitEthernet0/0/1 voice-class sip bind media source-interface

© 2016 Cisco Systems, Inc. All rights reserved. Important notices, privacy statements, and trademarks of Cisco Systems, Inc. can be found on cisco.com

Page 1 of 55

Cisco UCM 11.0.1 to CenturyLink SIP Trunk via Cisco

Unified Border Element 11.1 [v15.5.3.S2, ASR1K/XE-

3.16.02.S]

Mar 17, 2016

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Page 2 of 55

Table of Contents

Introduction .................................................................................................................................................. 3

Network Topology ......................................................................................................................................... 4

System Components ..................................................................................................................................... 5

Hardware Requirements ........................................................................................................................... 5

Software Requirements ............................................................................................................................ 6

Features ........................................................................................................................................................ 6

SIP Registration Method ........................................................................................................................... 6

Features Supported .................................................................................................................................. 6

Features Not Supported............................................................................................................................ 7

Caveats ...................................................................................................................................................... 7

Configuration ................................................................................................................................................ 7

Configuring Cisco Unified Border Element ............................................................................................... 7

Network Interface ................................................................................................................................. 8

Global Cisco UBE settings .................................................................................................................... 10

Codecs ................................................................................................................................................. 11

SIP Profiles ........................................................................................................................................... 11

Dial peer .............................................................................................................................................. 12

SIP-ua Registration Method .................................................................................................................... 14

Configuration example ........................................................................................................................ 15

Cisco UCM Configuration ........................................................................................................................ 36

Cisco UCM Version .............................................................................................................................. 36

Cisco UCM Device Pool Configuration ................................................................................................ 37

SIP Profile Configuration used by SIP trunk to Cisco UBE ................................................................... 41

SIP Trunk Security Profile Configuration ............................................................................................. 44

SIP trunk to PSTN (Via Cisco UBE) Configuration ................................................................................ 46

Route Pattern Configuration ............................................................................................................... 50

Acronyms .................................................................................................................................................... 53

Important Information ................................................................................................................................ 54

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Page 3 of 55

Table of Figures

Figure 1 Network Topology ........................................................................................................................... 5

Figure 2 High Availability topology ............................................................................................................... 8

Figure 3 Cisco UCM Version ........................................................................................................................ 36

Figure 4 Audio codec preference ................................................................................................................ 37

Figure 5 Region Configuration .................................................................................................................... 38

Figure 6 Device Pool configuration(cont.) .................................................................................................. 39

Figure 7 Device Pool Configuration (cont.) ................................................................................................. 40

Figure 8 Device Pool Configuration ............................................................................................................. 41

Figure 9 SIP Profile (cont.) ........................................................................................................................... 42

Figure 10 SIP Profile (cont.) ......................................................................................................................... 43

Figure 11 SIP Profile .................................................................................................................................... 44

Figure 12 SIP Trunk Security Profile ............................................................................................................ 45

Figure 13 SIP Trunk (cont.) .......................................................................................................................... 46

Figure 14 SIP Trunk (cont.) .......................................................................................................................... 47

Figure 15 SIP Trunk ..................................................................................................................................... 48

Figure 16 Route Pattern .............................................................................................................................. 50

Figure 17 Route Pattern (cont.) .................................................................................................................. 51

Figure 18 Route Pattern ............................................................................................................................. 52

Introduction

Service Providers today, such as CenturyLink, are offering alternative methods to connect to the PSTN

via their IP network. Most of these services utilize SIP as the primary signaling method and centralized

IP to TDM POP gateways to provide on-net and off-net services.

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Page 4 of 55

CenturyLink is a service provider offering that allows connection to the PSTN and may offer the end

customer a viable alternative to traditional PSTN connectivity. A demarcation device between these

services and customer owned services is recommended. As an intermediary device between Cisco UCM

11.0.1 and CenturyLink, Cisco Unified Border Element (Cisco UBE) 15.5(3) s2 can be used. The Cisco

Unified Border Element (Cisco UBE) 15.5(3) s2 provides demarcation, security and inter-working and

session control services for Cisco UCM connected to CenturyLink network.

This document assumes the reader is knowledgeable with the terminology and configuration of Cisco

UCM 11.0.1 and CUBE. Only configuration settings specifically required for CenturyLink interoperability

are presented. Feature configuration and most importantly the dial plan are customer specific and need

individual approach.

This application note describes how to configure a Cisco UCM and Cisco Unified Border Element (Cisco UBE) 15.5(3) s2 for connectivity to CenturyLink Trunking service. The deployment model covered in this application note is Cisco UCM11.0.1 to PSTN via Cisco Unified Border Element (Cisco UBE) 15.5(3) s2.

Testing was performed in accordance to Cisco generic SIP Trunking test methodology and among features verified were – basic calls, DTMF transport, Music on Hold (MOH), unattended and attended transfers, call forward, conferences and High Availability.

The Cisco Unified Border Element (Cisco UBE) configuration detailed in this document is based on a lab environment with a simple dial-plan used to ensure proper interoperability between CenturyLink SIP network and Cisco UCM. The configuration described in this document details the important configuration settings to have enabled for interoperability to be successful and care must be taken by the network administrator deploying Cisco UCM to interoperate to CenturyLink SIP Trunking network.

Network Topology

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Page 5 of 55

Figure 1 Network Topology

The lab network consists of the following components:

Cisco UCM IP-PBX for voice features and SIP trunk termination.

Digital phones and FAX machine connected to the Cisco UCM.

Cisco UBE serves as the demarcation point between customer’s LAN network and CenturyLink’s WAN network.

System Components

Hardware Requirements

UCS-C240 VMWare server running ESXi 5.5 used for CUCM.

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Page 6 of 55

Cisco IP Phones. This solution was tested with Cisco 9971, Cisco 8961 Phones.

Cisco ASR - Cisco ASR1001 (1RU) Processor (revision 1RU) with 1067053K/6147K bytes of memory.

Processor board ID SSI17360FV1.

Software Requirements

Cisco Unified Communication Manager (CUCM): System version: 11.0.1.10000-10

Cisco IOS XE Software, Version 03.16.02.S

Cisco IOS Software, ASR1000 Software (X86_64_LINUX_IOSD-UNIVERSALK9-M), Version 15.5(3) S2, RELEASE SOFTWARE (fc2)

Venta Fax client version 7.4.237.590

Features

SIP Registration Method

CenturyLink Network requires SIP REGISTER support to allow the IP-PBX to originate calls from the IP-

PBX and to send calls to the PBX from the PSTN. Cisco UBE supports SIP Register with authentication.

CenturyLink implementation team provides the Pilot number and the authentication key, which should

be provisioned in the Cisco UBE.

Features Supported These Features are supported by activating and deactivating the features via the Cisco UCM user phone

or Cisco UCM Administration.

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Page 7 of 55

Basic calls using G.729 and G.711ulaw

Call Hold and Resume

Call Transfer

Call Forwarding

Call Waiting

Three-Way Calling

Anonymous call

Fax T.38 and G711 Pass-through

DTMF- RFC2833

Calling Party Number Presentation

Calling Party Number Restriction

Features Not Supported

Cisco IP phones used in this test do not support blind transfer. Early attended and attended

transfer scenarios were tested

CenturyLink do not support international calling, and therefore those test cases are marked as not supported

In HA redundancy mode the primary cube when comes up after a reboot/network outage will not take over the primary/active role.

Caveats

Cisco IOS Enhanced Conference Bridge is used to establish conference with PSTN using G729

codec

CLID is not updated on PSTN phones for transfer scenarios

Testing is done with only one PBX

Configuration

Configuring Cisco Unified Border Element

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Page 8 of 55

Network Interface The IP address used are for illustration only, the actual IP address can vary. The Active/Standby pair

share the same virtual IP address and continually exchange status messages.

Figure 2 High Availability topology

CUBE 1:

interface GigabitEthernet0/0/0

description WAN interface

ip address 192.65.79.180 255.255.255.128

negotiation auto

redundancy rii 1

redundancy group 1 ip 192.65.79.155 exclusive

!

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interface GigabitEthernet0/0/1

description LAN interface

ip address 10.80.11.10 255.255.255.0

negotiation auto

redundancy rii 2

redundancy group 1 ip 10.80.11.20 exclusive

!

interface GigabitEthernet0/0/2

ip address 10.64.4.19 255.255.255.0

negotiation auto

CUBE 2:

!

interface GigabitEthernet0/0/0

description WAN interface

ip address 192.65.79.181 255.255.255.128

negotiation auto

redundancy rii 1

redundancy group 1 ip 192.65.79.155 exclusive

!

interface GigabitEthernet0/0/1

description LAN interface

ip address 10.80.11.15 255.255.255.0

negotiation auto

redundancy rii 2

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Page 10 of 55

redundancy group 1 ip 10.80.11.20 exclusive

!

interface GigabitEthernet0/0/2

ip address 10.64.4.20 255.255.255.0

negotiation auto

!

Global Cisco UBE settings In order to enable Cisco UBE IP2IP SBC functionality, following command has to be entered:

voice service voip

no ip address trusted authenticate

rtcp keepalive

address-hiding

mode border-element license capacity 20

allow-connections sip to sip

redundancy-group 1

no supplementary-service sip moved-temporarily

no supplementary-service sip refer

supplementary-service media-renegotiate

sip

bind control source-interface GigabitEthernet0/0/2

bind media source-interface GigabitEthernet0/0/2

session refresh

asserted-id pai

localhost dns:voip.centurylink.com

early-offer forced

header-passing

midcall-signaling passthru

conn-reuse

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Page 11 of 55

Explanation

Command Description

allow-connections sip to sip Allow IP2IP connections between two SIP call legs

redundancy-group 1 Enable High Availability for the VoIP service

rtcp keepalive Enables the CUBE to send rtcp keepalive packets for the session keepalive

Bind control and media SIP Global binding is done with the interface pointing to CenturyLink

Codecs

voice class codec 2

codec preference 1 g729r8

codec preference 2 g711ulaw

SIP Profiles

voice class sip-profiles 101

!==================================================

! This command modifies P-Asserted-Identity header information to Pilot number. “4695737016” is used

!for this testing.

!==================================================

request INVITE sip-header P-Asserted-Identity modify "(<.*:)(.*@)" "\4695737016@"

!==================================================

! This command modifies Diversion header information to Pilot number. “4695737016” is used for this

!testing

!==================================================

request INVITE sip-header Diversion modify "(<.*:)(.*@)" "\4695737016@"

!==================================================

! This command removes unwanted SIP header Cisco-Guid, User-Agent to reduce the packet size of the

INVITE !message

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!==================================================

request ANY sip-header Cisco-Guid remove

response ANY sip-header Cisco-Guid remove

request ANY sip-header User-Agent remove

response ANY sip-header User-Agent remove

Dial peer

Dial-peer to CUCM:

!

dial-peer voice 201 voip

description " Incoming PSTN to IP-PBX - IP-PBX facing side "

destination-pattern 469.......

session protocol sipv2

session target ipv4:10.80.11.2

session transport tcp

voice-class codec 2

voice-class sip options-keepalive

voice-class sip bind control source-interface GigabitEthernet0/0/1

voice-class sip bind media source-interface GigabitEthernet0/0/1

dtmf-relay rtp-nte

no vad

!

dial-peer voice 102 voip

description incoming from IP-PBX

session protocol sipv2

session transport tcp

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incoming called-number 921424259..

voice-class codec 2

voice-class sip bind control source-interface GigabitEthernet0/0/1

voice-class sip bind media source-interface GigabitEthernet0/0/1

dtmf-relay rtp-nte

no vad

Dial-peer to CenturyLink:

dial-peer voice 100 voip

description "Outgoing To PSTN"

destination-pattern .T

session protocol sipv2

session target ipv4:216.206.66.81:5100

session transport udp

voice-class codec 2

voice-class sip privacy-policy passthru

voice-class sip profiles 101

voice-class sip bind control source-interface GigabitEthernet0/0/0

voice-class sip bind media source-interface GigabitEthernet0/0/0

dtmf-relay rtp-nte

no vad

!

dial-peer voice 101 voip

description incoming from PSTN

session protocol sipv2

session transport udp

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incoming called-number 469.......

voice-class codec 2

voice-class sip bind control source-interface GigabitEthernet0/0/0

voice-class sip bind media source-interface GigabitEthernet0/0/0

dtmf-relay rtp-nte

no vad

SIP-ua Registration Method

CenturyLink Network requires SIP REGISTER support to allow the IP-PBX to originate calls from the IP-

PBX and to send calls to the PBX from the PSTN. Cisco UBE supports SIP Register with authentication.

CenturyLink implementation team provides the Pilot number and the authentication key, which should

be provisioned in the Cisco UBE.

sip-ua

!===================================================

! This Command Enters SIP digest credentials in sip-ua configuration mode for an Cisco UBE to register

!with registrars

!===================================================

credentials number <pilot number> username <auth-name> password <password> realm

voip.centurylink.com

!===================================================

! This Enables SIP digest authentication globally

!===================================================

authentication username <auth-name> password <password> realm voip.centurylink.com

no remote-party-id

registrar dns:voip.centurylink.com expires 60

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Page 15 of 55

Configuration example The following configuration snippet contains a sample configuration of Cisco UBE with all parameters

mentioned previously.

Active Cisco UBE: sh run

Building configuration...

Current configuration: 5936 bytes

!

! Last configuration change at 07:29:41 UTC Thu Mar 17 2016 by cisco

!

version 15.5

service timestamps debug datetime msec

service timestamps log datetime msec

no platform punt-keepalive disable-kernel-core

!

hostname centurylink1

!

boot-start-marker

boot system flash boot flash: asr1001-universalk9.03.16.02.S.155-3.S2-ext.bin

boot-end-marker

!

!

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vrf definition Mgmt-intf

!

address-family ipv4

exit-address-family

!

address-family ipv6

exit-address-family

!

logging queue-limit 1000000000

logging buffered 10000000

logging rate-limit 10000

no logging console

no logging monitor

enable secret 4 Pe0NhiWw5IXZpE.k5VhTSCoGPcuVeRyrer9kEPz20Z6

!

no aaa new-model

!

!

!

!

!

!

!

!

!

!

!

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ip host voip.centurylink.com 216.206.66.81

no ip domain lookup

ip domain name voip.centurylink.com

!

!

!

!

!

!

!

!

!

!

subscriber templating

!

multilink bundle-name authenticated!

!

!

!

voice service voip

no ip address trusted authenticate

rtcp keepalive

address-hiding

mode border-element license capacity 20

allow-connections sip to sip

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redundancy-group 1

no supplementary-service sip moved-temporarily

no supplementary-service sip refer

sip

bind control source-interface GigabitEthernet0/0/0

bind media source-interface GigabitEthernet0/0/0

session refresh

header-passing

asserted-id pai

localhost dns:voip.centurylink.com

early-offer forced

midcall-signaling passthru

g729 annexb-all

!

!

voice class sip-profiles 101

request INVITE sip-header P-Asserted-Identity modify "(<.*:)(.*@)" "\4695737016@"

request INVITE sip-header Diversion modify "(<.*:)(.*@)" "\4695737016@"

request ANY sip-header Cisco-Guid remove

response ANY sip-header Cisco-Guid remove

request ANY sip-header User-Agent remove

response ANY sip-header User-Agent remove

!

license udi pid ASR1001 sn JAE17430GQ5

license accept end user agreement

license boot level advipservices

!

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spanning-tree extend system-id

!

username cisco privilege 15 password 0 cisco

!

redundancy

mode none

application redundancy

group 1

name voice-b2bha

priority 100 failover threshold 75

timers delay 30 reload 60

control GigabitEthernet0/0/2 protocol 1

data GigabitEthernet0/0/2

track 1 shutdown

track 2 shutdown

!

!

!

!

!

vlan internal allocation policy ascending

!

track 1 interface GigabitEthernet0/0/0 line-protocol

!

track 2 interface GigabitEthernet0/0/1 line-protocol

!

!

!

!

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!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

interface GigabitEthernet0/0/0

description WAN interface

ip address 192.65.79.180 255.255.255.128

negotiation auto

redundancy rii 1

redundancy group 1 ip 192.65.79.155 exclusive

!

!

!

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interface GigabitEthernet0/0/1

description LAN interface

ip address 10.80.11.10 255.255.255.0

negotiation auto

redundancy rii 2

redundancy group 1 ip 10.80.11.20 exclusive

!

interface GigabitEthernet0/0/2

ip address 10.64.4.19 255.255.255.0

negotiation auto

interface GigabitEthernet0

vrf forwarding Mgmt-intf

no ip address

shutdown

negotiation auto

!

!

ip forward-protocol nd

no ip http server

no ip http secure-server

ip route 0.0.0.0 0.0.0.0 192.65.79.129

ip route 10.64.0.0 255.255.0.0 10.80.11.1

ip route 10.80.0.0 255.255.0.0 10.80.11.1

!

!

!

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!

!

!

!

!

control-plane

!

dial-peer voice 201 voip

description " Incoming PSTN to IP-PBX - IP-PBX facing side "

destination-pattern 469.......

session protocol sipv2

session target ipv4:10.80.11.2

session transport tcp

voice-class codec 2

voice-class sip options-keepalive

voice-class sip bind control source-interface GigabitEthernet0/0/1

voice-class sip bind media source-interface GigabitEthernet0/0/1

dtmf-relay rtp-nte

no vad

!

dial-peer voice 102 voip

description incoming from IP-PBX

session protocol sipv2

session transport tcp

incoming called-number 921424259..

voice-class codec 2

voice-class sip bind control source-interface GigabitEthernet0/0/1

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voice-class sip bind media source-interface GigabitEthernet0/0/1

dtmf-relay rtp-nte

no vad

dial-peer voice 100 voip

description "Outgoing To PSTN"

destination-pattern .T

session protocol sipv2

session target ipv4:216.206.66.81:5100

session transport udp

voice-class codec 2

voice-class sip privacy-policy passthru

voice-class sip profiles 101

voice-class sip bind control source-interface GigabitEthernet0/0/0

voice-class sip bind media source-interface GigabitEthernet0/0/0

dtmf-relay rtp-nte

no vad

!

dial-peer voice 101 voip

description incoming from PSTN

session protocol sipv2

session transport udp

incoming called-number 469.......

voice-class codec 2

voice-class sip bind control source-interface GigabitEthernet0/0/0

voice-class sip bind media source-interface GigabitEthernet0/0/0

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Page 24 of 55

dtmf-relay rtp-nte

no vad

!

gateway

media-inactivity-criteria all

timer receive-rtcp 5

timer receive-rtp 86400

!

sip-ua

no remote-party-id

credentials number <Pilot number> username <authentication id>password <Password> realm <Realm

id>

authentication username <authentication-id> password <password> realm <Realm id>

retry invite 5

timers expires 1800000

registrar <registrar IP>

connection-reuse

!

!

line con 0

stopbits 1

line aux 0

stopbits 1

line vty 0 4

exec-timeout 0 0

login local

!

!

end

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Standby Cisco UBE:

sh run

Building configuration...

Current configuration: 5936 bytes

!

! Last configuration change at 07:29:41 UTC Thu Mar 17 2016 by cisco

!

version 15.5

service timestamps debug datetime msec

service timestamps log datetime msec

no platform punt-keepalive disable-kernel-core

!

hostname centurylink1

!

boot-start-marker

boot system flash boot flash: asr1001-universalk9.03.16.02.S.155-3.S2-ext.bin

boot-end-marker

!

!

vrf definition Mgmt-intf

!

address-family ipv4

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exit-address-family

!

address-family ipv6

exit-address-family

!

logging queue-limit 1000000000

logging buffered 10000000

logging rate-limit 10000

no logging console

no logging monitor

enable secret 4 Pe0NhiWw5IXZpE.k5VhTSCoGPcuVeRyrer9kEPz20Z6

!

no aaa new-model

!

!

!

!

!

!

!

!

!

!

!

ip host voip.centurylink.com 216.206.66.81

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no ip domain lookup

ip domain name voip.centurylink.com

!

!

!

!

!

!

!

!

!

!

subscriber templating

!

multilink bundle-name authenticated!

!

!

!

voice service voip

no ip address trusted authenticate

rtcp keepalive

address-hiding

mode border-element license capacity 20

allow-connections sip to sip

redundancy-group 1

no supplementary-service sip moved-temporarily

no supplementary-service sip refer

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sip

bind control source-interface GigabitEthernet0/0/0

bind media source-interface GigabitEthernet0/0/0

session refresh

header-passing

asserted-id pai

localhost dns:voip.centurylink.com

early-offer forced

midcall-signaling passthru

g729 annexb-all

!

!

voice class sip-profiles 101

request INVITE sip-header P-Asserted-Identity modify "(<.*:)(.*@)" "\4695737016@"

request INVITE sip-header Diversion modify "(<.*:)(.*@)" "\4695737016@"

request ANY sip-header Cisco-Guid remove

response ANY sip-header Cisco-Guid remove

request ANY sip-header User-Agent remove

response ANY sip-header User-Agent remove

!

license udi pid ASR1001 sn JAE17430GQ5

license accept end user agreement

license boot level advipservices

!

spanning-tree extend system-id

!

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username cisco privilege 15 password 0 cisco

!

redundancy

mode none

application redundancy

group 1

name voice-b2bha

priority 100 failover threshold 75

timers delay 30 reload 60

control GigabitEthernet0/0/2 protocol 1

data GigabitEthernet0/0/2

track 1 shutdown

track 2 shutdown

!

!

!

!

!

vlan internal allocation policy ascending

!

track 1 interface GigabitEthernet0/0/0 line-protocol

!

track 2 interface GigabitEthernet0/0/1 line-protocol

!

!

!

!

!

!

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!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

interface GigabitEthernet0/0/0

description WAN interface

ip address 192.65.79.181 255.255.255.128

negotiation auto

redundancy rii 1

redundancy group 1 ip 192.65.79.155 exclusive

!

!

!

interface GigabitEthernet0/0/1

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description LAN interface

ip address 10.80.11.15 255.255.255.0

negotiation auto

redundancy rii 2

redundancy group 1 ip 10.80.11.20 exclusive

!

interface GigabitEthernet0/0/2

ip address 10.64.4.20 255.255.255.0

negotiation auto

interface GigabitEthernet0

vrf forwarding Mgmt-intf

no ip address

shutdown

negotiation auto

!

!

ip forward-protocol nd

no ip http server

no ip http secure-server

ip route 0.0.0.0 0.0.0.0 192.65.79.129

ip route 10.64.0.0 255.255.0.0 10.80.11.1

ip route 10.80.0.0 255.255.0.0 10.80.11.1

!

!

!

!

!

!

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!

!

control-plane

!

dial-peer voice 201 voip

description " Incoming PSTN to IP-PBX - IP-PBX facing side "

destination-pattern 469.......

session protocol sipv2

session target ipv4:10.80.11.2

session transport tcp

voice-class codec 2

voice-class sip options-keepalive

voice-class sip bind control source-interface GigabitEthernet0/0/1

voice-class sip bind media source-interface GigabitEthernet0/0/1

dtmf-relay rtp-nte

no vad

!

dial-peer voice 102 voip

description incoming from IP-PBX

session protocol sipv2

session transport tcp

incoming called-number 921424259..

voice-class codec 2

voice-class sip bind control source-interface GigabitEthernet0/0/1

voice-class sip bind media source-interface GigabitEthernet0/0/1

dtmf-relay rtp-nte

no vad

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Page 33 of 55

dial-peer voice 100 voip

description "Outgoing To PSTN"

destination-pattern .T

session protocol sipv2

session target ipv4:216.206.66.81:5100

session transport udp

voice-class codec 2

voice-class sip privacy-policy passthru

voice-class sip profiles 101

voice-class sip bind control source-interface GigabitEthernet0/0/0

voice-class sip bind media source-interface GigabitEthernet0/0/0

dtmf-relay rtp-nte

no vad

!

dial-peer voice 101 voip

description incoming from PSTN

session protocol sipv2

session transport udp

incoming called-number 469.......

voice-class codec 2

voice-class sip bind control source-interface GigabitEthernet0/0/0

voice-class sip bind media source-interface GigabitEthernet0/0/0

dtmf-relay rtp-nte

no vad

!

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Page 34 of 55

gateway

media-inactivity-criteria all

timer receive-rtcp 5

timer receive-rtp 86400

!

sip-ua

no remote-party-id

credentials number <Pilot number> username <authentication id>password <Password> realm <Realm

id>

authentication username <authentication-id> password <password> realm <Realm id>

retry invite 5

timers expires 1800000

registrar <registrar IP>

connection-reuse

!

!

line con 0

stopbits 1

line aux 0

stopbits 1

line vty 0 4

exec-timeout 0 0

login local

!

!

End

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Page 35 of 55

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Page 36 of 55

Cisco UCM Configuration

The configuration screen shots show general over view of lab configuration for this interoperability testing.

Cisco UCM Version

Figure 3 Cisco UCM Version

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Page 37 of 55

Cisco UCM Device Pool Configuration

Navigation Path: System Region Information Audio Codec Preference List

Figure 4 Audio codec preference

Cisco UCM Region Configuration (Continued…)

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Page 38 of 55

Navigation Path: System Region

Figure 5 Region Configuration

Cisco UCM Device pool Configuration (Continued…)

Navigation Path: System Device Pool.

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“G729 Pool” & “Default” - Device Pool is configured for testing the interoperability. No special consideration needs to be taken when configuring the Device Pools. Optionally, a Media Resource Group List can be added to the Device Pools, if needed, to assign selected Media Resources (Conference Bridges, Transcoders, MOH servers, Annunciators) to devices

Figure 6 Device Pool configuration(cont.)

Device Pool Configuration (Continued…)

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Figure 7 Device Pool Configuration (cont.)

Device Pool Configuration (Continued…)

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Page 41 of 55

Figure 8 Device Pool Configuration

SIP Profile Configuration used by SIP trunk to Cisco UBE

Navigation: Device Device Settings SIP Profile Set SIP profile Name * = Century SIP Profile. This is used for this example Set SIP Rel1xx Options* = Send PRACK if 1xx contains SDP Note*= Some PSTN network call prompters utilize early-media cut-through to offer menu options to the caller (DTMF select menu) before the call is connected. In order for Cisco UCM/Cisco UBE solution to achieve successful early-media cut-through, the Cisco UCM to Cisco UBE call leg must be enabled with SIP PRACK. To enable SIP PRACK on the Cisco UCM, the SIP Profile “SIP Rel1XX Options” setting must be set to “Send PRACK”.

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Figure 9 SIP Profile (cont.)

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SIP Profile Configuration used by SIP trunk to Cisco UBE (Continued…)

Figure 10 SIP Profile (cont.)

SIP Profile Configuration used by SIP trunk to Cisco UBE (Continued…)

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Page 44 of 55

Figure 11 SIP Profile

SIP Trunk Security Profile Configuration

Navigation: SystemSecurity Non Secure SIP Trunk Profile Set Name* = Non Secure SIP Trunk Profile Set Description = Non Secure Profile for CTL authenticated by null String Select Incoming Transport Type* = TCP+UDP. This is used for this example.

Select Outgoing Transport Type = TCP. This is used for this example

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Page 45 of 55

Figure 12 SIP Trunk Security Profile

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Page 46 of 55

SIP trunk to PSTN (Via Cisco UBE) Configuration

Navigation: Devicetrunk Set Device Name* = CenturyLink. This is used for this example Set Description = CenturyLink SIP trunk To CUBE. This is used for this example Set Device Pool* = G729 Pool. This is used for this example Set Media Resource Group List = MRGL_NO_MTP

Figure 13 SIP Trunk (cont.)

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SIP Trunk to Cisco UBE Configuration (Contd…)

Figure 14 SIP Trunk (cont.)

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SIP Trunk to Cisco UBE Configuration (Contd…)

Set Destination Address = Set IP address of ASR-Cisco UBE Set SIP Trunk Security Profile* = Non Secure SIP Trunk Profile for CTL Set SIP Profile* = Century SIP Profile. This is used in this example Set DTMF Signaling Method* = No Preference

Figure 15 SIP Trunk

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Page 49 of 55

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Page 50 of 55

Route Pattern Configuration

Navigation: Call Routing Route/Hunt Route Pattern Set Route Pattern* = 9. @ This is used to route to CenturyLink via ASR Cisco UBE. Set Description = To PSTN via CenturyLink. This text is used to identify this Route Pattern. Set Gateway/Route List* = CenturyLink. This is used for this example. Check ‘Use Calling Party's External Phone Number Mask’ All other values are default

Figure 16 Route Pattern

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Route Pattern Configuration (Continued…)

Figure 17 Route Pattern (cont.)

Route Pattern Configuration (Continued …)

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Page 52 of 55

Figure 18 Route Pattern

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Acronyms

Acronym Definitions

CPE Customer Premise Equipment

Cisco UBE Cisco Unified Border Element

Cisco UCM Cisco Unified Communications Manager

MTP Media Termination Point

POP Point of Presence

PSTN Public Switched Telephone Network

ESBC Enterprise Session Border Controller

SCCP Skinny Client Control Protocol

SIP Session Initiation Protocol

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Important Information THE SPECIFICATIONS AND INFORMATION REGARDING THE PRODUCTS IN THIS MANUAL ARE SUBJECT TO CHANGE

WITHOUT NOTICE. ALL STATEMENTS, INFORMATION, AND RECOMMENDATIONS IN THIS MANUAL ARE BELIEVED TO

BE ACCURATE BUT ARE PRESENTED WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED. USERS MUST TAKE

FULL RESPONSIBILITY FOR THEIR APPLICATION OF ANY PRODUCTS. IN NO EVENT SHALL CISCO OR ITS SUPPLIERS BE

LIABLE FOR ANY INDIRECT, SPECIAL, CONSEQUENTIAL, OR INCIDENTAL DAMAGES, INCLUDING, WITHOUT

LIMITATION, LOST PROFITS OR LOSS OR DAMAGE TO DATA ARISING OUT OF THE USE OR INABILITY TO USE THIS

MANUAL, EVEN IF CISCO OR ITS SUPPLIERS HAVE BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGES

Corporate Headquarters Cisco Systems, Inc. 170 West Tasman Drive San Jose, CA 95134-1706 USA www.cisco.com Tel: 408 526-4000 800 553-NETS (6387) Fax: 408 526-4100

European Headquarters CiscoSystems International BV Haarlerbergpark Haarlerbergweg 13-19 1101 CH Amsterdam The Netherlands www-europe.cisco.com Tel: 31 0 20 357 1000 Fax: 31 0 20 357 1100

Americas Headquarters Cisco Systems, Inc. 170 West Tasman Drive San Jose, CA 95134-1706 USA www.cisco.com Tel: 408 526-7660 Fax: 408 527-0883

AsiaPacific Headquarters Cisco Systems, Inc. Capital Tower 168 Robinson Road #22-01 to #29-01 Singapore 068912 www.cisco.com Tel: +65 317 7777 Fax: +65 317 7799

Cisco Systems has more than 200 offices in the following countries and regions. Addresses, phone numbers, and fax numbers are listed on the Cisco Web site at http://www.cisco.com/go/offices. Argentina • Australia • Austria • Belgium • Brazil • Bulgaria • Canada • Chile • China PRC • Colombia • Costa Rica • Croatia • Czech Republic • Denmark • Dubai, UAE • Finland • France • Germany • Greece • Hong Kong SAR • Hungary • India • Indonesia • Ireland • Israel • Italy • Japan • Korea • Luxembourg • Malaysia • Mexico • The Netherlands • New Zealand • Norway • Peru • Philippines • Poland • Portugal • Puerto Rico • Romania • Russia • Saudi Arabia • Scotland • Singapore • Slovakia • Slovenia • South Africa • Spain • Sweden • Switzerland • Taiwan • Thailand • Turkey Ukraine • United Kingdom • United States • Venezuela • Vietnam • Zimbabwe © 2015 Cisco Systems, Inc. All rights reserved. CCENT, Cisco Lumin, Cisco Nexus, the Cisco logo and the Cisco Square Bridge logo are trademarks of Cisco Systems, Inc.; Changing the Way We Work, Live, Play, and Learn is a service mark of Cisco Systems, Inc.; and Access Registrar, Aironet, BPX, Catalyst, CCDA, CCDP, CCVP, CCIE, CCIP, CCNA, CCNP, CCSP, Cisco, the Cisco Certified Internetwork Expert logo, Cisco IOS, Cisco Press, Cisco Systems, Cisco Systems Capital, the Cisco Systems logo, Cisco Unity, EtherFast, EtherSwitch, Fast Step, Follow Me Browsing, FormShare, GigaDrive, HomeLink, Internet Quotient, IOS, iPhone, iQ Expertise, the iQ logo, iQ Net Readiness Scorecard, iQuick Study, LightStream, Linksys, Meeting Place, MGX, Networking Academy, Network Registrar, Packet, PIX, ProConnect, ScriptShare, SMARTnet, StackWise, The Fastest Way to Increase Your Internet Quotient, and TransPath are registered trademarks of Cisco Systems, Inc. and/or its affiliates in the United States and certain other countries. All other trademarks mentioned in this document or Website are the property of their respective owners. The use of the word partner does not imply a partnership relationship between Cisco and any other company. (0705R) Printed in the USA

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