cisco ucm 11.0.1 to centurylink sip trunk via cisco ... · voice-class codec 2 voice-class sip bind...
TRANSCRIPT
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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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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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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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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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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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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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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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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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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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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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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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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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Page 32 of 55
!
!
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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Page 39 of 55
“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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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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Page 42 of 55
Figure 9 SIP Profile (cont.)
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Page 43 of 55
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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Page 47 of 55
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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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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Page 51 of 55
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
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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Page 54 of 55
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
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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