Cox Sip Trunking Ucm Ube Cube PDF

Application Note
COX SIP Trunking:
Cisco Unified Communications Manager 9.1.2 with Cisco Unified

Border Element 10.0.1 (IOS 15.4(2)T) using SIP
May 06, 2014
2013 Cisco Systems, Inc. All rights reserved.

Important notices, privacy statements, and trademarks of Cisco Systems, Inc. can be found on cisco.com
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Table of Contents
Introduction .............................................................................................................................................................................................................. 3
Network Topology .................................................................................................................................................................................................... 4
System Components ................................................................................................................................................................................................. 5
Hardware Components........................................................................................................................................................................................ 5
Software Requirements ........................................................................................................................................................................................ 5
Features Supported ............................................................................................................................................................................................. 5
Features Not Supported....................................................................................................................................................................................... 5
Caveats ................................................................................................................................................................................................................ 6
Configuration ............................................................................................................................................................................................................ 7
Configuring the Cisco Unified Border Element................................................................................................................................................... 7
Network interface ......................................................................................................................................................................................... 7
Global CUBE settings .................................................................................................................................................................................. 7
Media Passing through CUBE (media flow-through vs. media flow-around)............................................................................................. 8
Codecs ........................................................................................................................................................................................................... 8
Voice translation rule ................................................................................................................................................................................... 8
Dial peer ........................................................................................................................................................................................................ 9
Call flow ...................................................................................................................................................................................................... 13
Configuring the Cisco Unified Communications Manager ............................................................................................................................... 26
Cisco CallManager service parameter ..................................................................................................................................................... 26
Device Pool, Region and Audio Codec Preference List ........................................................................................................................... 39
Offnet calls via Cox SIP Trunk ................................................................................................................................................................. 41
Dialplan ....................................................................................................................................................................................................... 50
Acronyms ............................................................................................................................................................................................................... 54
Important Information ............................................................................................................................................................................................ 55
Appendix A: Test Results ...................................................................................................................................................................................... 56
Table of Figures
Figure 1 Network Topology ................................................................................................................................................................................... 4
Figure 2: Outbound Voice Call ............................................................................................................................................................................ 13
Figure 3: Outboud Fax Call ................................................................................................................................................................................. 13
Figure 4Service Parameter .................................................................................................................................................................................. 26
Figure 5 Service Parameter Cont. ....................................................................................................................................................................... 27
Figure 8 Service Paramater Cont. ....................................................................................................................................................................... 30
Figure 10 Service Parameter Cont. ..................................................................................................................................................................... 32
Figure 15 Service parameter Cont. ..................................................................................................................................................................... 37
Figure 17 Device Pool ........................................................................................................................................................................................... 39
Figure 18 Audio Codec Preference List .............................................................................................................................................................. 39
Figure 19 Region ................................................................................................................................................................................................... 40
Figure 20 SIP Trunk Security Profile ................................................................................................................................................................. 41
Figure 21 SIP Profile ............................................................................................................................................................................................ 42
Figure 22 SIP Profile Cont. .................................................................................................................................................................................. 43
Figure 23 SIP Trunks List.................................................................................................................................................................................... 45
Figure 24 SIP Trunk to CUBE ............................................................................................................................................................................ 46
Figure 25 SIP Trunk to CUBE Cont. .................................................................................................................................................................. 47
Figure 26 SIP Trunk to CUBE Cont. .................................................................................................................................................................. 48
Figure 27 Translation Pattern ............................................................................................................................................................................. 50
Figure 28 Route Pattern ....................................................................................................................................................................................... 51
Figure 29 Route Pattern for Voice ...................................................................................................................................................................... 52
Figure 30 Route Pattern for Fax.......................................................................................................................................................................... 53

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Introduction
Service Providers today, such as Cox, 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.
Cox 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 Unified Communications Manager and Cox Session Border Controller(EdgeMarc), Cisco Unified Border Element (CUBE) 10.0.1 can be
used. The Cisco Unified Border Element 10.0.1 provides demarcation, security, interworking and session control services for Cisco Unified
Communications Manager 9.1.2 connected to Cox IP network.
This document assumes the reader is knowledgeable with the terminology and configuration of CUCM (Cisco Unified Communications
Manager). Only configuration settings specifically required for Cox 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 Unified Communications Manager (Cisco UCM) 9.1.2 and Cisco Unified
Border Element (CUBE) 10.0.1 for connectivity to Cox SIP trunking service. The deployment model covered in this application note is
CPE (Cisco UCM 9.1.2) to PSTN (Cox).
Testing was performed in accordance to Cox generic SIP trunking test methodology and among features verified were basic calls,
DTMF transport, Music on Hold, Semi-attendant and attendant transfers, call forward, conferences, hunt groups, call pickup, call park,
and interoperability with Cisco Unity Connection
The CUCM configuration detailed in this document is based on a lab environment with a simple dial-plan used to ensure proper
interoperability between Cox SIP network and Cisco Unified Communications. 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 Cox SIP trunking network.
This application note does not cover the use of calling search spaces (CSS) or partitions on Cisco Unified Communications Manager. To
understand and learn how to apply CSS and partitions refer to the cisco.com link below:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/collab09/dialplan.html#wpmkr1044275

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Network Topology
Basic Call Setup
SIP Call Server
10.64.0.0/16 Customer Premise Cox

CUBE Cox E-SBC
WAN LAN IP: 10.64.3.64/16
IP: 10.64.3.66 WAN IP:97.79.185.38/26
SIP Cox
Network
SIP
CUBE(15.4)
LAN Session Border
CUCM 9.1.2 Controller (SBC)
SIP IP: 10.80.21.15
IP: 10.80.21.2
PSTN
SIP
10.80.21.0/24
VG224
IP: 10.80.18.60
DID: 678-238-3710 SIP
SIP
SIP SCCP
IP: 10.80.21.23
DID: 678-238-3715
Unity Connection 9.1.2

IP: 10.80.21.5
IP: 10.80.21.18
DID: 678-238-3711
Fax IP: 10.80.21.22
DID: 678-238-3700
Figure 1 Network Topology

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System Components
Hardware Components
Cisco UCM and Unity Connection run on VMware
ISR G2 3945 router
IP phones 9971(SIP), 8961(SIP) and 7975(SCCP)( please consult Features not supported for restrictions)
Cisco Voice Gateways 224
Software Requirements
Cisco Unified Communications Manager 9.1.2
IOS 15.4(2)T for Cisco Unified Border Element 10.0.1

IOS 12.4(22)T5 for VG224 Voice Gateways
Cisco Unity Connection 9.1.2
Features Supported
Incoming and outgoing offnet calls using G711Ulaw (Cox only offer G711Ulaw) with 20ms packetization
Call hold
Call transfer (Semi-attendant and Attendant)
Call conference
Call forward (all, busy, no answer)
Call park
Hunt groups
Calling line (number) identification presentation (CLIP)
Calling line (number) identification restriction (CLIR)
DTMF (RFC2833)
Unified Mobility (Single Number Reach feature)
Media flow-through on CUBE
Features Not Supported

Outbound SIP REFER with Replaces. Cisco UCM does not currently support generation of an outbound SIP REFER with Replaces
message,
Cisco IP phones used in this test do not support Blind transfer, only Semi-attendant and Attendant transfer were tested

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Caveats
VG224 software version does not support T.38 SG3 fax, hence SG3 fax test call is not executed
Cox Lab test network is closed network, some international test cases and early-media test are not executed

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Configuration
Configuring the Cisco Unified Border Element
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 configuredLAN and WAN.
interface GigabitEthernet0/0
description COX Cisco CUCM9.1.2 test CUBE
ip address 10.80.21.15 255.255.255.0
duplex auto
speed auto
ip address 10.64.3.66 255.255.0.0
duplex auto
speed auto
Global CUBE settings
In order to enable CUBE IP2IP gateway functionality, following command has to be entered:
voice service voip
ip address trusted list
ipv4 10.80.21.2
ipv4 10.64.3.64
address-hiding
mode border-element
allow-connections sip to sip
no supplementary-service sip refer
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
sip
asserted-id pai
privacy pstn

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early-offer forced
midcall-signaling passthru
sip-profiles 1
Explanation
Command Description
allow-connections sip to sip Allow IP2IP connections between two SIP call legs
fax protocol Specifies the fax protocol
asserted-id Specifies the type of privacy header in the outgoing SIP requests and response messages
early-offer forced Enables SIP Delayed-Offer to Early-Offer globally
midcall-signaling passthru Passes SIP messages from one IP leg to another IP leg
sip-profiles 1 Apply sip profile at global level
Media Passing through CUBE (media flow-through vs. media flow-around)
Default CUBE configuration enables CUBE to work in flow-through mode(this test use Flow-through mode). In order to enable flow-around
mode, please perform the following actions:
voice service voip
media flow-around
Codecs
Cox offer only G.711ulaw codec for voice call, it allows codecs other than G.711ulaw but will only accept G.711ulaw.
For customers using G.711 alaw codec:
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
Voice translation rule
We can configure voice translation rules to modify the number we want to send. You can create the rule using command voice translation-rule,
apply the rule to the voice translation-profile. Then assign it to associated dial peer.
voice translation-rule 1
Rule 1 /^8\(.*\)/ /\1/

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!
voice translation-profile DIGITSTRIP8
translate called 1
dial-peer voice 100 voip
description outgoing fax to COX
translation-profile outgoing DIGITSTRIP8
destination-pattern 8678.......
session protocol sipv2
session target sip-server
codec g711ulaw
ip qos dscp af32 signaling
Command Description
rule x Detail Match and Replace Rule x
translate called Define the translation profile rule for the called number.
translation-profile outgoing Define a call number translation profile for outgoing calls
Dial peer
CUCM uses dial-peer to route the call based on the digit to route the call accordingly.
! incoming voice call to CUCM
destination-pattern 67823837..
session target ipv4:10.80.21.2
voice-class codec 1
voice-class sip early-offer forced
dtmf-relay rtp-nte

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ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!for outgoing calls to Cox ESBC
description **NA 11 digits dial**
destination-pattern 1[2-9]..[2-9]......
voice-class codec 1
dtmf-relay rtp-nte
no vad
description **NA 10 digits**
destination-pattern [2-9]..[2-9]......
voice-class codec 1
dtmf-relay rtp-nte
no vad

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description **0 or 00 calls to Local or International Operator**
destination-pattern 0T
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
description **NA international call**
voice-class codec 1
dtmf-relay rtp-nte
no vad
description **CCA*North American-3-Digit*Service Numbers**
destination-pattern [2-9]11

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voice-class codec 1
dtmf-relay rtp-nte
no vad
description **Dial Carrier Access Code**
destination-pattern 101....1[2-9]..[2-9]......
voice-class codec 1
no voice-class sip early-offer forced
dtmf-relay rtp-nte
no vad
!outgoing fax call
codec g711ulaw

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Call flow
In the sample configuration presented here, CUCM is provisioned with four-digit directory numbers corresponding to the last four DID digits.
No digit manipulation is performed on the CUBE.
For incoming PSTN calls, the CUBE presents the full ten-digit DID number to CUCM. The CUCM Translation Pattern strips all but the last
four digits and routes the call based on those digits. Voice calls are routed to IP phones; fax calls are routed via a 4-digit route pattern to a SIP
trunk that terminates on the VG224
CPE callers make outbound PSTN calls by dialing a 9 prefix followed by the destination number. For outbound fax calls from the analog fax
endpoint, VG224 translate leading access code 9 to 8. A 9.@ route pattern strips the prefix and routes the call with the remaining digits
via a SIP trunk terminating on the CUBE for Voice call, and 8.@ Route Pattern send all digits to CUBE to use different dial peer and remove
the leading digit 8 and send to Cox network for outbound Fax.
Figure 2: Outbound Voice Call
Figure 3: Outboud Fax Call
Configuration example
The following configuration snippet contains a sample configuration of Cisco Unified Border Element with all parameters mentioned
previously.
! Last configuration change at 19:43:58 UTC Tue May 6 2014 by cisco

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!
version 15.4
service timestamps debug datetime msec
service timestamps log datetime msec
service password-encryption
hostname COX_CUBE
boot-start-marker
boot-end-marker
aqm-register-fnf
logging buffered 51200 warnings
enable secret 4 tnhtc92DXBhelxjYk8LWJrPV36S2i4ntXrpb4RFmfqY
no aaa new-model
no ip domain lookup
ip name-server 10.64.1.3
ip cef

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no ipv6 cef
multilink bundle-name authenticated
password encryption aes
crypto pki trustpoint TP-self-signed-2131491120
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-2131491120
revocation-check none
rsakeypair TP-self-signed-2131491120
crypto pki certificate chain TP-self-signed-2131491120
certificate self-signed 01
3082022B 30820194 A0030201 02020101 300D0609 2A864886 F70D0101 05050030
31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274
69666963 6174652D 32313331 34393131 3230301E 170D3133 31313031 31363436
31315A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649
4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D32 31333134
39313132 3030819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281
8100AC64 457DC991 57967FE0 A1AD6097 4F1358E1 3721B264 13A1D71B 90556619
D711054C F27B071E 91464C54 EACBD884 DC242E08 1BC34A7E 1FA49C2F 4A130BD1
461AC476 BA1352B7 F54C4714 5990E43E 1FF4824D 8A75A247 F4AB488A 3F9EFD9C
6CED7728 4CE96D86 B43594A1 6684B645 4302389A 99F337D9 5C04D4D6 ECD7BA8C

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1AEF0203 010001A3 53305130 0F060355 1D130101 FF040530 030101FF 301F0603
551D2304 18301680 14B70ED5 6EF1FA77 9D2F8B0B 644BF4DE 972096BC 27301D06
03551D0E 04160414 B70ED56E F1FA779D 2F8B0B64 4BF4DE97 2096BC27 300D0609
2A864886 F70D0101 05050003 81810063 E882FC60 E29C53FE 5A982721 14405614
B1A00023 124C03D7 677F2A10 178A4A9A B83448B1 EFBC136A 4080D4FC 493C3CDB
623B6343 A3639AEB 2A7753B8 9DFB4C79 F3BF9E03 A3146AA0 11AA9FC1 9F739424
2E4D57CB 78413BD3 10C790EE CBBBE796 A8490BE1 D0524A64 0259DC8B 91E6A14C
6FAF8DB9 3139310F 425B3B8C 713265
quit
voice-card 0
dsp services dspfarm
voice rtp send-recv
voice service voip
ip address trusted list
ipv4 10.80.21.2
ipv4 10.64.3.64
address-hiding
mode border-element
allow-connections sip to sip
no supplementary-service sip refer
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
sip
asserted-id pai
privacy pstn
early-offer forced
midcall-signaling passthru

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sip-profiles 1
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
voice class sip-profiles 1
request INVITE sip-header SIP-Req-URI modify "; SIP/2.0" ";user=phone SIP/2.0"
request REINVITE sip-header SIP-Req-URI modify "; SIP/2.0" ";user=phone SIP/2.0"
request INVITE sip-header Contact modify ">" ";user=phone>"
request REINVITE sip-header Contact modify ">" ";user=phone>"
request INVITE sip-header To modify ">" ";user=phone>"
request REINVITE sip-header To modify ">" ";user=phone>"
request INVITE sip-header From modify ">" ";user=phone>"
request REINVITE sip-header From modify ">" ";user=phone>"
voice translation-rule 1
rule 1 /^8\(.*\)/ /\1/
voice translation-profile DIGITSTRIP8
translate called 1

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!
license udi pid C3900-SPE250/K9 sn FOC17426ADY
hw-module pvdm 0/0
username cisco privilege 15 password 7 021201503D5715701C40
redundancy
description COX Cisco CUCM9.1.2 test CUBE
ip address 10.80.21.15 255.255.255.0
duplex auto
speed auto
ip address 10.64.3.66 255.255.0.0
duplex auto
speed auto
no ip address
shutdown

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duplex auto
speed auto
no ip address
shutdown
duplex auto
speed auto
ip forward-protocol nd
ip http server
ip http access-class 23
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
ip route 0.0.0.0 0.0.0.0 10.64.3.64
ip route 10.64.0.0 255.255.0.0 10.80.21.1
nls resp-timeout 1
cpd cr-id 1
control-plane

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!
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
destination-pattern 67823837..
session target ipv4:10.80.21.2
voice-class codec 1
dtmf-relay rtp-nte
no vad
description **NA 11 digits dial**
destination-pattern 1[2-9]..[2-9]......

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voice-class codec 1
dtmf-relay rtp-nte
no vad
description **NA 10 digits**
destination-pattern [2-9]..[2-9]......
voice-class codec 1
dtmf-relay rtp-nte
no vad
description **0 or 00 calls to Local or International Operator**
voice-class codec 1
dtmf-relay rtp-nte

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no vad
description **NA international call**
voice-class codec 1
dtmf-relay rtp-nte
no vad
description **CCA*North American-3-Digit*Service Numbers**
destination-pattern [2-9]11
voice-class codec 1
dtmf-relay rtp-nte
no vad
description **Dial Carrier Access Code**

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destination-pattern 101....1[2-9]..[2-9]......
voice-class codec 1
no voice-class sip early-offer forced
dtmf-relay rtp-nte
no vad
codec g711ulaw
sip-ua
sip-server ipv4:10.64.3.64:5060
gatekeeper
shutdown

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telephony-service
max-conferences 8 gain -6
transfer-system full-consult
after-hours block pattern 1 1900
after-hours day Sun 00:00 23:59
after-hours day Mon 00:00 23:59
after-hours day Tue 00:00 23:59
after-hours day Wed 00:00 23:59
after-hours day Thu 00:00 23:59
after-hours day Fri 00:00 23:59
after-hours day Sat 00:00 23:59
line con 0
exec-timeout 0 0
password 7 060506324F41
logging synchronous
login local
line aux 0
line vty 0 4
exec-timeout 0 0
password 7 1511021F0725
logging synchronous
login local
transport input telnet ssh
line vty 5 15
exec-timeout 0 0
logging synchronous

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login local
transport input telnet ssh
scheduler allocate 20000 1000
ntp server 10.10.10.5
end

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Configuring the Cisco Unified Communications Manager
Cisco CallManager service parameter
Go to System > Service Parameters .we leave all fields in the service parameter as default values for this test
Figure 4Service Parameter

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Figure 5 Service Parameter Cont.

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Figure 8 Service Paramater Cont.

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Figure 15 Service parameter Cont.

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Device Pool, Region and Audio Codec Preference List
Navigate to System > Device Pool and click on Add New to create the Device Pool
Navigate to System >Region Information>Audio Codec Preference List to create Codec Preference List
Navigate to System >Region Information>Region to create Region

In our test, we are using the default Factory Default low loss for Audio Codec Preference List, Default for Region and Device Pool.
Figure 17 Device Pool
Figure 18 Audio Codec Preference List

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Figure 19 Region

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Offnet calls via Cox SIP Trunk
Off-net calls are served by SIP trunks configured between CUCM and Cox ESBC. Calls are routed via CUBE.
SIP Trunk Security Profile
Go to System > Security > SIP Trunk Security Profile and click on Add New. The default Non Secure SIP Trunk Profile is used in this test
Figure 20 SIP Trunk Security Profile
Parameter Value Description
Incoming Transport Type TCP + UDP
Outgoing Transport Type UDP SIP trunks to Cox ESBC 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.

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SIP Profile
SIP Profile will be later associated with the SIP trunk.
Navigate to Device > Device Settings > SIP Profile and modify default SIP Profile by clicking on a Copy button in its row.
Figure 21 SIP Profile

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Figure 22 SIP Profile Cont.

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Default MTP Telephony Event 101 RFC2833 DTMF payload type

Payload Type
Require SDP Inactive Exchange for Checked Send SDP with Inactive when call on hold
Mid-Call Media Change
SIP Rel1XX Options Send PRACK for Enable Provisional Acknowlesgements (Reliable 100 messages)
1xx Messages
Ping Interval for In-service and 300 OPTIONS message parameters- interval time
Partially In-service Trunks (seconds)
Ping Interval for Out-of-service 5 OPTIONS message parameters- interval time

Trunks (seconds)

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SIP Trunk Configuration
Create SIP trunks to Cox by navigating to Device > Trunk and clicking Add New button. Same apply to create Sip trunks to Cisco Unity
Connection and VG224
Figure 23 SIP Trunks List

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Figure 24 SIP Trunk to CUBE

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Figure 25 SIP Trunk to CUBE Cont.

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Figure 26 SIP Trunk to CUBE Cont.

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Device Name Cox Name for the trunk
Device Pool Default Default Device Pool is used for this trunk
Media Resource Group List MRGL_MTP MRG with resources:ANN, CFB, MOH and MTP
Significant Digits 4 4 digits Extension for all CPE phones
Redirecting Diversion Header Checked Adding Diversion Header for calls outband from site
Delivery - outbound
Destination Address 10.80.21.15 LAN IP address of the CUBE
SIP Trunk Security Profile Non Secure SIP Trunk SIP Trunk Security Profile configured earlier
Profile
SIP Profile Cox SIP Profile SIP Profile configured earlier
DTMF Signaling Method RFC 2833 RFC 2833 is supported for DTMF transport to/from Cox
Reset the trunk after the configuration is completed. Apply same procedure to create SIP trunks to Cisco Unity Connection and VG224

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Dialplan
Translation pattern Configuration
CUBE send 10 digits number as dialed digits to CUCM, need to configure Translation Pattern to strip the leading 6 digits so the call can deliver
to the 4 digits extension. Navigation to Call Routing > Translation Pattern > Add New
Figure 27 Translation Pattern

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Setting Value Description
Translation pattern 678238.XXXX Incoming called number Pattern from Cox
Discard Digits PreDot specifies how to modify digit before they

are sending to CUBE and then to Cox
Route Pattern configuration

Route patterns are configured as below, Cisco IP phone dial 9+10digits number to access PSTN via CUBE, 9 is removed before send to
CUBE; for FAX call, Access Code 9 is translate to 8 at VG224 and 8+10digits number is send to CUBE to use different dial peer to Cox
network. Incoming fax call to 3710 will send to VG224. 3720 is the Pilot Number for Voice mail to Unity Connection.
Navigate to Call Routing > Route/Hunt > Route Pattern and press Add New button to create Route Patterns
Figure 28 Route Pattern

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Figure 29 Route Pattern for Voice

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Figure 30 Route Pattern for Fax

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Setting Value Description
Route Pattern 9.@ for Voice call and 8.@ for Specify appropriate Route Pattern
fax call
Gateway/Route List COX SIP Trunk name configured earlier
Require Forced Authorization Checked when doing Specify if Authorization Code required when make call through this
Code Authorization Code test Route Pattern
Require Client Matter Code Check when doing Account Specify if Account Code required when make call through this Route
Code test Pattern
Calling Party Transform mask 678238XXXX Specify the Calling Line ID for outgoing call through this Route
Pattern
Discard Digits PreDot for RP 9.@ specifies how to modify digit before they are sending to Cox ESBC
None for Route Pattern 8.@
Acronyms
Acronym Definitions
CPE Customer Premise Equipment
CUBE Cisco Unified Border Element
CUCM 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.

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Application Note
Appendix A: Test Results
ServiceProvider_Det
ailed_Test_Plan.doc

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Tel: 408 526-4000 Tel: 408 526-7660 www.cisco.com
www-europe.cisco.com
800 553-NETS (6387) Fax: 408 527-0883 Tel: +65 317 7777
Tel: 31 0 20 357 1000
Fax: 408 526-4100 Fax: +65 317 7799
Fax: 31 0 20 357 1100
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 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
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, MeetingPlace, MGX, Networking Academy,
Network Registrar, Packet, PIX, ProConnect, ScriptShare, SMARTnet, StackWise, The Fastest Way to Increase Your Internet Quotient, and TransPath

Page 57 of 58
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

Page 58 of 58

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Application Note

COX SIP Trunking:

Cisco Unified Communications Manager 9.1.2 with Cisco Unified

May 06, 2014

2013 Cisco Systems, Inc. All rights reserved.

2013 Cisco Systems, Inc. All rights reserved.

2013 Cisco Systems, Inc. All rights reserved.

Basic Call Setup

SIP Call Server

10.64.0.0/16 Customer Premise Cox

Unity Connection 9.1.2

Figure 1 Network Topology

2013 Cisco Systems, Inc. All rights reserved.

ISR G2 3945 router

Cisco Voice Gateways 224

IOS 15.4(2)T for Cisco Unified Border Element 10.0.1

Cisco Unity Connection 9.1.2

Call transfer (Semi-attendant and Attendant)

Call forward (all, busy, no answer)

Calling line (number) identification presentation (CLIP)

Calling line (number) identification restriction (CLIR)

Unified Mobility (Single Number Reach feature)

Media flow-through on CUBE

Features Not Supported

2013 Cisco Systems, Inc. All rights reserved.

2013 Cisco Systems, Inc. All rights reserved.

description COX Cisco CUCM9.1.2 test CUBE

ip address 10.80.21.15 255.255.255.0

ip address 10.64.3.66 255.255.0.0

Global CUBE settings

voice service voip

ip address trusted list

allow-connections sip to sip

no supplementary-service sip refer

fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw

2013 Cisco Systems, Inc. All rights reserved.

fax protocol Specifies the fax protocol

early-offer forced Enables SIP Delayed-Offer to Early-Offer globally

sip-profiles 1 Apply sip profile at global level

Media Passing through CUBE (media flow-through vs. media flow-around)

voice service voip

For customers using G.711 alaw codec:

voice class codec 1

codec preference 1 g729r8

codec preference 2 g711ulaw

Voice translation rule

Rule 1 /^8\(.*\)/ /\1/

2013 Cisco Systems, Inc. All rights reserved.

voice translation-profile DIGITSTRIP8

dial-peer voice 100 voip

description outgoing fax to COX

translation-profile outgoing DIGITSTRIP8

session protocol sipv2

session target sip-server

ip qos dscp af32 signaling

rule x Detail Match and Replace Rule x

! incoming voice call to CUCM

dial-peer voice 1 voip

session protocol sipv2

session target ipv4:10.80.21.2

voice-class sip early-offer forced

2013 Cisco Systems, Inc. All rights reserved.

ip qos dscp cs4 signaling

description NA 11 digits dial

description NA 10 digits

description 0 or 00 calls to Local or International Operator

description NA international call

description **CCANorth American-3-DigitService Numbers**

description Dial Carrier Access Code