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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
IP: 10.80.21.23
DID: 678-238-3715
IP phones 9971(SIP), 8961(SIP) and 7975(SCCP)( please consult Features not supported for restrictions)
Software Requirements
Cisco Unified Communications Manager 9.1.2
Features Supported
Incoming and outgoing offnet calls using G711Ulaw (Cox only offer G711Ulaw) with 20ms packetization
Call hold
Call conference
Call park
Hunt groups
DTMF (RFC2833)
Cisco IP phones used in this test do not support Blind transfer, only Semi-attendant and Attendant transfer were tested
Cox Lab test network is closed network, some international test cases and early-media test are not executed
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
duplex auto
speed auto
interface GigabitEthernet0/1
duplex auto
speed auto
In order to enable CUBE IP2IP gateway functionality, following command has to be entered:
ipv4 10.80.21.2
ipv4 10.64.3.64
address-hiding
mode border-element
sip
asserted-id pai
privacy pstn
midcall-signaling passthru
sip-profiles 1
Explanation
Command Description
allow-connections sip to sip Allow IP2IP connections between two SIP call legs
asserted-id Specifies the type of privacy header in the outgoing SIP requests and response messages
midcall-signaling passthru Passes SIP messages from one IP leg to another IP leg
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:
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.
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
translate called 1
destination-pattern 8678.......
codec g711ulaw
Command Description
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.
destination-pattern 67823837..
voice-class codec 1
dtmf-relay rtp-nte
no vad
destination-pattern 1[2-9]..[2-9]......
voice-class codec 1
dtmf-relay rtp-nte
no vad
destination-pattern [2-9]..[2-9]......
voice-class codec 1
dtmf-relay rtp-nte
no vad
destination-pattern 0T
voice-class codec 1
dtmf-relay rtp-nte
no vad
destination-pattern 011T
voice-class codec 1
dtmf-relay rtp-nte
no vad
destination-pattern [2-9]11
voice-class codec 1
dtmf-relay rtp-nte
no vad
destination-pattern 101....1[2-9]..[2-9]......
voice-class codec 1
dtmf-relay rtp-nte
no vad
destination-pattern 8678.......
codec g711ulaw
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.
Configuration example
The following configuration snippet contains a sample configuration of Cisco Unified Border Element with all parameters mentioned
previously.
version 15.4
service password-encryption
hostname COX_CUBE
boot-start-marker
boot-end-marker
aqm-register-fnf
no aaa new-model
no ip domain lookup
ip name-server 10.64.1.3
ip cef
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-2131491120
revocation-check none
rsakeypair TP-self-signed-2131491120
certificate self-signed 01
quit
voice-card 0
ipv4 10.80.21.2
ipv4 10.64.3.64
address-hiding
mode border-element
sip
asserted-id pai
privacy pstn
early-offer forced
midcall-signaling passthru
voice translation-rule 1
translate called 1
redundancy
interface GigabitEthernet0/0
duplex auto
speed auto
interface GigabitEthernet0/1
duplex auto
speed auto
interface GigabitEthernet0/2
no ip address
shutdown
speed auto
interface GigabitEthernet0/3
no ip address
shutdown
duplex auto
speed auto
ip forward-protocol nd
ip http server
ip http access-class 23
ip http secure-server
nls resp-timeout 1
cpd cr-id 1
control-plane
destination-pattern 67823837..
voice-class codec 1
dtmf-relay rtp-nte
no vad
destination-pattern 1[2-9]..[2-9]......
dtmf-relay rtp-nte
no vad
destination-pattern [2-9]..[2-9]......
voice-class codec 1
dtmf-relay rtp-nte
no vad
destination-pattern 0T
voice-class codec 1
dtmf-relay rtp-nte
no vad
destination-pattern 011T
voice-class codec 1
dtmf-relay rtp-nte
no vad
destination-pattern [2-9]11
voice-class codec 1
dtmf-relay rtp-nte
no vad
voice-class codec 1
dtmf-relay rtp-nte
no vad
destination-pattern 8678.......
codec g711ulaw
sip-ua
sip-server ipv4:10.64.3.64:5060
gatekeeper
shutdown
max-conferences 8 gain -6
transfer-system full-consult
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
line vty 5 15
exec-timeout 0 0
logging synchronous
end
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
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
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.
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)
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
Redirecting Diversion Header Checked Adding Diversion Header for calls outband from site
Delivery - outbound
SIP Trunk Security Profile Non Secure SIP Trunk SIP Trunk Security Profile configured earlier
Profile
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
Route Pattern 9.@ for Voice call and 8.@ for Specify appropriate Route Pattern
fax call
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
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
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.
ServiceProvider_Det
ailed_Test_Plan.doc
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