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© 2010 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 52 EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
Application Note
Verizon IP Trunking and IP Contact Center Services: Connecting Cisco Unified Communications Manager 8.0.3 with Cisco Unified Border Element 8.5 (Enterprise Edition)
May 10, 2011 - Initial Version
Table of Contents
Introduction .............................................................................................................................................................................................................. 2 Verizon IP Trunking Overview ........................................................................................................................................................................... 2 Verizon IPCC Overview ...................................................................................................................................................................................... 2
Network Topology .................................................................................................................................................................................................... 3 System Components ................................................................................................................................................................................................. 3
Hardware Components ........................................................................................................................................................................................ 3 Software Requirements ....................................................................................................................................................................................... 3
Sample Bill of Materials ........................................................................................................................................................................................... 4 Features and Known Limitations .............................................................................................................................................................................. 4
Features Supported (IP Trunking) ....................................................................................................................................................................... 4 Known Limitations (IP Trunking) ....................................................................................................................................................................... 4 Features Supported (IPCC) .................................................................................................................................................................................. 5 Known Limitations (IPCC) ................................................................................................................................................................................. 5 Cisco UBE Features Roadmap ............................................................................................................................................................................ 5 CISCO UCM 8.X SIP Trunk Deployment Considerations .................................................................................................................................. 6
Call Flow Overview .................................................................................................................................................................................................. 6 Outbound Call Flows ........................................................................................................................................................................................... 6 Inbound Call Flows ............................................................................................................................................................................................. 7
Failover ..................................................................................................................................................................................................................... 7 Known Issues ............................................................................................................................................................................................................ 7
Inbound Call Issues ............................................................................................................................................................................................. 7 New Security Operation in Cisco IOS 15.1.2T .................................................................................................................................................... 9 Redirected Dialed Number Identification Service and Diversion Header ........................................................................................................... 9 RDNIS Configuration in Cisco Unified Communications Manager Administration ......................................................................................... 10 CISCO UCM Administrator>Device>Device Settings>SIP Profile .................................................................................................................. 12 CISCO UCM Administrator>Device>Device Settings>SIP Profile .................................................................................................................. 14
Communications Manager Configuration ............................................................................................................................................................... 15 Media Resource Group List ............................................................................................................................................................................... 15 Media Resource Group ...................................................................................................................................................................................... 15 CODEC Selection using Device Pools and Regions ......................................................................................................................................... 17 Clusterwide Parameters (System- Location and Region) .................................................................................................................................. 19 List of Device Pools and the associated Regions ............................................................................................................................................... 20 List of Phones and ATA Devices ...................................................................................................................................................................... 20 SIP Trunk Configuration ................................................................................................................................................................................... 21 Route Group Configuration ............................................................................................................................................................................... 22 Route List for Voice .......................................................................................................................................................................................... 24 Route List Details for Voice .............................................................................................................................................................................. 24 Route List for FAX ............................................................................................................................................................................................ 25 Route List Details for FAX ............................................................................................................................................................................... 26 Route Plan report for Voice and FAX Offnet calls ............................................................................................................................................ 27 CISCO UBE Example Configuration (North America)..................................................................................................................................... 29
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
EMEA Configuration.............................................................................................................................................................................................. 38 EMEA CISCO UCM Configuration .................................................................................................................................................................. 38 EMEA CISCO UBE dial-peer Configuration .................................................................................................................................................... 43
IPCC Configuration ................................................................................................................................................................................................ 44 IPCC CISCO UCM Configuration .................................................................................................................................................................... 44 IPCC CISCO UBE dial-peer Configuration ...................................................................................................................................................... 46
Troubleshooting ...................................................................................................................................................................................................... 47 References .............................................................................................................................................................................................................. 50 Acronyms ............................................................................................................................................................................................................... 50 Important Information ............................................................................................................................................................................................ 51
Introduction
This application note describes how to configure a Cisco Unified Communications Manager (Cisco UCM) 8.0 and Cisco Unified Border
Element (Cisco UBE) Enterprise Edition 8.5 for connectivity to Verizon’s IP trunking service. The deployment model covered in this
application note utilizes Verizon’s Private IP (commercial MPLS network) to access Verizon IP Trunking. Supplemental guidelines are also
included for using Verizon IP trunking to interface to their IP-based Contact Center Service or IPCC. Please note that in the context of this
document, “IPCC” refers to a cloud-based Contact Center product from Verizon, and should not be confused with a Cisco product. Additional
supplemental guidelines are provided for an EMEA configuration.
Testing was performed in accordance with the test plans for the Verizon IP trunking (US and EMEA), and IP Contact Center services. All
features were verified.Although this document does not detail the results of the testing performed it provides the essential configurations
required for SIP interoperability with Cisco UCM/Cisco UBE and the Verizon IP Trunking and IPCC services.
Verizon IP Trunking Overview
Verizon IP trunking services simplify management of your network and can help drive operational efficiencies. They do this by consolidating
your voice services onto a SIP-based VoIP network, thereby optimizing your data IP network, and controlling costs associated with maintaining
traditional TDM local lines, trunks, and dedicated PRI circuits. Verizon also offer a native IP Trunking option that provides a SIP trunk directly
to your IPPBX, and an IP Integrated Access option that leverages a gateway device so you can interface with legacy Key or PBX systems.
And, Verizon’s latest Burstable Enterprise Shared Trunking (BEST) feature enhancement allows you to share all your voice trunking resources
across your enterprise and lets you use idle trunk capacity in one location to accommodate a traffic increase in another location. BEST helps
control costs, as fewer concurrent calls need to be purchased at each location and resources can be shared to provide time of day benefits and
peak usage management.
Verizon IPCC Overview
Verizon VoIP Inbound is a component of the IP Contact Center (IPCC) portfolio of internetworking services, which tightly couples signaling
and functionality from the Advanced Toll Free and IP networks to deliver the intelligent routing and call treatment required by contact centers.
The IPCC services are network-based and include IP Interactive Voice Response (IVR) in addition to VoIP Inbound.
VoIP Inbound is standards-compliant and provides single-call service that allows PSTN-originated Toll Free calls to seamlessly terminate and
transfer to a SIP or TDM endpoints, without call re-originations that tie up CPE port capacity. VoIP Inbound includes advanced toll free features
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EDCS# 994149 Rev # Initial Version
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including automatic ISDN User Part and SIP Error overflow for reliable termination to SIP or TDM devices anywhere; and, when combined
with IP IVR, supports customer-driven pre/post call routing and/or call treatment and queuing for customers using Cisco ICM
Network Topology
Figure 1. Typical Reference Network
System Components
Hardware Components
CISCO UBE IOS version 15.1.2T2. Primary and Secondary CISCO UBE routers are used for high availability.
Cisco Unified Border Element is an integrated Cisco IOS Software application that runs on various hardware platforms, for more details:
http://www.cisco.com/go/cube
Packet Voice Data Module (PVDM). You will need to install DSP modules on a supported ISR platform if you require MTP,
Transcoding or Conference Bridge resources. These DSP resources are co-resident on the CISCO UBE routers in our lab configuration.
CISCO UCM cluster with (2) Cisco MCS 7800 Series servers (Cisco Unified Communications Manager)
Cisco Unified IP Phones
Analog Telephony Adapter for FAX, modem, or analog phones
Ethernet Switch
WAN router used to terminate the Verizon MPLS network
Software Requirements
Cisco Unified Call Manager 8.0.3
Cisco Unified Border Element CISCO UBE running IOS version 15.1.2T2
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
Sample Bill of Materials
Features and Known Limitations
Features Supported (IP Trunking)
For a full list of supported SIP features please refer to the “Verizon Business Retail VoIP Network Interface
Specification (for non-registering devices)” document.
All Tests were performed according to the” Verizon Business Retail VoIP Interoperability Test Plan” and the “EMEA
Retail - Test Plan” documents.
These documents may be obtained by contacting your Verizon Business Account Representative.
Known Limitations (IP Trunking)
DTMF as RFC2833 NTE (named telephone events) when a compressed audio codec is used. Note: RFC2833 is not currently supported
when using CTI Route-Points on CISCO UCM 8.0. An MTP resource is required to enable DTMF relay for any calls that utilize a CTI
Route-point.
CISCO UBE performs Delayed-Offer to Early-Offer interworking of the initial SIP INIVTE from CISCO UCM. The Cisco UBE device
receives the invite with no SDP then forwards the invite to the SIP network with SDP included.
T.38 Fax relay is not supported by Verizon IP Trunking Service at this time Note: If you have a Cisco Fax Server or other T.38 Fax
device, you will need to ensure that design considerations have been made to support this outside of the Verizon IP Trunking service.
(i.e.…T1 PRI)
Product Description Quantity
MCS7835I3-K9-CMD1 Unified CM 8.0 7835-I3 Appliance 2
CAB-AC AC Power Cord (North America), C13, NEMA 5-15P, 2.1m 4
C2921-VSEC-CUBE/K9 C2921 VSEC CUBE Bundle, PVDM3-32, UC SEC Lic, FL-CUBEE-25 2
S29UK9-1512T Cisco 2901-2921 IOS UNIVERSAL 2
CAB-AC AC Power Cord (North America), C13, NEMA 5-15P, 2.1m 2
WS-C2960G-24TC-L Catalyst 2960 24 10/100/1000, 4 T/SFP LAN Base Image 2
CAB-AC AC Power Cord (North America), C13, NEMA 5-15P, 2.1m 2
CUCM-USR-LIC Top Level Sku For User License 1
LIC-CUCM-BASIC License - 1 Basic User 50
UCM-7835-80 CUCM 8.0 7835 2
VG202 Cisco VG202 Analog Voice Gateway 1
CAB-AC AC Power Cord (North America), C13, NEMA 5-15P, 2.1m 1
SVGXAISK9-15001M Cisco Voice Gateway 20x Series ADVANCED IP SERVICES 1
CP-7962G Cisco Unified IP Phone 7962 2
SW-CCM-UL-7962 CUCM 3.x or 4.x RTU lic. for single IP Phone 7962 2
CP-7965G Cisco Unified IP Phone 7965, Gig Ethernet, Color 3
SW-CCM-UL-7965 CUCM 3.x or 4.x RTU lic. for single IP Phone 7965 3
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EDCS# 994149 Rev # Initial Version
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Outbound SIP REFER with Replaces. CISCO UCM does not currently support generation of an outbound SIP Refer with replaces
messaging.
CISCO UCM 8.0 can only support a single codec between the end device (i.e. IP Phone, ATA) and the SIP Trunk. A workaround for
this used during testing was to create multiple Regions and Device pools in order to control the codec selection prior to being presented
to the SIP Trunk. The end devices were configured with a specific Device Pool based on the codec used for off-net calls. This is
especially true for mid-call codec negotiation, for any calls that require changing of the initial negotiated codec the CISCO UBE device
will insert a transcoder resource in order to avoid a codec mis-match between the SIP provider and the CUCM end-points.
The CISCO UBE device must have transcoder resources configured on the Cisco UBE device and registered with CUCM to support the
mid-call codec change features on Cisco UBE. This feature allows for dissimilar Voice Class Codec configurations on the incoming and
outgoing dial peers.
In order to comply with Verizon’s requirement of supporting g711u as a secondary codec for all calls and for performing Mid-Call
Codec Negotiation, Cisco has provided an acceptable solution of providing this feature via the Cisco UBE by performing transcoding
from Call Manager to SIP Trunk out to the network
Features Supported (IPCC)
For a full list of supported SIP features please refer to the “Verizon Business IP Contact Center (IPCC) Trunk
Interface Network Interface Specification” document.
All Tests were performed according to the” Verizon Business IPCC Interoperability Lab Test Plan”
These documents may be obtained by contacting your Verizon Business Account Representative.
Known Limitations (IPCC)
The IPCC service does not currently support SIP Diversion Headers
The IPCC service does not support FAX
Outbound SIP REFER with Replaces. CISCO UCM does not currently support generation of an outbound SIP Refer with replaces
messaging.
CISCO UCM 8.0 can only support a single codec between the end device (i.e. IP Phone, ATA) and the SIP Trunk. A workaround for
this used during testing was to create multiple Regions and Device pools in order to control the codec selection prior to being presented
to the SIP Trunk. The end devices were configured with a specific Device Pool based on the codec used for off-net calls. This is
especially true for mid-call codec negotiation, for any calls that require changing of the initial negotiated codec the CISCO UBE device
will insert a transcoder resource in order to avoid a codec mis-match between the SIP provider and the CUCM end-points.
The CISCO UBE device must have transcoder resources configured on the Cisco UBE device and registered with CUCM to support the
mid-call codec change features on Cisco UBE. This feature allows for dissimilar Voice Class Codec configurations on the incoming and
outgoing dial peers.
Cisco UBE Features Roadmap
This roadmap lists the features documented in the Cisco Unified Border Element Configuration guide and maps them
to the chapters in which they appear. Also listed here is the Cisco IOS software release that introduced support for a
given feature in a given Cisco IOS software release train.
http://www.cisco.com/en/US/docs/ios/voice/cube/configuration/guide/vb_roadmap_ps5640_TSD_Products_Configuration_Guide_Chapt
er.html
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
CISCO UCM 8.X SIP Trunk Deployment Considerations
There are several design considerations to be taken into account when deploying SIP trunks. The following URL describes those design
considerations.
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/trunks.html#wp1045842
Call Flow Overview
Outbound Call Flows
The same SIP trunks are utilized between CISCO UCM to CISCO UBE for both Voice and FAX off-net calls. However, the call type (i.e.,
Voice vs. FAX) must be differentiated to ensure the desired codec is used. This delineation is achieved by performing digit manipulation at the
Route List prior to the call being delivered to the Route Group.
Each type of device (i.e., IP Phones vs. analog devices for FAX) will have separate Route-Patterns that belong to their respective partition. The
route patterns will then route the call to the specified Route List.
The Route List is used to distinguish a Voice call from a FAX call by manipulating the called party numbers. A voice call is forwarded with a
leading 9. FAX calls will strip the leading 9 and prepend the called party number with an 8. After the digit manipulation, the Route List then
forwards the call to the Route Group, which routes the call to the SIP trunks.
The SIP trunks are the same for ALL calls from CISCO UCM to CISCO UBE (see example call flows below).
The CISCO UBE will then forward the 10 digit user ID (DID) to the SIP Provider to allow the appropriate call routing
Outbound calls can either be sent to the SIP Trunks in a “Top-Down” or “Round-Robin” method.
Regardless of the method used, if when the call gets routed to the CISCO UBE and the CISCO UBE is not able to complete the call , the call is
then routed to the next SIP Trunk or CISCO UBE in the Route-group.
This provides redundancy for outbound calls by using multiple CISCO UBE devices connecting the VZ VoIP network.
Example call flow for Voice Calls (G.729)
Example call flow for FAX Calls (G.711ulaw)
Route
Pattern
9@
For Voice Calls
Route
List
Route
Group
CUCM Cluster
CUBE 2
CUBE
CUBE
VZ
VoIP CUBE 1
SIP
Trunk
No digits stripped on
Voice calls in CUCM 9 is stripped in CUBE
for Voice calls
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
Inbound Call Flows
Inbound calls are received from either the IP Trunking or IPCC services. These services provide a 10 digit DID for
domestic customers and a variable length DID (10, 11,12, or 13 dependent upon country) for EMEA customers for
delivery of the SIP call. The IP PBX (Cisco UCM) is then responsible for routing this call to the appropriate IP Phone
or analog device.
Failover
The VoIP Network sends periodic SIP options messages as a keepalive mechanism to determine the state of the
CISCO UBE devices. If the primary CISCO UBE does not respond to these options messages, the calls are then
routed to the Secondary CISCO UBE router.
Note: The CISCO UBE will respond to the SIP options pings by default. NO additional configuration is necessary.
The VoIP network will also re-route any calls to the secondary CISCO UBE if it receives a temporary call setup
failure SIP message from the primary CISCO UBE. (Example: 503 or 404 messages)
To allow failover for inbound calls when the primary CISCO UBE device is unable to contact the CISCO UCM
cluster.
In the CISCO UBE:
Configure “voice-class sip options-keepalive” on all dial-peers connecting to the CISCO UCM cluster.
Change the PSTN cause code mapping under the SIP-UA configuration "set pstn-cause 1 sip-status 503"
Without this configuration the incoming call setup from the VZ IP trunking service may time-out and the call would
be cancelled before trying the secondary CISCO UBE device.
Known Issues
Inbound Call Issues
When an inbound (from PSTN to Customer IP PBX) call to a DID that terminates on the SIP trunk is not defined/registered on the IP-PBX, the
IP-PBX should respond with a 40X error message.
Route
Pattern
9@
For FAX Calls
Route
List
Route
Group
CUCM Cluster
CUBE 2
CUBE
CUBE
A VZ
VoIP CUBE 1
SIP
Trunk
9 is stripped on FAX calls
in CUCM and replaced
with 8 8 is stripped in CUBE
for FAX calls
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
There are configurations on the Cisco UBE device that can cause this type of call failure to result in a “call loop”. This is where the call setup is
routed between the VZ VoIP network and the Cisco UBE device continually until it exceeds a timeout threshold.
An Example of this scenario is when the outbound dial-peer on the CISCO UBE is configured with a destination-pattern of .T, which is used as
a gateway of last resort for all calls.
When the Cisco UCM responds with a 40X error message the CISCO UBE will “hunt” for the next available dial-peer to route the call through.
Example:
dial-peer Voice 100 voip
description OUTBOUND G729 Voice SIP calls to VzB
translation-profile outgoing DIGITSTRIP-9
destination-pattern .T **This will match any combination of dialed digits and is not
the recommended configuration for matching outbound calls.
It is recommended to prohibit the matching of assigned DIDs on a dial-peer that is
used to route calls towards the VoIP network.
Voice-class codec 1
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
ip qos dscp af32 signaling
no vad
If the dial-plan requires the use of the above configuration it will become necessary to configure the CISCO UCM facing dial-peers with the
“huntstop” feature to prevent inbound calls from trying to route back to the Verizon VoIP network.
Example:
dial-peer Voice 102 voip
description To/From CISCO UCM subscriber for Voice
preference 2
**The preferred dial-peer with a session target of the subscriber CISCO UCM(huntstop
is not applied here).
destination-pattern [1-5]...
voice-class sip options-keepalive
Voice-class codec 1
session protocol sipv2
session target ipv4:192.168.3.11
incoming called-number 9T
FAX rate disable
no vad
!
dial-peer Voice 103 voip
description To/From CISCO UCM publisher for Voice
preference 5
**The preferred dial-peer with a session target of the subscriber CISCO UCM (huntstop
is applied)
huntstop
destination-pattern 1...
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EDCS# 994149 Rev # Initial Version
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voice-class sip options-keepalive
Voice-class codec 1
session protocol sipv2
session target ipv4:192.168.3.10
incoming called-number 9T
dtmf-relay rtp-nte
no vad
New Security Operation in Cisco IOS 15.1.2T
To help mitigate toll fraud opportunities, as of 15.1.2T CISCO IOS no longer allows connections from "unknown"
sources to connect by default. Only sources on the IP Trust List are allowed (by default) and all other calls are
rejected.
IP addresses defined in the "session target ipv4:" commands on dial-peers are automatically included in the IP Trust
List. Additional valid source IP addresses can be added manually to the Trust List if needed by using the following
CLI:
voice service voip
ip address trusted list
ipv4 10.0.1.24
While it is recommended to use the increased security operation available in 15.1.2T, pre-15.1.2T IOS operation can
be restored by using the CLI:
no ip address trusted authenticate
Redirected Dialed Number Identification Service and Diversion Header
Starting with CISCO UCM Release 6.1(4) adds the Redirected Dialed Number Identification Service (RDNIS) and
diversion header capability for certain calls that use the Cisco Unified Mobility Mobile Connect feature.
The RDNIS/diversion header for Mobile Connect enhances this Cisco Unified Mobility feature to include the RDNIS
or diversion header information on the forked call to the mobile device. Service providers and customers use the
RDNIS for correct billing of end users who make Cisco Unified Mobility Mobile Connect calls.
For Mobile Connect calls, the Service Providers use the RDNIS/diversion header to authorize and allow calls to
originate from the enterprise, even if the caller ID does not belong to the enterprise Direct Inward Dial (DID) range.
Example Use Case
Consider a user that has the following setup:
Desk phone number specifies 89012345.
Enterprise number specifies 4089012345.
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EDCS# 994149 Rev # Initial Version
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Remote destination number specifies 4088810001.
User gets a call on desk phone number (89012345) that causes the remote destination (4088810001) to ring as well.
If the user gets a call from a nonenterprise number (5101234567) on the enterprise number (4089012345), the user
desk phone (89012345) rings, and the call gets extended to the remote destination (4088810001) as well.
Prior to the implementation of the RDNIS/diversion header capability, the fields populated as follows:
Calling Party Number (From header in case of SIP): 5101234567
Called Party Number (To header in case of SIP): 4088810001
After implementation of the RDNIS/diversion header capability, the Calling Party Number and Called Party Number
fields populate as before, but the following additional field gets populated as specified:
Redirect Party Number (Diversion Header in case of SIP): 4089012345
Thus, the RDNIS/diversion header specifies the enterprise number that is associated with the remote destination.
RDNIS Configuration in Cisco Unified Communications Manager Administration
To enable the RDNIS/diversion header capability for Mobile Connect calls, ensure the following configuration takes
place in Cisco Unified Communications Manager Administration:
All gateways and trunks must specify that the Redirecting Number IE Delivery — Outbound check box gets
checked.
In Cisco Unified Communications Manager Administration, you can find this check box by following the following
menu paths:
For H.323 and MGCP gateways, execute Device > Gateway and find the gateway that you need to configure. In the
Call Routing Information - Outbound calls pane, ensure that the Redirecting Number IE Delivery - Outbound
check box gets checked. For T1/E1 gateways, check the Redirecting Number IE Delivery - Outbound check box in
the PRI Protocol Type Information pane.
• For SIP trunks, execute Device > Trunk and find the SIP trunk that you need to configure.
In the Outbound Calls pane, ensure that the Redirecting Diversion Header Delivery - Outbound check box
gets checked
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EDCS# 994149 Rev # Initial Version
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Early-Media Cut-thru: Enable PRACK on CISCO UCM
Early media refers to media (e.g., audio and video) that is exchanged before the called-party accepts a particular session.
Typical examples of early media generated by the called-party are ringing tone and announcements (e.g., queuing status).
Early media generated by the caller typically consists of voice commands or dual tone multi-frequency (DTMF) tones to drive interactive voice
response (IVR) systems.
Enabling PRACK is required in order to allow early media between CISCO UCM and CISCO UBE.
PRACK- Provisional Acknowledgement to a Session not yet established
• Purpose is to acknowledge progress information on a requested process
• The INVITE Includes a Require header stipulating the User Agent Client (UAC) wants a reliable provisional response
SIP Rel1XX Enabled: This parameter determines whether all SIP provisional responses (other than 100 Trying messages) get sent reliably to
the remote SIP endpoint.
If this parameter is disabled, CISCO CallManager does not acknowledge or confirm 18X messages. Valid values specify True (acknowledge
18X messages with PRACK) or False (do not acknowledge 18X messages with PRACK).
The SIP REL1XX parameter is located in the SIP Profile. Once the SIP Profile has been changed to support PRACK for all messages, the
profile will then need to be applied to the appropriate SIP Trunk device.
CISCO UCM Administrator>Device>Device Settings>SIP Profile
Change the SIP Rel1XX Options from default value of disabled to enabled for all 1xx messages
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EDCS# 994149 Rev # Initial Version
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Note: No changes are required on the CISCO UBE. The CISCO UBE supports PRACK and Early Media by default.
Known Issue on CUCM 8.0 with PRACK enabled:
Semi-attended call transfers over SIP Trunk results in one-way audio with Prack enabled.
Current workaround is to disable Prack on the SIP Trunk interface in CUCM 8.0.
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EDCS# 994149 Rev # Initial Version
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CISCO UCM Administrator>Device>Device Settings>SIP Profile
If Early-media is required as mentioned previously in this document, then PRACK will need to be enabled and the end-users will need
to ensure they use fully attended transfer method to transfer calls.
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EDCS# 994149 Rev # Initial Version
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Communications Manager Configuration
Media Resource Group List
List of Media Resource Groups configured for the SIP Trunk MRGL
Media Resource Group
Configured Conference Bridge resource associated with DSP resources configured on CISCO UBE
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
CODEC Selection using Device Pools and Regions
All Voice calls through the SIP trunk should use G.729 and FAX devices should use G.711. Note in the configuration
below, there are two regions. Calls between the “Default” and “SIP Trunk Offnet” region will use G.729 and calls
between “Default” and “SIP Trunk Offnet” use G.711. Applying this configuration to our testbed, the SIP trunk is
placed in a Device Pool with the “SIP Trunk Offnet” region, and phone devices should be placed in a Device Pool that
with the “Default” region. Devices used for analog FAX should use a Device Pool with the “SIP Trunk Offnet”
region. Devices that belong to the same region are configured to use the G.711 codec
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
Note: With CISCO UCM 8.0 the system defaults for Intra-Region codec preference is to use the highest quality audio
codec. By default this is G722 or G711.
The system default for Inter-Region codec preference is G729.
The above region configuration is used to ensure that these codecs will be used if the system defaults are changed.
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
Clusterwide Parameters (System- Location and Region)
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
List of Device Pools and the associated Regions
List of Phones and ATA Devices
Configured Device Pools will determine the codec used by each endpoint for Offnet SIP calls
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
SIP Trunk Configuration
The SIP Trunk Offnet Device Pool is configured for codec negotiation and the SIP_Trunk_MRGL is selected for Conference Bridge resources.
Note: MTP required Not Selected
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
Route Group Configuration
Both SIP Trunks are members of the same Route Group
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
Route List for Voice
The previously defined ROUTE GROUP is selected in the Voice Route List
Route List Details for Voice
No Digits are discarded for off-net Voice calls.
The leading “9” is preserved when the call is forwarded to the CISCO UBE, this allows the CISCO UBE to differentiate the call as Voice and
use the corresponding G.729 CODEC.
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
Route List for FAX
The previously defined ROUTE GROUP is selected in the FAX Route List (similar to Voice Route List)
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
Route List Details for FAX
The “9” is stripped from the called party number and replaced with an “8”.
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
This dial plan configuration ensures that the user only needs to dial a “9” for Voice and FAX off-net calls.
Route Plan report for Voice and FAX Offnet calls
The configured partition on each endpoint will determine how the Offnet SIP calls get routed and allows for a leading
9 to be dialed regardless of type of device. Phone or FAX devices will be able to use the same dial-plan from the user
perspective.
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
CISCO UBE Example Configuration (North America)
Configuration of Cisco Unified Border Element (CISCO UBE) IOS version 15.1.2T2
Critical commands are marked in Bold with footnotes at bottom of each page
version 15.1
service timestamps debug datetime msec localtime
service timestamps log datetime msec
no service password-encryption
service sequence-numbers
!
hostname CUBE
!
boot-start-marker
boot-end-marker
!
!
logging buffered 5000000
no logging rate-limit
no logging console
!
no aaa new-model
!
no ipv6 cef
ip source-route
ip cef
!
!
ip dhcp pool IPPHONES1
network 192.168.3.0 255.255.255.0
option 150 ip 192.168.3.10
default-router 192.168.3.103
!
ip domain name pipiptrunksit2.gsiv.com2
ip name-server 166.38.98.2
ip name-server 10.0.1.4
multilink bundle-name authenticated
!
!
!
!
!
crypto pki token default removal timeout 0
!
!
voice-card 0
dspfarm
1 (Optional ) DHCP Service: automatically assign IP address and TFTP server (option 150) configuration to IP Phones
2 DNS Domain name for SIP Realm and name server list for DNS resolution
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
dsp services dspfarm
!
!
!
voice service pots
!
voice service voip
ip address trusted list3
ipv4 10.0.1.13 255.255.255.255
ipv4 10.0.1.17 255.255.255.255
ipv4 10.1.0.25 255.255.255.255
ipv4 10.1.0.24 255.255.255.255
address-hiding
allow-connections sip to sip4
fax protocol none
sip
early-offer forced5
midcall-signaling passthru6
!
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
!
voice class codec 2
codec preference 1 g711ulaw
!
!
voice cause-code
!
!
!
voice translation-rule 5
rule 1 /^91\(.*\)/ /+1\1/
rule 2 /^2/ /5302222/
rule 3 /^1/ /5302221/
rule 4 /^4/ /5302224/
rule 5 /^9\(.*\)/ /\1/
!
voice translation-rule 10
rule 2 /^9\(.*\)/ /\1/
!
voice translation-rule 11
rule 2 /^8\(.*\)/ /\1/
3 Only sources on the IP Trust List are allowed (by default) and all other calls are rejected.
4 Allow SIP to SIP call Processing
5 Use this command to forcefully configure a Cisco Unified Border Element to send a SIP invite with SDP on the Out-Leg
(OL), Delayed-Offer to Early-Offer for SIP calls. This is applied to all voip dial-peers. 6 Enables support for SIP Supplementary Services
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
!
!
voice translation-profile DIGITSTRIP87
translate called 11
translate redirect-target 5
translate redirect-called 5
!
voice translation-profile DIGITSTRIP98
translate calling 5
translate called 10
translate redirect-target 5
translate redirect-called 5
!
!
license udi pid CISCO2911/K9
hw-module pvdm 0/0
!
!
!
!
redundancy
!
!
!
translation-rule 711
!
!
!
!
!
interface GigabitEthernet0/0
description connection to Vz IP Network
ip address 172.17.8.10 255.255.255.0
duplex auto
speed auto
!
interface GigabitEthernet0/1
description connection to CUCM LAN
ip address 192.168.3.103 255.255.255.0
duplex auto
speed auto
!
interface GigabitEthernet0/2
ip address dhcp
shutdown
duplex auto
speed auto
7 Strip the leading “8” from outgoing called number, also performs digit manipulation for transferred calls.
8 Strip the leading “9” from outgoing called number, also performs digit manipulation for transferred calls and calling
party number.
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
!
ip forward-protocol nd
!
no ip http server
no ip http secure-server
!
ip route 0.0.0.0 0.0.0.0 172.17.8.1
!
!
!
!
control-plane
!
call treatment on9
call threshold global cpu-avg low 68 high 75
call threshold global total-mem low 75 high 85
call threshold global total-calls low 20 high 40
!
voice-port 0/0/010
input gain -6
output attenuation 4
no non-linear
no vad
playout-delay maximum 120
playout-delay nominal 15
playout-delay minimum low
timeouts interdigit 2
timing digit 300
station-id number 2168
caller-id enable
!
voice-port 0/0/1
input gain -6
output attenuation 4
no non-linear
no vad
playout-delay maximum 120
playout-delay nominal 15
playout-delay minimum low
timeouts interdigit 2
timing digit 300
station-id number 5302221167
caller-id enable
!
voice-port 0/0/2
!
voice-port 0/0/3
!
9 Global Call Admission Control based on Resource utilization
10 Optional FXS port for FAX devices connected directly to the CISCO UBE
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
!
mgcp fax t38 ecm
!
sccp local GigabitEthernet0/1
sccp ccm 192.168.3.10 identifier 2 priority 2 version 7.0
sccp ccm 192.168.3.11 identifier 1 priority 1 version 7.0
sccp
!
sccp ccm group 10
associate ccm 1 priority 1
associate ccm 2 priority 2
associate profile 12 register CONF001
associate profile 10 register XCODE001
!
dspfarm profile 10 transcode11
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 12
associate application SCCP
!
dspfarm profile 12 conference12
description conference bridge
codec g711ulaw
codec g729ar8
maximum sessions 12
associate application SCCP
!
dial-peer voice 1 pots
service session
destination-pattern 2168
incoming called-number 2168
port 0/0/0
!
!
!
dial-peer voice 100 voip
description OUTBOUND to VzB
translation-profile outgoing DIGITSTRIP913
destination-pattern 9T14
session protocol sipv2
session target sip-server
11 DSP Resources for Transcoding registered with CISCO UCM cluster 12
DSP resources for Conferencing registered with CISCO UCM cluster 13
Strip the leading “9” from outgoing called number 14
Match on outbound calls from CISCO UCM with leading “9”
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
voice-class codec 1 offer-all15
voice-class sip asserted-id pai
dtmf-relay rtp-nte16
ip qos dscp af32 signaling
no vad
!
dial-peer voice 101 voip
description INBOUND from VzB
session protocol sipv2
session target sip-server
incoming called-number [1-5]...17
voice-class codec 1 offer-all
dtmf-relay rtp-nte
no vad
!
!
!
!
!
dial-peer voice 102 voip
description OUTBOUND FAX to VzB
translation-profile outgoing DIGITSTRIP818
destination-pattern 8T
no modem passthrough
session protocol sipv2
session target sip-server
voice-class codec 1 offer-all
voice-class sip asserted-id pai
voice-class sip privacy disable
fax rate 14400
ip qos dscp af32 signaling
no vad
!
dial-peer voice 103 voip
description INBOUND FAX dial peer from VzB
translation-profile outgoing DIGITSTRIP8
session protocol sipv2
session target sip-server
incoming called-number 117419
15
Sends a list of all available CODECs to the SIP Network. The “offer-all” keyword sends all available codecs without
filtering based on list configured in the associated voice-class
16
Forwards DTMF tones by using RTP with the Named Telephone Event (NTE) payload type.
17
Enables CISCO UBE to set configuration parameters to incoming calls based on the received called number. This is
required to set the DTMF, FAX, and CODEC parameters for the “In-Leg” of the VoIP call. 18
Strip the leading “8” from outgoing called number
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
voice-class codec 1 offer-all
ip qos dscp af32 signaling
no vad
!
!
!
!
!
!
!
!
!
!
!
dial-peer voice 200 voip
description connection to CM3
preference 5
destination-pattern [1-5]...20
session protocol sipv2
session target ipv4:192.168.3.10
incoming called-number 9T21
voice-class codec 1
voice-class sip options-keepalive22
dtmf-relay rtp-nte
fax rate 14400
no vad
!
dial-peer voice 201 voip
description connection to CM4
preference 2
destination-pattern [1-5]...
session protocol sipv2
session target ipv4:192.168.3.11
incoming called-number 9T
voice-class codec 1
voice-class sip options-keepalive
dtmf-relay rtp-nte
19
Enables CISCO UBE to set configuration parameters to incoming calls based on the received called number. This is
required to set the DTMF, FAX, and CODEC parameters for the “In-Leg” of the VoIP call.
20
Enables CISCO UBE to set configuration parameters and call routing for incoming calls destined to CISCO UCM
endpoints. This will match the incoming called party information.
21
Enables CISCO UBE to set configuration parameters to incoming calls based on the received called number. This is
required to set the DTMF, FAX, and CODEC parameters for the “In-Leg” of the VoIP call.
22 Enables monitoring of dial-peer targets using Out of Dialog Options PING messages. Used here to monitor the status of
the CISCO UCM SIP interface.
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
fax rate 14400
no vad
!
dial-peer voice 2 pots
service session
destination-pattern 5302221167
port 0/0/1
!
dial-peer voice 1167 voip
description INBOUND FAX dial peer from VzB to local FXS
session protocol sipv2
session target sip-server
incoming called-number 5302221167
voice-class codec 1 offer-all
ip qos dscp af32 signaling
no vad
!
dial-peer voice 300 voip
description FAX dial peer from/to CUCM
preference 10
destination-pattern 117423
session protocol sipv2
session target ipv4:192.168.3.10
incoming called-number 8T
voice-class codec 224
ip qos dscp af32 signaling
no vad
!
dial-peer voice 301 voip
description FAX dial peer from/to CUCM
preference 1
destination-pattern 1174
session protocol sipv2
session target ipv4:192.168.3.11
incoming called-number 8T25
voice-class codec 2
ip qos dscp af32 signaling
no vad
!
!
sip-ua
set pstn-cause 1 sip-status 503
set pstn-cause 3 sip-status 50326
23
Enables CISCO UBE to set configuration parameters and call routing for incoming calls destined to CISCO UCM
endpoints. This will match the incoming called party information.
24
CODEC is set on dial-peer to force use of g711ulaw for FAX calls. 25
Match on outbound calls from CISCO UCM with leading “8”
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
retry invite 2
retry bye 2
retry cancel 2
timers trying 550
sip-server dns:pcclv1n0022.pipiptrunksit2.gsiv.com27
g729-annexb override
!
!
!
gatekeeper
shutdown
!
!
line con 0
stopbits 1
line aux 0
stopbits 1
line vty 0 4
exec-timeout 0 0
privilege level 15
logging synchronous
login local
transport input ssh
line vty 5 15
exec-timeout 0 0
privilege level 15
password password
logging synchronous
login local
transport input ssh
!
exception data-corruption buffer truncate
scheduler allocate 20000 1000
ntp master 3
ntp peer 199.249.19.1
ntp peer 199.249.18.1
end
26 Overrides the default value of the SIP status code to correspond with the PSTN cause code. 27
SIP Proxy FQDN name for outbound SIP calls to the IP Trunking service
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
EMEA Configuration
EMEA CISCO UCM Configuration
The following steps are required to enable localised Network tones and User Interface:
1. Download necessary localisation files from http://www.cisco.com/cisco/web/download/index.html (requires valid CCO account)
2. Install localisation software on every Communications Manager in the cluster.
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
This does require a restart to enable the localisation file after installation.
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
3. Using the CISCO UCM Administration website either change the locale information at the device pool level or at the Phone device level.
Example shows change to Network Locale on the Phone configuration page:
Note: User Locale changes the User interface only and is controlled independently of the network tones.
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
4. All EMEA Phones should be setup in similar regions as the North America phone configuration to ensure G729 is the preferred CODEC.
5. Next create a variable-length Route-Pattern with “#” as terminating digit.
Example: 9.011!#
Note: The previously configured Voice Route List is utilized for this route-pattern in order to allow the complete
calling number to be sent to CISCO UBE.
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
EMEA CISCO UBE dial-peer Configuration
The CISCO UBE configuration for EMEA is very similar to the US (Domestic) IP Trunking configuration.
dial-peer Voice 100 voip
description OUTBOUND G729 Voice SIP calls to VzB
translation-profile outgoing DIGITSTRIP9
destination-pattern 9T
voice-class codec 1 offer-all
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
ip qos dscp af32 signaling
no vad
!
dial-peer Voice 101 voip
description INBOUND Voice SIP calls from VzB EMEA
voice-class codec 1 offer-all
session protocol sipv2
session target sip-server
incoming called-number [1-5]...
dtmf-relay rtp-nte
no vad
!
dial-peer Voice 102 voip
description To/From CISCO UCM subscriber for Voice
preference 2
destination-pattern [1-5]...
voice-class sip options-keepalive
voice-class codec 1
session protocol sipv2
session target ipv4:192.168.0.4
incoming called-number 9T
FAX rate disable
no vad
!
dial-peer Voice 103 voip
description To/From CISCO UCM publisher for Voice
preference 5
destination-pattern 1...
voice-class sip options-keepalive
voice-class codec 1
session protocol sipv2
session target ipv4:192.168.0.6
incoming called-number 9T
dtmf-relay rtp-nte
no vad
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
IPCC Configuration
IPCC CISCO UCM Configuration
The CISCO UCM Configuration changes required for IPCC services to work properly are:
Verify all IPCC end-points (Phones and Gateways) are in the same Region to allow negotiation of the G.711ulaw
codec.
Disable diversion-header support on the SIP Trunk device configuration.***Need to check this graphic
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
For out-bound IPCC calls a 9.1800632XXXX Route-pattern must be configured in the Communications Manager.
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
IPCC CISCO UBE dial-peer Configuration
The CISCO UBE dial-peers must be configured to negotiate only the G.711 codec for all IPCC inbound calls.
This requires specific incoming called numbers for IP Toll-Free calls.
Example: User calls 8005551212 and IPCC routes the call to 1212 with the following dial-peer configured on the
CISCO UBE router.
Note: In this example the IPCC network is only sending the last 4 digits of the called number.
dial-peer voice 800 voip
description OUTBOUND to VzB IP Toll Free
translation-profile outgoing DIGITSTRIP9
destination-pattern 91800632T
codec g711ulaw
session protocol sipv2
session target dns:rchtcsd05011.vzbi.com
dtmf-relay rtp-nte
ip qos dscp af32 signaling
no vad
!
!
!
dial-peer voice 801 voip
description G.711 INBOUND from VzB IP Toll Free
codec g711ulaw
session protocol sipv2
session target sip-server
incoming called-number 1212
dtmf-relay rtp-nte
no vad
!
!
!
!
dial-peer voice 802 voip
description G.711 To/From CISCO UCM subscriber IP Toll Free
preference 2
destination-pattern 1212
voice-class sip options-keepalive
codec g711ulaw
session protocol sipv2
session target ipv4:192.168.0.4
dtmf-relay rtp-nte
no vad
!
!
!
!
dial-peer voice 803 voip
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
description G.711 To/From CISCO UCM publisher IP Toll Free
preference 5
destination-pattern 1212
voice-class sip options-keepalive
voice-class codec 2
voice-class sip early-offer forced
session protocol sipv2
session target ipv4:192.168.0.6
dtmf-relay rtp-nte
no vad
Troubleshooting
Always capture logs by enabling logging buffer: “logging buffered 200000”
Remember to disable the console logging: “no logging console”
Add sequence numbering for debugs: “service sequence-number”
Debug Commands
debug ccsip all
debug voip ccapi inout
debug voip dialpeer inout
debug transcoding
debug dspfarm all
The following table lists key "show" commands giving output that enables you to monitor Cisco UBE health, traffic and activity.
Key "show" Commands on Cisco UBE
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
Category Command Information Provided
Configuration show version Displays the version of the image on the router
show flash: Displays information about flash: file system
show ip interface brief Displays brief summary of IP status and configuration
show startup-config Displays the startup configuration on the router
show running-config Displays the present/running con configuration on the
router
show debug Displays the debugs currently enabled
show voice iec desc <> Displays definition of an Internal Error Code
show logging Displays the contents of logging buffers
Traffic show call active voice Displays complete details of an active call like media
settings, call statistics, SRTP on/off, etc.
show call active voice brief Displays a brief version of active voice calls, e.g.
transmitted and received packets and duration of call
show call active voice compact Displays a compact version of active voice calls
show voip rtp connections Displays active RTP connections
show call history voice Displays calls stored in the history table for voice
Router Health show processes cpu sorted
<1min/5min/5sec>"
Displays sorted output based on percentage of CPU
utilization
show processes cpu sorted history Displays CPU history information in a graph format
show memory processor Displays memory statistics
show process memory <> Displays memory per process
show memory debug leaks Runs the memory leak detector
show alignment Displays alignment data and spurious memory references
CAC show call threshold config Displays configured resource information
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
show call treatment config Displays call admission control information
show call treatment stats Displays call treatment statistics
SIP show sip-ua connections udp brief Displays summary of SIP UDP connection information
show sip-ua connections udp detail Displays details of SIP UDP connection information
show sip-ua connections tcp brief Displays summary of SIP TCP connection information
show sip-ua connections tcp detail Displays details of SIP TCP connection information
show sip-ua register status Displays SIP registration status
Transcoding and DSPs show diag Displays diagnostic and hardware information for port
adapters and modules
show sdspfarm units Displays transcoder registration status
show sccp connection Displays the active SCCP connections
show sccp Displays SCCP protocol information
show dspfarm dsp active Displays the active DSPs
show call active voice | inc
CoderTypeRate="
Displays call connectivity, codec and the media type
information
show call active voice comp Displays codec information for transcoding calls
DTMF Relay show call active voice | inc
tx_DtmfRelay
Displays the DTMF-relay used for the call
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
References
Cisco UBE on Cisco.com
http://www.cisco.com/go/cube
CISCO UCM 8x SIP Trunk Documentation:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/trunks.html - wp1044916
Cisco UBE PBX / SP Interoperability
http://www.cisco.com/go/interoperability
Verizon Business IP Trunking Services
http://www.verizonbusiness.com/us/products/voip/trunking/
Early Media and Ringing Tone Generation in the Session Initiation Protocol (SIP)
http://www.ietf.org/rfc/rfc3960.txt
Redirected Dialed Number Identification Service and Diversion Header
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/rel_notes/6_1_4/cucm-rel_note-614.html#wp854592
Cisco Unified Border Element (CUBE) Management and Manageability Specification
http://www.cisco.com/en/US/partner/prod/collateral/voicesw/ps6790/gatecont/ps5640/white_paper_c11-613550.html
Acronyms
Acronym Definition
SIP Session Initiation Protocol
SCCP Skinny Client Control Protocol
TDM Time Division Multiplexing
CISCO UCM Cisco Unified Communications Manager
CISCO UBE Cisco Unified Border Element
PRACK Provisional Response Acknowledgement
TUI Telephony User Interface
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EDCS# 994149 Rev # Initial Version
Note: Testing was conducted in Verizon lab.
Important Information
THE SPECIFICATIONS AND INFORMATION REGARDING THE PRODUCTS IN THIS MANUAL ARE SUBJECT TO CHANGE
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EDCS# 994149 Rev # Initial Version
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