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Application Note

Verizon IP Trunking and IP Contact Center Services: Connecting Cisco Unified Communications Manager 7.1(3) with Cisco Unified Border Element (Enterprise Edition) 1.3 (IOS 15.01M1 Verizon Engineering Special)

© 2010 Cisco Systems, Inc. All rights reserved.Important notices, privacy statements, and trademarks of Cisco Systems, Inc. can be found on cisco.com

Note: Results and technical information in this document were obtained through joint testing of Verizon business and Cisco engineers within Verizon laboratories.EDCS# 856854 Revision: 2

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March 1, 2010

Table of Contents

Introduction.......................................................................................................................................................................................................................................................... 3Verizon IP Trunking Overview........................................................................................................................................................................................................................... 3Verizon IPCC Overview...................................................................................................................................................................................................................................... 3Typical Reference Network:................................................................................................................................................................................................................................ 4System Components............................................................................................................................................................................................................................................ 4

Hardware Components................................................................................................................................................................................................................................... 4Software Requirements.................................................................................................................................................................................................................................. 5

Features and Known Limitations......................................................................................................................................................................................................................... 5Features Supported (IP Trunking)..................................................................................................................................................................................................................5Known Limitations (IP Trunking)..................................................................................................................................................................................................................5Features Supported (IPCC)............................................................................................................................................................................................................................ 6Known Limitations (IPCC)............................................................................................................................................................................................................................ 6Cisco UBE Features Roadmap....................................................................................................................................................................................................................... 6Cisco UCM 7.1 SIP Trunk Deployment Considerations...............................................................................................................................................................................6

Call Flow Overview............................................................................................................................................................................................................................................. 7Outbound Call Flows...................................................................................................................................................................................................................................... 7Example call flow for Voice Calls (G.729).................................................................................................................................................................................................... 7Example call flow for FAX Calls (G.711 ulaw)............................................................................................................................................................................................8Inbound Call Flows........................................................................................................................................................................................................................................ 8

Communications Manager Configuration.........................................................................................................................................................................................................12Cisco UCM SIP Session Expires Timer.......................................................................................................................................................................................................12

Early-Media Cut-thru: Enable PRACK on Cisco UCM..............................................................................................................................................................................13Redirected Dialed Number Identification Service and Diversion Header.........................................................................................................................................................26

RDNIS Configuration in Cisco Unified Communications Manager Administration..................................................................................................................................26EMEA Configuration......................................................................................................................................................................................................................................... 28

EMEA Cisco UCM Configuration...............................................................................................................................................................................................................28EMEA Cisco UBE dial-peer Configuration.................................................................................................................................................................................................32

IPCC Configuration........................................................................................................................................................................................................................................... 34IPCC Cisco UCM Configuration.................................................................................................................................................................................................................34IPCC Cisco UBE dial-peer Configuration...................................................................................................................................................................................................36

Cisco UBE Example Configuration (North America).......................................................................................................................................................................................38Configuration of Cisco Unified Border Element (Cisco UBE) IOS version 15.01M1 ES..........................................................................................................................38

Troubleshooting................................................................................................................................................................................................................................................. 48References.................................................................................................................................................................................................................................................... 49


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Introduction

This application note describes how to configure a Cisco Unified Communications Manager (Cisco UCM) 7.1.3 and Cisco Unified Border Element (Cisco UBE) Enterprise Edition 1.3 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 (United States and EMEA (Europe Middle East and Africa), 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 IP Contact Center (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 IP PBX, 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 -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


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Typical Reference Network:

System Components

Hardware Components

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


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http://www.cisco.com/go/cube

WAN router used to terminate the Verizon MPLS network

Software Requirements

Cisco Unified Call Manager 7.1.3

Cisco Unified Border Element Cisco UBE running IOS version 15.01M1 Verizon Engineering Special

A Verizon specific Engineering Special is required to address the following defects found during testing:

CSCsz40585 http://tools.cisco.com/Support/BugToolKit/search/getBugDetails.do?method=fetchBugDetails&bugId=CSCsz40585

CSCsz45332 http://tools.cisco.com/Support/BugToolKit/search/getBugDetails.do?method=fetchBugDetails&bugId=CSCsz45332

Both of the these defects are related to mid-call codec negotiations that result in a disconnected call. The fixes contained in this engineering special do not allow for changing codecs mid-call, only that a response will be sent to the VoIP network that the call should remain at the codec negotiated in the initial call setup.

If an external end-point is requesting a mid-call change in codecs and only supports a single codec with this change request (i.e. G729 to G711) the Cisco UBE may still drop the call.

*A TAC case must be opened to obtain the Verizon Engineering Special IOS, please reference the above defects when opening the service request.

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)

RFC2833 is not currently supported when using CTI Route-Points on Cisco UCM 7.1.3. An MTP resource is required to enable DTMF relay for any calls that utilize a CTI Route-point.

Cisco UBE performs Delay-Offer-to-Early-Offer interworking of an initial SIP INIVTE from Cisco UCM (Cisco UCM does not currently support Early-Offer)

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)

Mid-call codec negotiation (example: G.729 upspeed to G.711) this capability is not currently supported with Cisco UCM or Cisco UBE.

Outbound SIP REFER with Replaces. Cisco UCM does not currently support generation of an outbound SIP Refer with replaces messaging.

Cisco UCM 7.1.3 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. See configuration section for details


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http://tools.cisco.com/Support/BugToolKit/search/getBugDetails.do?method=fetchBugDetails&bugId=CSCsz45332

http://tools.cisco.com/Support/BugToolKit/search/getBugDetails.do?method=fetchBugDetails&bugId=CSCsz40585

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)

Cisco UBE performs Delay-Offer-to-Early-Offer interworking of an initial SIP INVITE from IPCC to Cisco UCM

The IPCC service does not currently support SIP Diversion Headers

The IPCC services does not support FAX

Outbound SIP REFER with Replaces. Cisco UCM does not currently support generation of an outbound SIP Refer with replaces messaging.

Due to the codec negotiation issues for certain IPCC call flows, (Enhanced Transfer) it is necessary to configure the DIDs used for incoming IPCC calls for the G.711ulaw codec only. This will allow all calls presented by IPCC to negotiate a single codec (G.711ulaw) and allow proper media flow when using advanced call transfer services.

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-gw-roadmap_ps5640_TSD_Products_Configuration_Guide_Chapter.html

Cisco UCM 7.1 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/7x/trunks.html#wp1045842


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http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/trunks.html#wp1045842

http://www.cisco.com/en/US/docs/ios/voice/cube/configuration/guide/vb-gw-roadmap_ps5640_TSD_Products_Configuration_Guide_Chapter.html

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).

Example call flow for Voice Calls (G.729)


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Example call flow for FAX Calls (G.711 ulaw)

Outbound Call Failover

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.

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.

Inbound Call 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” to any dial-peers connecting to the Cisco UCM cluster.


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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 Verizon IP trunking service may time-out and the call would be cancelled before trying the secondary Cisco UBE device.

Known 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.

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 Verizon 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


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If there are dial-plan issues that require 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 prefered 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 prefered dial-peer with a session target of the subscriber Cisco UCM (huntstop is applied)huntstop destination-pattern 1... voice-class sip options-keepalive Voice-class codec 1 session protocol sipv2 session target ipv4:192.168.3.10 incoming called-number 9T


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dtmf-relay rtp-nte no vad


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Communications Manager Configuration

Cisco UCM SIP Session Expires TimerCisco UCM will invoke the Session Expires Timer if the SIP Session Timer for all SIP calls.

This timer is calculated at seconds/2 and the default value is 1800, with this default timer setting SIP calls may disconnect after 15 Min. As a workaround we set this parameter to the maximum value of 86400 in the Cisco UCM Service Parameters.This allows the SIP call to be active for 12 hours before the Cisco UCM SIP session expires timer engages and disconnects the call.

Cisco UCM System Cluster wide Parameters (SIP)


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Early-Media Cut-thru: Enable PRACK on Cisco UCMEarly 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).

Cisco UCM Administrator>System>Service Parameters

Change the SIP Rel1XX Enabled from the default of False to True

Note: No changes are required on the Cisco UBE. The Cisco UBE supports PRACK and Early Media by default.


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Media Resource Group List:List of Media Resource Groups configured for the SIP Trunk MRGL


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Media Resource Group:Configured Conference Bridge resource associated with DSP resources configured on Cisco UBE


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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 “Offsite” region will use G.729 and calls between “Default” and “Offsite” use G.711. Applying this configuration to our testbed, the SIP trunk is placed in a Device Pool with the “Offsite” 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 “Offsite” region. Devices that belong to the same region are configured to use the G.711 codec

Note: With Cisco UCM 7.1 the system defaults for Intra-Region codec preference is to use the highest quality 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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List of Device pools and the associated Regions

List of Phones and ATA Devices: The Device Pools selected determine the codec used for Off-net calls


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SIP Trunk Configuration: The Offsite 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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Route Group Configuration: Both SIP Trunks are members of the same Route Group

Route List for Voice: The previously defined ROUTE GROUP is selected in the Voice Route List


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Route List Details for Voice:


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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.

Route List for FAX:


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The previously defined ROUTE GROUP is selected in the FAX Route List (similar to Voice Route List)

Route List Details for FAX:


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The “9” is stripped from the called party number and replaced with an “8”.

This dial plan configuration ensures that the user only needs to dial a “9” for Voice and FAX off-net calls.

Route Pattern for Voice: This Route Pattern is in the Voice partition and is serviced by the Voice Route List (No Digits are stripped)


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Route Pattern for FAX: This Route Pattern is in the FAX partition and is serviced by the FAX Route List (No Digits are stripped)


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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.


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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.

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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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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http://www.cisco.com/cisco/web/download/index.html

A system restart is required to enable the localisation file after installation.


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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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4. Verify that all devices (Phones and Gateways) that are in the same Region to allow use of the G.711alaw codec. This is similar to the above configuration for FAX end-points.

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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EMEA Cisco UBE dial-peer Configuration

The Cisco UBE configuration for EMEA is very similar to the US (Domestic) IP Trunking configuration.

The major difference being that the dial-peers are configured to support G.711alaw.

Note: With EMEA requiring only a single codec, the creation of separate dial-peers for FAX is not required.

dial-peer Voice 100 voip description OUTBOUND G729 Voice SIP calls to VzB translation-profile outgoing DIGITSTRIP-9 destination-pattern 9T codec g711alaw 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 codec g711alaw 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 codec g711alaw session protocol sipv2 session target ipv4:192.168.3.10 incoming called-number 9T FAX rate disable


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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 codec g711alaw session protocol sipv2 session target ipv4:192.168.3.11 incoming called-number 9T dtmf-relay rtp-nte no vad


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IPCC Configuration

IPCC Cisco UCM Configuration

The Cisco UCM Configuration changes required for IPCC services to work properly are:

1. Verify all IPCC end-points (Phones and Gateways) are in the same Region to allow negotiation of the G.711ulaw codec.

2. Disable diversion-header support on the SIP Trunk device configuration.

3. For out-bound IPCC calls a 9.1800632XXXX Route-pattern must be configured in the Communications Manager.


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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 DIGITSTRIP-9 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


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session target ipv4:192.168.3.10 dtmf-relay rtp-nte no vad!!!!dial-peer voice 803 voip 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 ipv4:192.168.3.11 dtmf-relay rtp-nte no vad


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Cisco UBE Example Configuration (North America)

Configuration of Cisco Unified Border Element (Cisco UBE) IOS version 15.01M1 ES

Critical commands are marked in Bold with footnotes at bottom of the page

version 15.0service timestamps debug datetime msecservice timestamps log datetime msecno service password-encryptionservice sequence-numbers!hostname CUBE3!boot-start-markerboot-end-marker!logging message-counter sysloglogging buffered 100000no logging consoleenable password cisco!no aaa new-modelno network-clock-participate wic 0 !dot11 syslogip source-routeno ip dhcp use vrf connectedip dhcp excluded-address 192.168.3.1 192.168.3.150!!!ip dhcp pool IPPHONES1 network 192.168.3.0 255.255.255.0 default-router 192.168.3.103 option 150 ip 192.168.3.11 !!

1 (Optional ) DHCP Service: automatically assign IP address and TFTP server (option 150) configuration to IP Phones


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ip cef!!ip domain name pipiptrunksit2.gsiv.com2ip name-server 166.38.98.2ip name-server 10.0.1.4!no ipv6 cefmultilink bundle-name authenticated!!!Voice-card 0 dspfarm dsp services dspfarm!!!Voice service voip address-hiding allow-connections sip to sip3 sip early-offer forced4

localhost dns:cucm71.pipiptrunksit2.gsiv.com5

midcall-signaling passthru6!!voice class codec 1codec preference 1 g729r8codec preference 2 g711ulawcodec preference 3 g711alaw!

2 DNS Domain name for SIP Realm and name server list for DNS resolution3 Allow SIP to SIP call processing4

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.5 Configures global settings for substituting a DNS localhost name in place of the physical IP address in the From, Call-ID, and Remote-Party-ID headers of outgoing messages. 6 Enables support for SIP Supplementary Services


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!!!voice translation-rule 8 rule 2 /^8\(.*\)/ /\1/!voice translation-rule 9 rule 2 /^9\(.*\)/ /\1/!!voice translation-profile DIGITSTRIP-87 translate called 8!voice translation-profile DIGITSTRIP-98 translate called 9!!license udi pid CISCO2921/K9!username cisco privilege 15 secret ciscoarchive log config hidekeys!!ip ssh version 2!!

interface GigabitEthernet0/0 description connection to VzB Private IP network ip address 172.16.7.10 255.255.255.0 duplex auto speed auto!interface GigabitEthernet0/1

7 Strip the leading “8” from outgoing called number8 Strip the leading “9” from outgoing called number


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description CISCO UBE inside interface9 ip address 192.168.3.103 255.255.255.0 duplex auto speed auto!interface GigabitEthernet0/2 shutdown duplex auto speed auto!ip route 0.0.0.0 0.0.0.0 172.16.2.1no ip http secure-server!!ip rtcp report interval 10000!!!control-plane!call treatment oncall threshold global cpu-avg low 68 high 75call threshold global total-mem low 75 high 85call threshold global total-calls low 20 high 4010!!!!!!

Voice-port 0/0/011 no non-linear playout-delay maximum 120 playout-delay nominal 15 playout-delay minimum low timeouts interdigit 2

9 No SIP bind commands configured, SIP is sourced from both inside and outside interfaces10 Global Call Admission Control based on Resource utilization 11 Optional FXS port for FAX devices connected directly to the Cisco UBE


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timeouts call-disconnect 3 timing digit 300 caller-id enable!Voice-port 0/0/1!mgcp fax t38 ecmmgcp behavior g729-variants static-pt!sccp local GigabitEthernet0/0sccp ccm 192.168.3.11 identifier 2 priority 2 version 7.0sccp ccm 192.168.3.10 identifier 5 priority 1 version 7.0sccp!sccp ccm group 10 associate ccm 5 priority 1 associate ccm 2 priority 2 associate profile 12 register conf003 associate profile 11 register xcode003!dspfarm profile 11 transcode12codec g711ulawcodec g711alawcodec g729ar8codec g729abr8codec g729r8maximum sessions 10! dspfarm profile 12 conference13 description conference bridge codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 10 associate application SCCP!12 DSP Resources for Transcoding registered with Cisco UCM cluster13 DSP resources for Conferencing registered with Cisco UCM cluster


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!!dial-peer Voice 10 pots description connection to FXS port for FAX calls preference 1 service session destination-pattern 1017 FAX rate disable port 0/0/0!dial-peer Voice 100 voip description OUTBOUND G729 Voice SIP calls to VzB translation-profile outgoing DIGITSTRIP-914

destination-pattern 9T15 Voice-class codec 1 session protocol sipv2 session target sip-server dtmf-relay rtp-nte16 ip qos dscp af32 signaling no vad!dial-peer Voice 101 voip description INBOUND G729 Voice SIP calls from VzB Voice-class codec 1 session protocol sipv2 session target sip-server incoming called-number [1-5]...17 dtmf-relay rtp-nte no vad!dial-peer Voice 102 voip description To/From CISCO UCM subscriber for Voice preference 2

14 Strip the leading “9” from outgoing called number15 Match on outbound calls from Cisco UCM with leading “9”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.


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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 9T18 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 ipv4:192.168.3.1019 incoming called-number 9T dtmf-relay rtp-nte no vad!dial-peer Voice 200 voip description inbound FAX dial peer from VZ session protocol sipv2 session target sip-server incoming called-number 101820

codec g711ulaw21

FAX rate disable22 no vad!dial-peer Voice 201 voip

18

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.19 Multiple SIP trunks configured for redundant connections to the Cisco UCM cluster20 Match on inbound call-leg for FAX calls. This is required to set the DTMF, FAX, and CODEC parameters for the “In-Leg” of the VoIP call.21

CODEC is set on dial-peer to force use of g711ulaw for FAX calls.22

Disables FAX relay transmission capability. FAX-Passthrough is the supported FAX method.


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description outbound FAX calls to VZ translation-profile outgoing DIGITSTRIP-823

destination-pattern 8T24 session protocol sipv2 session target sip-server codec g711ulaw FAX rate disable ip qos dscp af32 signaling no vad!dial-peer Voice 202 voip description To/From CISCO UCM subscriber for FAX Calls preference 2 destination-pattern 1018 voice-class sip options-keepalive25 session protocol sipv2 session target ipv4:192.168.3.10 incoming called-number 8T26 codec g711ulaw FAX rate disable no vad!!!dial-peer Voice 203 voip description To/From CISCO UCM publisher for FAX Calls preference 5 destination-pattern 1018 voice-class sip options-keepalive session protocol sipv2 session target ipv4:192.168.3.1127 incoming called-number 8T

23

Strip leading “8” from outbound FAX calls before sending to VZ24

Match outbound FAX calls from Cisco UCM cluster with leading “8” 25 Enables monitoring of dial-peer targets using Out of Dialog Options PING messages.26 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.27 Multiple SIP trunks configured for redundant connections to the Cisco UCM cluster


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codec g711ulaw FAX rate disable no vad!dial-peer voice 800 voip description OUTBOUND to VzB IP Toll Free translation-profile outgoing DIGITSTRIP-9 destination-pattern 91800632T codec g711ulaw session protocol sipv2 session target dns:rchtcsd05011.vzbi.com28 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 ipv4:192.168.3.10 dtmf-relay rtp-nte no vad!dial-peer voice 803 voip description G.711 To/From CISCO UCM publisher IP Toll Free

28 A unique SIP server is used to route calls to the IPCC service vs. the IP Trunking service.


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preference 5 destination-pattern 1212 voice-class sip options-keepalive codec g711ulaw voice-class sip early-offer forced session protocol sipv2 session target ipv4:192.168.3.11 dtmf-relay rtp-nte no vad!sip-ua set pstn-cause 1 sip-status 503 set pstn-cause 102 sip-status 50329 retry invite 2 retry bye 2 retry cancel 2 sip-server dns:pcclv1n0022.pipiptrunksit2.gsiv.com30 g729-annexb override!line con 0line aux 0line vty 0 4!exception data-corruption buffer truncatescheduler allocate 20000 1000ntp master 3ntp peer 199.249.18.1ntp peer 199.249.19.1end

29 Sets the value of the SIP status code that is to correspond with the PSTN cause code.30 SIP Proxy FQDN name for outbound SIP calls to the IP Trunking service


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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

Show Commands

Show voip rtp connection

Show call active voice brief

Show sip-ua calls


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References

Cisco UBE on Cisco.com


Cisco UCM 7.x SIP Trunk Documentation:

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/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


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http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/rel_notes/6_1_4/cucm-rel_note-614.html#wp854592

http://www.ietf.org/rfc/rfc3960.txt

http://www.verizonbusiness.com/us/products/voip/trunking/

http://www.cisco.com/g

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/trunks.html%20-%20wp1044916


Acronyms

Acronym DefinitionsSIP 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

VZ Verizon


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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 DAMAGE


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