Abstract. Avaya Solution Interoperability Test Lab

Transcription

1 Avaya Solution Interoperability Test Lab Configuring AudioCodes Mediant 3000 Gateway, Avaya Aura Session Manager R6.0, Avaya Aura Communication Manager R6.0 and Avaya Voice Portal R5.1 using Intelligent Customer Routing 2.0 Issue 1.0 Abstract These Application Notes describe the configuration of AudioCodes Mediant 3000 Gateway, Avaya Aura Session Manager (SM), Avaya Aura Communication Manager and Avaya Voice Portal (VP) using Intelligent Customer Routing. AudioCodes Mediant 3000 Gateway consolidates PSTN facilities by concentrating and routing the calls over SIP interfaces to Avaya Aura Session Manager. Avaya Aura Session Manager provides SIP proxy/routing functionality, routing SIP sessions across a TCP/IP network with centralized routing policies. Avaya Voice Portal is a Web services based speech enabled interactive voice response system that can accept both traditional DTMF touch tone inputs and prerecorded audio files for output, as well as VoiceXML2.0 compliant speech applications to guide callers through call flows. Intelligent Customer Routing solution provides intelligent self-service capabilities integrating SIP proxy/routing functionality with the ability to provide superior customer to agent contact center operations. These Application Notes provide information for the setup, configuration, and verification of the call flows tested on the Intelligent Customer Routing solution. Testing was conducted at the Avaya Solution and Interoperability Test Lab. 1 of 71

2 Table of Contents: 1. Introduction Intelligent Customer Routing Solution Overview AudioCodes Mediant 3000 Gateway Avaya Aura Session Manager Avaya Voice Portal (VP) Avaya Aura Communication Manager Intelligent Customer Routing Application Server Assumptions Acronyms Lab Configuration Equipment and Software Validated Basic Intelligent Customer Routing Call Flow Call Flows Scenarios Verified Configuring Avaya Aura Communication Manager Verify System Capabilities and Licensing SIP Trunk Capacity Check AAR/ARS Routing Check IP Registration Licensing Check Codec Type Check Configure IP Network Region Add Node Names and IP Addresses Verify/List IP Interfaces Create a Signaling Group for SIP Trunk to Avaya Aura Session Manager Add/Configure a SIP Trunk Group to Connect to Avaya Aura Session Manager Configure AudioCodes Mediant Select all configurable parameters Set SIP Protocols and Ports Setup Codec Preferences/Order Add a SIP Trunk Group Setup Tel to IP Routing Setup Internal DNS Table Setup Alternative Routing for Avaya Aura Session Manager Failover Configure Avaya Aura Session Manager of 71

3 4.1. Specify SIP Domains Add Locations Adaptation Module Setup Add Avaya Voice Portal MPPs as Local Host Entries Add the AudioCodes Mediant 3000 SIP Element Add the Avaya Aura Communication Manager Procr Interface Add Avaya Voice Portal System SIP Element Add Intelligent Customer Routing Server Node SIP Element Define an Element Link for the AudioCodes Mediant Define Element Link for the Avaya Aura Communication Manager Procr Interface Define the Element Link for Avaya Voice Portal - Media Processing Platforms Define the Element Link for the Intelligent Customer Routing Server Node Setup Time Ranges Define Policies and Time of Day for the M3K Define Policies and Time of Day for Avaya Aura Communication Manager Define Policies and Time of Day for Voice Portal Define Dial Plan on Avaya Aura Session Manager for Voice Portal to Avaya Aura Communication Manager Define Dial Plan on Avaya Aura Session Manager to Route Calls to Avaya Voice Portal Configure the Avaya Voice Portal System Configuration Avaya Voice Portal Media Processing Platform Servers Put the MPPs in Service Add the Automated Speech Recognition Server (ASR) Add the Text-To-Speech Server (TTS) Add a SIP Connection for Avaya Aura Session Manager Configure Avaya Intelligent Customer Routing Avaya Voice Portal Administration Main Menu Business Hours and Holidays Configuration Setup ICR Call Center Define the ICR Configuration Skill Setup Self Service Applications ICR Manager Verification Steps of 71

4 7.1. Verify AudioCodes Mediant 3000 Configuration Verify Avaya Aura Session Manager Configuration Verify Avaya Voice Portal Configuration Verify Avaya Aura Communication Manager Configuration Verify Intelligent Customer Routing Configuration Additional References of 71

5 1. Introduction The Intelligent Customer Routing solution is comprised of the following products: Avaya Aura Communication Manager Avaya Aura Session Manager Avaya Voice Portal Avaya Custom Application Server providing the intelligent customer to agent contact center routing AudioCodes Mediant 3000 Gateway In a typical scenario, a customer call enters the AudioCodes Mediant 3000 through the Public Switched Telephone Network (PSTN) over a DS3. The AudioCodes Mediant 3000 delivers the call to Avaya Aura Session Manager over a network connection via SIP. The Avaya Aura Session Manager routes the call to Avaya Voice Portal for customer Self Service (interactive voice response - IVR). After completing Self Service, the customer may opt to speak to a contact center agent, at which point the Avaya Voice Portal solution contacts the Intelligent Customer Routing Application server for the optimal routing location for the customer. The Intelligent Customer Routing service contacts one of many Avaya Aura Communication Managers in the customer s network where agents can provide the best service for the customer. Once analysis of all possible locations is determined, the Intelligent Customer Routing service places the caller in queue on the Avaya Aura Communication Manager. The customer remains on the Avaya Voice Portal solution until an agent is available. Once an agent is available, the call is delivered to the agent. The entire communication beyond the AudioCodes Mediant 3000 gateway utilizes the SIP protocols. These Application Notes describe the administrative steps required for configuring the Avaya products that comprise the Intelligent Customer Routing solution. 5 of 71

6 The following depicts a representative top-level view of a typical solution. Figure 1: Intelligent Customer Routing Solution Overview 6 of 71

7 1.1. Intelligent Customer Routing Solution Overview The following section describes the components of the Intelligent Customer Routing Solution AudioCodes Mediant 3000 Gateway AudioCodes Mediant 3000 provides consolidation of PSTN facilities into SIP. AudioCodes Mediant 3000 is a carrier class product that offers channel scalability of up to 2016 DSOs in a compact 19" 2U chassis. AudioCodes Mediant 3000 provides a web-based user interface that is for operations, administration, management, and provisioning functions. Alternatively, a Solaris based EMS server can be used to provision the AudioCodes Mediant The EMS client communicates with the EMS server from a Microsoft Windows based PC and provides the graphical user interface Avaya Aura Session Manager Avaya Aura Session Manager is a SIP proxy/routing engine that is capable of routing SIP requests throughout a network. The Avaya Aura System Manager provides administration. Avaya Aura Session Manager provides the following functionality: SIP Routing Element The SIP Routing Element provides site to site routing services including number/name resolution, richly manages network ingress and egress including carrier selection for least cost, time of day, load balancing, and media preferences. There may be multiple SIP Routing Elements all share the same configuration data and some real time data. Avaya Aura Session Manager does the following: Routes SIP sessions across the network with centralized routing policies Centralizes SIP registrations and location services Scales to support up to 25,000 locations and up to 250,000 users 1 Enables applications to be decomposed and distributed across the enterprise network Introduces application sequencing preparing for applications to run alongside Avaya Aura Communication Manager. Provides the gateway for the enterprise for external SIP adjuncts. Is available with geographically dispersed redundancy, that is, the Avaya Aura Session Manager instances can be spread across distance (WAN) but in the event of loss of one Avaya Aura Session Manager instance, service still continues to operate normally. All the Avaya Aura Session Manager instances in an enterprise function as a whole, providing continuous service to all users in the event of Avaya Aura Session Manager one instance failures. Each Avaya Aura Session Manager instance operating in the active mode, processes INVITE, REGISTER, SUBSCRIBE and other SIP messages 1 If there are N Session Manager instances in an enterprise, then the total capacity of users that can be supported is = (N-1) X 50, of 71

8 Avaya Voice Portal (VP) Avaya Voice Portal (VP) is a Web based speech enabled interactive voice response system that can accept both traditional DTMF touch tone and/or voice inputs and prerecorded audio files for output. It uses VoiceXML2.0 compliant speech applications to guide callers through Self Service call flows. Avaya Voice Portal is comprised of a Voice Portal Management System (VPMS) server, a Media Processing Platform (MPP) server, a Web Application Server, and a Speech Processing server. Avaya Dialog Designer (DD) applications are deployed on a web application server to perform custom Self Service workflow. Voice Portal Management System (VPMS) manages the MPPs and provides a web interface for administering VP. Media Processing Platform (MPP) provides the main processing for self service applications. Details are described below: Uses H.323, SIP, and RTP protocols to communicate with external services, such as Avaya Aura Communication Manager Runs the VoiceXML browser to interpret VoiceXML2.0 compliant speech applications. Provides proxy interfaces to communicate with the TTS (Text To Speech) servers and ASR (Automatic Speech Recognition) servers. The MPP uses Media Resource Control Protocol (MRCP) to control ASR and TTS servers. Web Application Server The web application server utilizes a workflow defined in Dialog Designer to provide the Self Service application. The MPP calls the application server and coordinates media resources available for processing the call. Speech Server The speech server provides Automatic Speech Recognition (ASR) and Text To Speech (TTS) capabilities Avaya Aura Communication Manager Avaya Aura Communication Manager as an Evolution server provides Avaya Aura Communication Manager features to both SIP and non-sip endpoints. It uses the full call model with Avaya Aura Communication Manager as the only supported application. Additionally, Avaya Aura Communication Manager provides Call Center Software functionality when a customer elects to talk with an agent. Calls are delivered to Avaya Aura Communication Manager either via direct SIP trunks or through Avaya Aura Session Manager Intelligent Customer Routing Application Server A product developed as part of the Intelligent Customer Routing Solution that provides Best Service Routing, Contact Center Queuing on Avaya Aura Communication Manager, and determines optimal customer routing for an enterprise deployment. 8 of 71

9 1.2. Assumptions These Application Notes do not provide any configuration details for the following list of assumptions. AudioCodes Mediant 3000 Gateway is setup and operational (or Solution implementer has sufficient knowledge in AudioCodes Mediant 3000 Gateway setup and configuration) PSTN Connectivity to the AudioCodes Mediant 3000 Gateway has already established and is operational Solution implementer has fairly extensive knowledge in Avaya Voice Portal The Avaya Voice Portal, the speech servers, and the application server have been installed and are operational. The speech servers and the application server are configured with the Avaya Voice Portal. Refer to Avaya Voice Portal Documentation in Section 8. Avaya Aura Session Managers are installed and operational. Avaya Aura Communication Managers are duplicated and have already been installed and configured. The Agents and their respective stations are configured on Avaya Aura Communication Manager Acronyms ARS Alternative Routing Service (Routing on Avaya Aura Communication Manager) ASR Automatic Speech Recognition BSR Best Service Routing (Avaya Aura Communication Manager Feature) CM-ES Communication Manager Evolution Server CM-FS Communication Manager Feature Server DCP Digital Communications Protocol DD Avaya Voice Portal Dialog Designer DNIS Dialed Number Identification Service DNS Domain Naming Resolution DS3 Digital Signal 3 (Digital Signal Level 3 T-Carrier) DTMF Dual Tone Multi Frequency EMS Element Management System (AudioCodes M3K Administration Server) FQDN Fully Qualified Domain Name (hostname for Domain Naming Resolution) ICR Intelligent Customer Routing IP Internet Protocol IPSI IP-services interface (Control Card in Communication Manager) IVR Interactive Voice Response LAN Local Area Network M3K AudioCodes Mediant 3000 Gateway MPP Media Processing Platform (Avaya Voice Portal Server) MRCP Media Resource Control Protocol MUX Multiplexer NTP Network Time Protocol PSTN Public Switched Telephone Network RTP Real Time Protocol 9 of 71

10 SAT SIL SIP SM SMGR SNMP SRE SSH SSL T3 TAC TCP TCP/IP TLS TP TTS URL VDN VP VP 1 st VPMS WAN XML System Access Terminal (Avaya Aura Communication Manager Administration Interface) Solution Interoperability Lab Session Initiation Protocol Avaya Aura Session Manager Avaya Aura System Manager Simple Network Management Protocol SIP Routing Element Secure Shell Secure Socket Layer Digital Carrier Facility (equivalent to DS3) Trunk Access Code (Avaya Aura Communication Manager Trunk Access) Transmission Control Protocol Transmission Control Protocol/Internet Protocol Transport Layer Security Trunk Port Board (part of M3K) Text To Speech Uniform Resource Locator Vector Directory Number Voice Portal Intelligent Customer Routing Voice Portal Management Server Wide Area Network extensible Markup Language 10 of 71

11 1.4. Lab Configuration The following is the Lab configuration used for testing. Figure 2: Intelligent Customer Routing Lab Configuration 11 of 71

12 1.5. Equipment and Software Validated The following equipment and software were used for the sample configuration. Equipment/Software Software Version Intelligent Customer Routing Application Server Dell 1950 RedHat Linux OS RH 5.4 Intelligent Customer Routing Software 2.0 (build 238) AudioCodes Mediant 3000 Gateway 5.80A Audio Codes EMS client Audio Codes EMS Server Avaya Aura System Manager Avaya S8800 Media Server Avaya Aura Session Manager Avaya S8800 Media Server Avaya Voice Portal Voice Portal Management System Media Processing Platform Avaya Aura Communication Manager Avaya S8800 Media Server (R016x ) Avaya G650 Media gateway IPSI (TN2312BP) C-LAN (TN799DP) C-LAN (TN799DP) IP Media Processor (TN2602AP) IP Media Processor (TN2602AP) Avaya IP Telephones: TN2312BP HW26 FW044 TN799DP HW01 FW031 TN799DP HW16 FW031 TN2602AP HW28 FW105 TN2602AP HW08 FW105 FW:1.50 FW:1.50 FW:1.50 FW:2.8.3 Avaya IP-Agent Avaya Digital Telephones (8410D) - 12 of 71

13 1.6. Basic Intelligent Customer Routing Call Flow The following depicts the basic call flow through the Intelligent Customer Routing solution. Figure 3: Basic Call Flow Diagram Scenario: Customer calls Avaya Voice Portal for Self Service and wants to talk to an Agent. AudioCodes Mediant 3000 delivers a call over a SIP Trunk to Avaya Aura Communication Manager. 1. Customer call is received at AudioCodes Mediant AudioCodes Mediant 3000 directs the call to the Avaya Aura Session Manger via SIP. 13 of 71

14 3. Avaya Aura Session Manager routes the call to an Avaya Voice Portal Media Processing Platform 4. The Avaya Voice Portal Media Processing Platform maps the Dialed Number identification Service (DNIS) number to a speech application and begins the Self Service workflow. 5. Customer desires to talk to an agent and selects the appropriate option from the workflow. 6. The Avaya Voice Portal Media Processing Platform contacts the Intelligent Customer Routing solution. 7. The Intelligent Customer Routing solution places polling calls to one or many Avaya Aura Communication Managers to discover the Best Service Routing (BSR) information. 8. With the BSR information, the Intelligent Customer Routing solution determines the appropriate ACD Queue to queue the customers call. Note that the customer call remains on Avaya Voice Portal and does not occupy resources on Avaya Aura Communication Manager. 9. The call is placed in an ACD Queue until an agent become available. 10. When an agent becomes available, the caller and agent are connected through Avaya Aura Session Manager as part of a SIP REFER with REPLACE method. 11. Avaya Aura Session Manager forwards the REFER with REPLACE destination to the AudioCodes Mediant The AudioCodes Mediant 3000 and Avaya Aura Session Manager connect the customer to agent. Note that resulting communication path is from customer, through the AudioCodes Mediant 3000, to Avaya Aura Session Manager, then to Avaya Aura Communication Manager, thereby freeing resources on Avaya Voice Portal Call Flows Scenarios Verified The following outlines the numerous customer scenarios and various call flows: Basic SIP Call flow o Call enters the M3K, going through the SM, to the VP and ends at the Agents phone on CM Direct Agent dialing o Dialing the Agents ID to connect to the particular Agent Direct Calling on Agents extension o Dialing the extension number on which Agent is logged in Transfer and Conference o Using Agent IDs for Blind and Consultative transfer o Using VDN for Blind and Consultative transfer o Using Agent IDs for Conference o Transferring the customers call to the Voice Portal o Conferencing the Voice portal RONA o Short and Long number of rings o Short and Long number of rings to VDN o Short and Long number of rings to different CM s Phone States 14 of 71

15 o Long Ringing > 30 seconds o Short Hold/Long Hold o Long Talk Times > 60 minutes VDN o VDNs with various vector steps that include different wait times, skills having different adjust by values, etc. o Coverage to VDNs o Queuing on VDN with Skills Queuing o Call queued for > 60 minutes Calls Routed Back to VP for Self Service Failover o Failure of 1 MPP (simulated by disconnecting the network interface), active calls remain connected, new calls get distributed to the other MPPs o Failure of 1 MPP (simulated by maintenance down), active calls remain connected, new calls get distributed to the other MPPs o Failure of 1 SM (simulated by disconnecting the network interface), active calls disconnect, new calls get distributed to the other SM o Failure of 1 SM (simulated by maintenance down), active calls disconnect, new calls get distributed to the other SM Load Balancing o The calls get distributed appropriately across the MPPs. Performance/Capacity o Various call rates up to calls per hour. 15 of 71

16 2. Configuring Avaya Aura Communication Manager This section provides the procedures for configuring Communication Manager as an evolution server. The procedures include the following areas: Verify System Capabilities and Licensing Configure IP Network Region Add Node Names and IP Addresses Verify/List IP Interfaces Create Signaling Groups Add/Configure Trunk Groups Throughout this section the administration of Communication Manager is performed using a System Access Terminal (SAT). The following commands are entered on the system with the appropriate administrative permissions. Some administration screens have been abbreviated for clarity. These instructions assume that the Communication Manager has been installed, configured, licensed and provided with a functional dial plan. Additionally, size and scope of a production setup may require multiple SIP signaling groups and SIP trunks to properly handle high call volumes. Note: The Communication Manager is duplicated for testing and these Application Notes assume duplication has been previously administered. The Communication Manager is administered as an evolution server by disabling the IMS enabled? field on the signaling group form. With an evolution server: H.323, digital and analog endpoints are registered with Communication Manager SIP endpoints are registered with Session Manager All endpoints receive service from Communication Manager The Session Manager routes calls from and to SIP endpoints. The SIP endpoints can then communicate with all other endpoints that are connected to Communication Manager. 16 of 71

17 2.1. Verify System Capabilities and Licensing SIP Trunk Capacity Check Issue the display system-parameters customer-options command to verify that an adequate number of SIP trunk members are licensed for the system. The difference between the two values needs to be greater than or equal to the desired number of simultaneous SIP trunk connections. Verify highlighted value, as shown below. display system-parameters customer-options Page 2 of 11 OPTIONAL FEATURES IP PORT CAPACITIES USED Maximum Administered H.323 Trunks: Maximum Concurrently Registered IP Stations: Maximum Administered Remote Office Trunks: 0 0 Maximum Concurrently Registered Remote Office Stations: 0 0 Maximum Concurrently Registered IP econs: Max Concur Registered Unauthenticated H.323 Stations: Maximum Video Capable Stations: Maximum Video Capable IP Softphones: Maximum Administered SIP Trunks: AAR/ARS Routing Check Navigate to Page 3 and verify that ARS is set to y. display system-parameters customer-options Page 3 of 11 OPTIONAL FEATURES Abbreviated Dialing Enhanced List? n Audible Message Waiting? n Access Security Gateway (ASG)? n Authorization Codes? n Analog Trunk Incoming Call ID? n CAS Branch? n A/D Grp/Sys List Dialing Start at 01? n CAS Main? n Answer Supervision by Call Classifier? n Change COR by FAC? n ARS? y Computer Telephony Adjunct Links? y ARS/AAR Partitioning? y Cvg Of Calls Redirected Off-net? y ARS/AAR Dialing without FAC? y DCS (Basic)? y ASAI Link Core Capabilities? y DCS Call Coverage? y 17 of 71

18 IP Registration Licensing Check On Page 10 of system-parameters customer options, verify that there are sufficient AgentSC and IP_API_A licenses. If not, contact an authorized Avaya account representative to obtain these licenses. display system-parameters customer-options Page 10 of 11 MAXIMUM IP REGISTRATIONS BY PRODUCT ID Product ID Rel. Limit Used AgentSC : IP_API_A : IP_Agent : IP_IR_A : IP_Phone : IP_ROMax : IP_Soft : IP_eCons : onex_comm : Codec Type Check Issue the change ip-codec-set 1 command. Ensure Communication Manager is configured to use audio codecs G.729 and G.711MU. Voice Portal will negotiate the audio codec accordingly. Audio Codec: G729 Silence Suppression: n Frames Per Pkt: 2 Packet Size (ms): 20 Repeat for G.711MU Audio Codec: G.711MU Silence Suppression: y Frames Per Pkt: 2 Packet Size (ms): 20 change ip-codec-set 1 Page 1 of 2 IP Codec Set Codec Set: 1 Audio Silence Frames Packet Codec Suppression Per Pkt Size(ms) 1: G.729 n : G.711MU y of 71

19 2.2. Configure IP Network Region Issue the change ip-network-region 1 command and set the Intra-region IP-IP Direct Audio, and Inter-region IP-IP Direct Audio fields to yes. For the Codec Set enter the corresponding audio codec set configured in Section Set the Authoritative Domain to the SIP domain i.e. dr.avaya.com. *Note: In the test configuration, network region 1 was used. When creating a new network region or modifying an existing one, remember to configure it with the correct parameters. change ip-network-region 1 Page 1 of 19 IP NETWORK REGION Region: 1 Location: Authoritative Domain: dr.avaya.com Name: MEDIA PARAMETERS Intra-region IP-IP Direct Audio: yes Codec Set: 1 Inter-region IP-IP Direct Audio: yes UDP Port Min: 2048 IP Audio Hairpinning? n UDP Port Max: Add Node Names and IP Addresses Issue the change node-names ip command and add the node-names and IP addresses for the Session Managers, if not already previously added. In this sample configuration, silasm1 and silasm2 are the Session Managers. *Note: These may have been already created and do not need to be re-created if they are present in the node-names list. change node-names ip Page 1 of 2 IP NODE NAMES Name IP Address silasm silasm procr of 71

20 2.4. Verify/List IP Interfaces Issue the list ip-interface all command, and note PROCR to be used for SIP signaling groups (below) to the Session Managers. list ip-interface all IP INTERFACES Net ON Type Slot Code/Sfx Node Name/ Mask Rgn VLAN IP-Address/ Gateway Node y PROCR procr / n PROCR procr6 /64 1 :: :: 20 of 71

21 2.5. Create a Signaling Group for SIP Trunk to Avaya Aura Session Manager Issue the add signaling-group n command, where n is an available signaling group number for one of the SIP trunks to the Session Manager, and fill in the indicated fields. Default values can be used for the remaining fields. Group Type: sip Transport Method: tcp Near-end Node Name: procr Far-end Node Name: Session Manager node name from Section 2.3 Near-end Listen Port: 5060 Far-end Listen Port: 5060 DTMF over IP: rtp-payload *Note: Default values for the IMS Enabled, Peer Detection Enabled and Peer Server fields align with Communication Manager as an evolution server. add signaling-group 40 Page 1 of 1 SIGNALING GROUP Group Number: 40 Group Type: sip IMS Enabled? n Transport Method: tcp Q-SIP? n IP Video? n Peer Detection Enabled? y Peer Server: Others SIP Enabled LSP? n Enforce SIPS URI for SRTP? y Near-end Node Name: procr Far-end Node Name: silasm1 Near-end Listen Port: 5060 Far-end Listen Port: 5060 Far-end Network Region: Far-end Secondary Node Name: Far-end Domain: Bypass If IP Threshold Exceeded? n Incoming Dialog Loopbacks: eliminate RFC 3389 Comfort Noise? n DTMF over IP: rtp-payload Direct IP-IP Audio Connections? y Session Establishment Timer(min): 3 IP Audio Hairpinning? n Enable Layer 3 Test? n Initial IP-IP Direct Media? n H.323 Station Outgoing Direct Media? n Alternate Route Timer(sec): 6 21 of 71

22 2.6. Add/Configure a SIP Trunk Group to Connect to Avaya Aura Session Manager Add the corresponding trunk group controlled by the above signaling group via the add trunkgroup n command, where n is an available trunk group number and fill in the indicated fields. Group Type: sip Group Name: A descriptive name. TAC: An available trunk access code. Service Type: tie Signaling Group: The number of the signaling group added in Section 2.5 Number of Members: The number of SIP trunks to be allocated to calls routed to Session Manager (must be within the limits of the total trunks shown in Section 2.1.1). *Note: The number of members determines how many simulataneous calls can be processed by the trunk through SM. add trunk-group 40 Page 1 of 21 TRUNK GROUP Group Number: 40 Group Type: sip CDR Reports: y Group Name: sip tg to silasm1 COR: 1 TN: 1 TAC: #040 Direction: two-way Outgoing Display? n Dial Access? n Night Service: Queue Length: 0 Service Type: tie Auth Code? n Member Assignment Method: auto Signaling Group: 40 Number of Members: of 71

23 On Page 3: Numbering Format: UUI Treatment: public shared Use default values for all other fields. The trunk members will be automatically generated based on the value of Number of Members on Page 1. add trunk-group 40 Page 3 of 21 TRUNK FEATURES ACA Assignment? n Measured: both Maintenance Tests? y Numbering Format: public UUI Treatment: shared Maximum Size of UUI Contents: 128 Replace Restricted Numbers? n Replace Unavailable Numbers? n *Note: Additional Communication Manager administration is necessary to configure VDNs, vectors and agent. See the product documentation for Intelligent Customer Routing. To persist the Communication Manager changes issue the save translation all command. 23 of 71

24 3. Configure AudioCodes Mediant 3000 This section provides the procedures for configuring M3K using the web based graphical user interface. The procedures include the following areas: Select all configurable parameters Setting SIP Protocols and ports Setup Codec preferences Add a SIP Trunk Group Setup Tel to IP routing Setup Internal DNS Table Setup Tel-to-IP Alternative Routing These Application Notes assume the AudioCodes Mediant 3000 is already installed and functioning properly. See the reference document listed in Section 8 for the AudioCodes Mediant 3000 Setup Guide. Log in to the M3K with Login Name having administrative rights. 24 of 71

25 3.1. Select all configurable parameters The Mediant 3000 Home Page will be displayed. Be sure to display all configurable parameters by selecting Full in the left pane. In the screenshot below, both TP6310 and SA boards are shown. 25 of 71

26 3.2. Set SIP Protocols and Ports Double-click on Protocol Configuration, expand Protocol Definition and select SIP General Parameters. This will display the SIP General Parameters in the right pane. Scroll down and set the SIP Transport Type, SIP UDP Local Port, SIP TCP Local Port and SIP TLS Local Port fields appropriately. Scroll down further and set the SIP Destination Port and Use user=phone in SIP URL fields. 26 of 71

27 Optionally, scroll down further and set the User-Agent Information and SDP Session Owner fields. When finished editing the SIP General Parameters, click Submit and then Burn to save the changes. 27 of 71

28 3.3. Setup Codec Preferences/Order Expand Protocol Configuration and Coders And Profile Definitions in the left pane. Click on the Coders to display the Coders Table in the right pane. Add entries for the intended codecs. These Application Notes display settings for G.711MU and G.729 codec sets. Note: G.729 is supported in Voice Portal of 71

29 3.4. Add a SIP Trunk Group Expand Protocol Configuration and Trunk Group in the left pane. Select Trunk Group. The Trunk Group Table is displayed in the right pane. Select the Group Index to configure. Use the dropdown menu to select the trunks for From Trunk and To Trunk. For the Channels field type in the desired channels. For the Trunk Group ID field enter the desired trunk group ID. Click Submit and then Burn. 29 of 71

30 3.5. Setup Tel to IP Routing Expand Protocol Configuration and Routing Tables in the left pane. Select Tel to IP Routing. The Tel to IP Routing table is displayed in the right pane. Select the desired row of the table and enter routing information for the Dest. Phone Prefix and Source Phone Prefix fields. Input the IP address or FQDN DNS information for the Dest. IP Address field. Use the appropriate port number in the Port field and select the related Transport Type. Click Submit and then Burn. 30 of 71

31 3.6. Setup Internal DNS Table Expand Protocol Configuration and Routing Tables in the left pane. Select Internal DNS Table. The Internal DNS Table is displayed in the right pane. Select the desired row of the table and enter domain information for the Domain Name field. Input the IP address(es) for the downstream Sessions Manager(s) in the First IP Address, Second IP Address, Third IP Address and Fourth IP Address fields. Click Submit and then Burn. 31 of 71

32 3.7. Setup Alternative Routing for Avaya Aura Session Manager Failover Completing this administration will allow the AudioCodes M3K to route calls to the alternative Session Manager(s) upon receiving the appropriate SIP error message. Expand Protocol Configuration and Routing Tables in the left pane. Select Alternative Routing. The Reasons for Alternative Routing table is displayed in the right pane. In the Tel to IP Reasons section, use the dropdown menu to select the expected error code for alternative routing for Reason 1, Reason 2, etc. Click Submit and then Burn. 32 of 71

33 4. Configure Avaya Aura Session Manager This section provides the procedures for configuring Session Manager. Multiple Session Managers were used for redundancy and failover conditions. These instructions assume the System Manager and the Session Managers are functioning properly and do not describe the steps necessary for duplication. Configuration is accomplished by accessing the browser-based GUI of System Manager, using the URL where <ip-address> is the IP address of System Manager. Log in to the System Manager with the appropriate credentials. The menu below is displayed. Expand the Routing link on the left side as shown. The sub-menus displayed in the left column below will be used to configure. 33 of 71

34 4.1. Specify SIP Domains Configure the SIP Domain appropriately. Select Domains from the left hand window. Click New and in the General section, under Name add a descriptive name. Under Notes add a brief description. Click Commit to save. Screen below shows all the list of SIP domains added. Highlighted is the SIP domain used in the sample configuration. 34 of 71

35 4.2. Add Locations Locations are used to identify logical and/or physical locations where SIP Entities reside, for purposes of bandwidth management. Select Locations and click New. In the General section, under Name add a descriptive name. Under Notes add a brief description. Click Commit to save. 35 of 71

36 4.3. Adaptation Module Setup Configure Adaption module to perform digit conversion rules for incoming and outgoing calls appropriately. Select Adaptations from the Routing menu. Click New and in the General section, under Adaptation name add a descriptive name. Under Module name enter the appropriate module. Under Notes add a brief description. Click Commit to save. 36 of 71

37 4.4. Add Avaya Voice Portal MPPs as Local Host Entries Configure the MPPs as a Local Host Name Resolution entry. Expand Elements Session Manager Network Configuration. Select Local Host Name Resolution to enter the details of the MPPs with their respective IP addresses. Click New. Under Host Name add an identifier followed by the SIP domain (added in Section 4.1). This Host Name is going to be common for the n number of MPPs being added. Under IP Address enter the IP address of the respective MPPs. Under Port enter Enter the appropriate Priority and Weight as required. Under Transport select TCP. Click Commit to save. Shown below is the updated screen for the sample configuration. 37 of 71

38 4.5. Add the AudioCodes Mediant 3000 SIP Element Configure the SIP Element for the M3K. Expand Routing and select SIP Elements. Click New. In the General Section, add the following: Name Add an identifier for the M3K FQDN or IP Address Enter the IP Address of M3K Type Select the appropriate menu option for the gateway Notes Add a brief description Adaptation From the drop down select the Adaptation added in Section 4.3 Location From the drop down select the Location added in Section 4.2 Time Zone From the drop down select the appropriate time zone Click Commit to save. 38 of 71

39 4.6. Add the Avaya Aura Communication Manager Procr Interface Configure the SIP Element for the Communication Manager. Expand Routing and select SIP Elements. Click New. In the General Section, add the following: Name Add an identifier for the Communication Manager FQDN or IP Address Enter the IP Address of Communication Manager Type Select the appropriate menu option for Communication Manager Notes Add a brief description Adaptation From the drop down select the Adaptation added in Section 4.3 Location From the drop down select the Location added in Section 4.2 Time Zone From the drop down select the appropriate time zone Click Commit to save. 39 of 71

40 4.7. Add Avaya Voice Portal System SIP Element Configure the SIP Element for the Voice Portal. Expand Routing and select SIP Elements. Click New. In the General Section, add the following: Name Add an identifier for the Voice Portal FQDN or IP Address Enter the Host Name added in Section 4.4 Type Select Voice Portal Notes Add a brief description Adaptation From the drop down select the Adaptation added in Section 4.3 Location From the drop down select the Location added in Section 4.2 Time Zone From the drop down select the appropriate time zone Click Commit to save. 40 of 71

41 4.8. Add Intelligent Customer Routing Server Node SIP Element Configure the SIP Element for the ICR Server Node. Expand Routing and select SIP Elements. Click New. In the General Section, add the following: Name Add an identifier for the ICR Core Server Node FQDN or IP Address Enter the IP Address of ICR Core Server Node Type Select Other Notes Add a brief description Adaptation From the drop down select the Adaptation added in Section 4.3 Location From the drop down select the Location added in Section 4.2 Time Zone From the drop down select the appropriate time zone Click Commit to save. 41 of 71

42 4.9. Define an Element Link for the AudioCodes Mediant 3000 Configure Element Link for the M3K. Expand Routing and select Element Links. Click New. In the Element Links section, add the following: Name Enter an identifier for M3K SIP Element 1 From dropdown, select silasm1 Protocol From dropdown select the required protocol Port Enter the correct port for the Session Manager SIP Element 2 From dropdown, select the SIP Element added in Section 4.5 for M3K Port Enter the correct port for M3K Trusted Ensure the ticked box is clicked Notes Add a brief description Click Commit to save. 42 of 71

43 4.10. Define Element Link for the Avaya Aura Communication Manager Procr Interface Configure the Element Link for the Communication Manager. Expand Routing and select Element Links. Click New. In the Element Links section, add the following: Name Enter an identifier for Communication Manager. SIP Element 1 From dropdown, select silasm1. Protocol From dropdown select the required protocol. Port Enter the correct port for the Session Manager. SIP Element 2 From dropdown, select the SIP Element added in Section 4.6 for Communication Manager. Port Enter the correct port for Communication Manager. Trusted Ensure the ticked box is clicked. Notes Add a brief description. Click Commit to save. 43 of 71

44 4.11. Define the Element Link for Avaya Voice Portal - Media Processing Platforms Configure the Element Link for the MPPs. Expand Routing and select Element Links. Click New. In the Element Links section, add the following: Name Enter an identifier for Voice Portal MPPs SIP Element 1 From dropdown, select silasm1 Protocol From dropdown select the required protocol Port Enter the correct port for the Session Manager. SIP Element 2 From dropdown, select the SIP Element added in Section 4.7 for Voice Portal MPPs Port Enter the correct port for Voice Portal MPPs Trusted Ensure the ticked box is clicked Notes Add a brief description Click Commit to save. 44 of 71

45 4.12. Define the Element Link for the Intelligent Customer Routing Server Node Configure the Element Link for the ICR Core Server node. Expand Routing and select Element Links. Click New. In the Element Links section, add the following: Name Enter an identifier for ICR Core Server node. SIP Element 1 From dropdown, select silasm1. Protocol From dropdown select the required protocol. Port Enter the correct port for the Session Manager. SIP Element 2 From dropdown, select the SIP Element added in Section 4.8 for Intelligent Customer Routing. Port Enter the correct port for ICR Core Server node. Trusted Ensure the ticked box is clicked. Notes Add a brief description. Click Commit to save. 45 of 71

46 4.13. Setup Time Ranges Configure the Time Ranges. Expand Routing and select Time Ranges. Click New. Under Name enter an identifier. Select the days of the week, enter time values for Start Time and End Time. Under Notes add a brief description. When completed, click Commit to save. Screenshot below shows the updated information Define Policies and Time of Day for the M3K Configure Policies and Time of Day for the M3K. Expand Routing and select Policies. Click New. In the General section under Name, add an identifier to define the routing policy for the M3K Under Notes add a brief description. In the SIP Element as Destination section click on Select. 46 of 71

47 The SIP Element List page opens. Select the entry of the M3K added in Section 4.5, and click on Select. The selected SIP Entity displays on the Routing Policy Details page. In the Time of Day section, click on Add, and from the Time Ranges List page (not shown), select the desired Time Range and click on Select. The selected Time Range displays on the Routing Policy Details page. Click on Commit to save. 47 of 71

48 4.15. Define Policies and Time of Day for Avaya Aura Communication Manager Configure the Policies and Timer of Day for the Communication Manager. Expand Routing and select Policies and click New. In the General section, under Name add an identifier to define the routing policy for the Communication Manager. Under Notes add a brief description. In the SIP Element as Destination section click on Select. 48 of 71

49 The SIP Elements list page opens. Select the entry of Communication Manager added in Section 4.6 and click on Select. The selected SIP Entity displays on the Routing Policy Details page. In the Time of Day section, click on Add, and from the Time Ranges List page (not shown) select the desired Time Range and click on Select. The selected Time Range displays on the Routing Policy Details page. Click on Commit to save. 49 of 71

50 4.16. Define Policies and Time of Day for Voice Portal Configure the Policies and Time of Day for Voice Portal. Expand Routing and select Policies. Click New. In the General section, under Name add an identifier to define the routing policy for the Avaya Voice Portal. Under Notes add a brief description. In the SIP Element as Destination section, click on Select. 50 of 71

51 The SIP Element List page opens. Select the entry of the Voice Portal added in the above steps, and click Select. The selected SIP Entity displays on the Routing Policy Details page. In the Time of Day section, click on Add, and from the Time Ranges List page (not shown) select the desired Time Range and click on Select. The selected Time Range displays on the Routing Policy Details page. Click on Commit to save. 51 of 71

52 4.17. Define Dial Plan on Avaya Aura Session Manager for Voice Portal to Avaya Aura Communication Manager Configure the Dial Plan pattern to route calls from Voice Portal to Communication Manager. Expand Routing and select Dial Patterns. Click New. In the General section, under Pattern add the pattern that the Voice Portal will dial-out to reach an extension on Communication Manager. Under Min enter the minimum number digits that must be dialed. Under Max enter the maximum number digits that may be dialed. Select Emergency call if required. Under SIP Domain dropdown, select a SIP Domain added in Section 4.1 or another drop down option. Under Notes add a brief description. 52 of 71

53 In the Originating Locations and Routing Policies section click Add. In the Originating Location section, select the appropriate location. In the Routing Policies section, select the appropriate policy for routing. Click the Select button. Click on Commit to save. 53 of 71

54 4.18. Define Dial Plan on Avaya Aura Session Manager to Route Calls to Avaya Voice Portal Configure the Dial Plan pattern to route calls from Session Manager to Voice Portal. Expand Routing and select Dial Patterns. Click New. In the General section under Pattern, add the Application ID (the dial pattern used to route the call to the Voice Portal from where the prompts start to play). Under Min, enter the minimum number digits that must be dialed. Under Max, enter the maximum number digits that may be dialed. Select Emergency call if required. Under SIP Domain dropdown, select the SIP Domain. Under Notes, add a brief description. In the Originating Locations and Routing Policies section, click on Add. Select the Originating Location and Routing Policy. Select Commit to save. 54 of 71

55 5. Configure the Avaya Voice Portal Log in to the Web Administration page of Voice Portal with Administrative rights System Configuration Avaya Voice Portal Media Processing Platform Servers Under System Management, select MPP Servers, on the left hand side. Add the installed MPPs with details of their names, IP addresses, maximum simultaneous calls, etc. if they do not already exist in the Voice Portal Configuration. Refer to Voice Portal documentation for additional details in Section of 71

56 Put the MPPs in Service Under System Management, select MPP Manager, on the left hand side. This will list all the administered MPPs. To place a MPP in service, place a check mark in the box to the left of the Server Name and click on Start. Note: The MPPs may already be in service if the Voice Portal system is already operational. The screen shot below shows the MPPs are Online and are in Running State. 56 of 71

57 5.2. Add the Automated Speech Recognition Server (ASR) Under System Configuration, select Speech Servers then select the ASR tab and enter the required details such as typing a descriptive name in the Name field, selecting the appropriate Engine Type from the drop down menu, typing the IP address in the IP Address field, etc. Note: This may have already been accomplished if the Voice Portal System is in operation. Below is a screen shot of the ASR added. 57 of 71

58 5.3. Add the Text-To-Speech Server (TTS) Under System Configuration, select Speech Servers then select the TTS tab and enter the required details such as typing a descriptive name in the Name field, selecting the appropriate Engine Type from the drop down menu, typing the IP address in the IP Address field, etc. Note: This may have already been accomplished if the Voice Portal System is in operation. Below is a screen shot of the TTS added. 58 of 71

59 5.4. Add a SIP Connection for Avaya Aura Session Manager Under System Configuration, select VoIP Connections in the left pane. Select the SIP tab and click on Add. Add the SIP connection details which would contain details of the SM IP interface and SIP Domain. Set Enable appropriately. Note: This may have already been accomplished if the Voice Portal System is in operation. Below is a screen shot of the VOIP Connection being added with fields outlined in red. 59 of 71

60 6. Configure Avaya Intelligent Customer Routing This section provides the procedures for configuring Avaya Intelligent Customer Routing Avaya Voice Portal Administration Main Menu Configuration is accomplished by expanding ICR Configuration from the left pane. Select ICR Core then click Add. Enter a descriptive Name and the Host Address. Click Continue. 60 of 71

61 6.2. Business Hours and Holidays Configuration Configure the Business Hours and Holidays. Select Business Hours and Holidays Configuration from the left pane. Click Add in the Business Hours. Enter Business Hour Name and Description. Click Add to specify the Days using the dropdown menu and enter Start and Stop Time. Click Save. 61 of 71

62 From the Business Hours and Holidays page (shown at the beginning of this section) click Add in the Holidays. Enter a name for Holiday Group Name. Enter descriptive text for Holiday Group Description. Click Add in the Holiday Details and enter the specific information for the given holidays. Click Save. 62 of 71

63 6.3. Setup ICR Call Center Expand ICR Configuration in the left pane and select Call Center. The Call Center page is displayed. Click Add. Enter Name, SIP Domain, IP Address for the target Communicaton Manager. Set the Time Zone using the dropdown menu. Click Save when finished Define the ICR Configuration Skill Expand ICR Configuration in the left pane and select Skill. The Skill page is displayed. Click Add. Enter the Skill ID, Skill Name, Description and Default VDN. Use the dropdown menu to select Call Center and Agent Strategy. Modify Cache Freshness and Slack Time if warranted. Click Save. 63 of 71

64 The Skill page is displayed again. Click the far right column pencil icon to configure the VDN List for the Skill ID. Click Add. The Add VDN Configuration page is displayed. Enter Polling VDN, Queue VDN and Description. Use the dropdown menus to select Call Center, Business Group and Holiday Group. Click Save when finished. 64 of 71

65 6.5. Setup Self Service Applications Under System Configuration, select Applications. Click Add. Enter the Name and Enable. Use the dropdown menu to specify Type is ICR:CCA. Add the Application URL, the Speech Servers configured, and the DNIS number to access the application. Note that the applications listed below are only representative of the test lab configuration. Scroll down and expand Advanced Parameters. Set the highlighted parameters as shown. 65 of 71

66 Click Save. The Applications page is displayed. Click Managed Application pencil icon shown in the red circle below. 66 of 71

67 Select the appropriate options for the Default VDN and domain, SSA Application Detail, EHA Application Detail (Non Business Hrs), EHA Application Detail (Generic) and WTA Application Detail. Screenshot below shows input values via the Change ICR Applications page. 67 of 71

68 6.6. ICR Manager The ICR Manager is used to managed the ICR Core server. Select the checkbox for the selected ICR Core server. Using the Process Commands, click Start to initiate the ICR Core service. Using the Service Commands, click Start to initiate connectivity to the ICR Core service. Verify State is Running and Mode is Online. Screenshot below depicts a working ICR Core server. 68 of 71