CCNA Collaboration – Notes, Chapter 13

Cisco Unity Connection, Voice Messaging

Cisco Unity Connection (CUC) is a full featured voice messaging, auto-attendant and voice recognition system. CUC can support 20,000 mailboxes on a single cluster. Built in IMAP server allows email access to voice messages. Data and message store database are local to the server using Informix Database Service application.

CUC is installed as a VM guest on supported hardware. CUC 10.x is linux based OS.

CUC supports integrations with traditional PBX systems that support native IP or digital TDM circuit that can be connected via PBX IP Media Gateway (PIMG) or T1 IP Media Gateway (TIMG).

Users can be added manually, with BAT (CSV, imported from CUCM using Administrative XML Web Service (AXL) or synced directly from LDAP.

CUC integrates with MSoft Exchange using Web-Based Distributed Authoring and Versioning (WebDAV) providing calendar and journal information for integrations with MeetingPlace.

CUC provides traditional Telephone User Interface (TUI) for interaction with DTMF and Voice User Interface (VUI) for handsfree interaction and IP Phone View (Visual Voicemail).

CUC can be setup as redundant active/active pairs

Integrations

CUCM and CME are supported using SCCP or SIP

Voicemail Port Wizard in CUCM simplifies integration with CUC. The system generates voicemail ports in CUCM and adds them to a Line Group. Admins must manually configure the Hunt List and Hunt Pilot to support the Line Group.

The Hunt Pilot is referenced by the voicemail pilot

SCCP Integration with CUCM

  • Voicemail Profile
  • Voicemail Pilot
  • Hunt Pilot
  • Hunt List
  • Line Group
  • Voicemail Port 1 – X

In SCCP integrations the MWI (Message Waiting Indicator) is a unique DN for on and off. The DN’s must match in CUCM and CUC.

SIP Integration with CUCM

Instead of a Hunt Pilot and Voicemail Pilot, SIP points to a route pattern which points to a SIP trunk.

  • Voicemail Profile
  • Voicemail Pilot
  • Route Pattern
  • SIP Trunk

SIP is able to handle MWI within it’s signaling


CUC Features

General Configuration – System timezone, language, max greeting length

Roles 

  • Audio Text Admin – Manage call handlers, directory handlers and interview handlers
  • Audit Admin – Enable/disable application and database auditing, config audit settings, view or delete audit logs
  • Greeting Admin 
  • Help Desk admin – Reset user pins, unlock accounts and view user settings
  • Mailbox Access Delegate – Access to all messages, access on behalf of users
  • Remote Admin – Allowed to use remote tools
  • System Admin – Top level admin role
  • Technician – Access to all functions that enable management of CUC and phone system integrations
  • User Admin – Manage user accounts

Enterprise Parameters and Service Parameters

  • LDAP
  • Call Handlers
    • System Call Handlers – Greetings that can be customized to offer user input options and automation for playing different greeting depending on time of day
    • Directory Handler – Search CUC directory for a user they want to contact
    • Interview Handler – Record answers to questions
  • Call Routing
  • Direct Routing Rules
  • Forwarded Routing Rules
  • Distribution Lists
    • Send voice message to a group of users
  • Authentication Rules
  • Dial Plan
    • Concept of partitions and CSS

Users and Mailboxes

User Templates provide a way to create new user accounts. Template settings are applied as users are created. Changing the template does not go back and change existing users.

2 default templates exist: administrators and users

Template Basics:

  • Name – Name of the template
  • Phone – Dial plan, Class of Service and schedule
  • Location – Geographic location, language and timezone

Class of Service (CoS), different than L2 QOS CoS, method of assigning and restricting user privileges. Unlimited CoS’s can be defined

  • Class of Service
    • Timers
      • Recorded Name Length
      • Greeting Length
      • Message Length
    • Features
      • IMAP
      • Unified Personal Communicator
      • Personal Call Transfer Rules
    • Restrictions
      • Distribution Lists
      • Restriction Tables

End Users

New end users require little configuration because of templates. Unique settings: alias, name, mailbox store, extension and alternate extensions

Extension is required – should be primary users DN, it’s the caller ID CUC uses to determine the owner of the mailbox.

WAN Failures – If the users access CUC over the WAN and its down, SRST using AAR can allow calls to CUC to be routed over the PSTN, the users 10 digit PSTN caller ID will not be recognized unless the alternate extension is entered.

Voicemail Box- admin can choose to list a user in the directory, record the voice name and record a greeting

Private Distribution Lists – Each user can create 99 private DLs with a max of 999 members. The DL is only visible to the user that created it and the administrator of CUC.

Notification Devices – Users can be notified of a new message of up to 3 PSTN numbers and email

User Creation

  • Manual
  • BAT
  • Migration from Unity
  • Import from CUCM over AXL
  • LDAP

Voicemail Boxes

Associated with each user. Held in the database and synced between the 2 CUC servers in the active/active redundant pair. Admins can set message aging policies and move read messages to deleted items after a specified number of days (disabled by default). Messages in the deleted folder are deleted after 15 days by default.

Mailbox Size – Default 12MB, prevented from sending new messages at 13MB and cannot send or receive at 14MB.

12MB is approx 200 minutes of G.729 or 25 minute of G.711


Configuration

CUCM and CUC Integration

Start on CUCM

Don't forget to add an External Number Mask, as I did

Don’t forget to add an External Number Mask, as I did

I ended up going back into the config and deleted 249 voicemail ports to make the config simpler 

Create a new Hunt List

Add Line Group that was created in the Voicemail Integration Wizard to the Hunt List

Add Line Group that was created in the Voicemail Integration Wizard to the Hunt List

Create DN’s for MWI for SCCP integration – Advanced Features > Voice Mail > Message Waiting

Desc was changed to MWI On

Desc was changed to MWI On

Create Voicemail Pilot

CUCM Config is complete for integration. Moving to CUC, services have already been enabled

Clicked on default inside of Phone System, need to add a port group, choose in upper right corner

Clicked on default inside of Phone System, need to add a port group, choose in upper right corner

Add CUCM for AXL service for user data integration – Telephony Integration > Phone System

0 is highest preference 

0 is highest preference 

Edit Port Group servers

Do this if you have subscribers in the environment. My lab does not have any additional servers, so I will skip this step.

Do this if you have subscribers in the environment. My lab does not have any additional servers, so I will skip this step.


Config User Template

Change authentication rules as needed. Web is for Jabber

Change authentication rules as needed. Web is for Jabber

Add a user manually

CCNA Collaboration – Notes, Chapter 12

Enabling Mobility Features in CUCM

Mobility features allow users to interact with devices and applications no matter where they are. The intent is to extend users to use their enterprise phone number for both inbound and outbound calls seamlessly.

Mobile Connect

Mobile connect – Single Number Reach (SNR) allows the enterprise phone number to be a single number that will ring multiple other devices (cell phone, desk phone, etc.) This allows users to give out a single number that can reach the user regardless of location. If a user answers their enterprise phone from their cell phone they can move the call back to their desk phone or vice versa. If the call was answered on the desk phone, the mobility softkey can be pressed and the call seamlessly moved to the cell phone without dropping the call.

Mobile Connect is configured using Remote Destination Profiles to configure a virtual phone that share configuration settings with the users primary IP phone. Up to 10 remote destinations can be configured per user.

Access Lists can be configured (not related to router ACLs) to control which calls ring which rdp and time of day.

Mobile Voice Access (MVA)

MVA provides SNR consistency for outbound calls. Accessing CUCM from a cell the user can instruct CUCM to place a call as if it was from the users IP Phone. Users call into a specific PSTN DID to access the MVA service.


Configuring Mobile Connect (SNR)

  1. Configure softkey template to add Mobility key
  2. Configure user accounts for mobility
  3. Configure phone to support mobility features
  4. Create remote destination profile and assign to user
  5. Add remote destinations to remote destination profile
  6. Configure ring schedules for remote destination
  7. Configure access list and apply to remote destination
  8. Configure service parameters

Under the End User

Under the End User


Configure MVA

  1. Activate MVA service
  2. Configure Service Parameters
  3. Enable MVA for user
  4. Configure MVA media resources
  5. Configure MVA VXML application on Voice Gateway

Output below borrowed from CCNA Collaboration CICD 210-060 Official Certification Guide

https://www.safaribooksonline.com/library/view/ccna-collaboration-cicd/9780134171760/ch12lev3sec4.html

! Define the MVA Application and URL
application
 service mva http://10.1.1.1:8080/ccmivr/pages/IVRMainpage.vxml
dial-peer voice 50001 pots
! Associate the MVA application to this dial peer
 service mva
! Match the PSTN MVA access number to this inbound dial peer
 incoming called-number 4085555000
 direct-inward-dial
dial-peer voice 50002 voip
! Match the PSTN MVA access number to this outbound dial peer
 destination-pattern 4085555000
! Identify the CUCM server running the MVA service VXML app referenced above
 session target ipv4:10.1.1.1
 dtmf-relay h245-alphanumeric
codec g711ulaw
 no vad

CCNA Collaboration – Notes, Chapter 11

Enabling Telephony Features with CUCM

CUCM Extension Mobility

Allows users to log into any phone in the cluster. Used when users move desk to desk. Users personal configurations such as DN and speed dials follows the user to whatever phone they log into.

EM is an IP Phone service. It applies user specific device profiles to the phone after login. Administrators have 3 options to set system behavior if a user logs into multiple phones concurrently.

  • Allow multiple logins – User can be logged into multiple phones at the same time. Has a shared line effect. All phones will ring when the DN is called
  • Deny Login – Users can only log into one device at a time. Will receive error message, must log out of other phone first
  • Auto Logout – User can only be log into one device at a time. First device will be logged out of after successfully logging into second phone

When no user is logged into the phone a device profile can be applied (logout device profile).

  • MoH
  • Phone button template
  • Softkey template
  • user locale
  • DND
  • Privacy setting
  • Service subscriptions
  • Dialing name

Enabling Extension Mobility

  1. Activate EM service
  2. Configure EM service parameters
  3. Add EM service
  4. Create a default device profile for each phone model
  5. Create device profiles and subscribe to EM server
  6. Create end users and associate with device profile
  7. Enable EM for phones and subscribe phones to EM service

CUCM Telephony Features

Call Coverage – References features and mechanism used to ensure that calls are answered under most circumstances

  • Call Forward
    • Call Forward All (CFA) – Forward all calls to a destination number. Call Forward search space is used. Line and device search space are ignored
    • Call Forward Busy (CFB) Internal / External – When phone is offhook, calls to DN are forwarded to a specific voicemail pilot
    • Call Forward No Answer (CFNA) – Forwarded after Ring No Answer Reversion timer has expired, forward to voicemail pilot
    • Call Forward Unregistered (CFUR) – Used with SRST.
  • Shared Lines – 2 or more phones with the same DN configured on one of the lines. Calling the DN causes both phones to ring. First phone to pick up answers. 2nd phone cannot pickup without Barge feature
    • Barge – Force a 3 way conference with the first phone in a shared line scenario
      • Conference is hosted on the first phone
    • Privacy – Prevent barging into a call
  • Call Pickup
    • DN’s can be assigned to a call pickup group
    • Multiple DNs have the same group number and one of them is ringing another phone can answer by using the Call Pickup softkey
  • Call Hunting
    • Allows a single DN to distribute calls to several phones in sequence
    • Call Hunting Components
      • DN and Voicemail ports – Targets for the call hunting system
      • Line Groups – Assigned to hunt list, can be assigned to one or more hunt lists
      • Hunt Lists – Top/down ordered list of line groups
      • Hunt Pilots – Associated with hunt list. Unique DN, shared line or PSTN number
  • Call Park – Allows users to temporarily attach a call to a call park slot (DN). Any user can pickup a parked call by dialing the park DN.

Intercom

Allows a button to be configured that calls an intercom line on another phone. The receiving phone auto answers in speakerphone mode with the microphone muted.

Intercom lines cannot call DNs and DNs cannot call intercom lines. They have their own dial plan and permissions (Intercom partition and CSS).

Presence

Presence includes instant messaging status’s (online, offline, busy, in a call, etc) or in the phone system, onhook vs offhook.

Presence status can be monitored with the Busy Lamp Field (BLF) speed dial or preence-enabled call and directory lists.


Configuring Shared Lines

Associate a DN to more than 1 device

Associate a DN to more than 1 device


Configuring Barge

Service Paramters, choose the server select Cisco CallManager Service Scroll down to Clusterwide Parameters (Device-Phone)”/> System > Service Paramters, choose the server select Cisco CallManager Service Scroll down to Clusterwide Parameters (Device-Phone)

Scroll down to Clusterwide Parameters (Feature - Join Across Lines and Single Button Barge Feature Set

Scroll down to Clusterwide Parameters (Feature – Join Across Lines and Single Button Barge Feature Set


Configuring Call Pickup

Call Pickup Group”/> Call Routing > Call Pickup Group

Change the softkey layout to allow for Call Pickup Group

Device > Device Settings > Softkey Template

I copied a standard user and created a HELPDESK softkey template, added PickU Up from the left side

I copied a standard user and created a HELPDESK softkey template, added PickU Up from the left side

Add Call Pickup Group to DN

Add Call Pickup Group to DN


Configuring Call Park


Configuring Call Hunting

Pre-req – phones and DN’s already created

Create Line Group – Call Routing > Route/Hunt > Line Group

Create Hunt List – Call Routing > Route/Hunt > Hunt List

Create Hunt Pilot – Call Routing > Route/Hunt > Hunt Pilot


Configuring Intercom Features

  1. Call Routing > Intercom > Intercom Route Partition
  2. Call Routing > Intercom > Intercom Calling Search Space
  3. Call Routing > Intercom > Intercom Directory Number

Change device to add intercom button

Device > Device Settings > Phone Button Template


CCNA Collaboration – Notes, Chapter 10

Understanding CUCM Dial Plan Elements and Interactions


CUCM Call Flows

This chapter reviews:

  • Call signaling and voice traffic flow
  • Components of call routing
  • Call routing decision process
  • Component configuration
  • Redundancy
  • Restrictions

Call Flow – DNS

DNS is not recommended with IP phones

If DNS is used, the phone must complete a DNS name resolution to learn the IP address of CUCM before signaling can occur. This process introduces delay and also reliance on another system (DNS) that could break the call setup process.

After DNS has resolved the name of CUCM to an IP address the call flow is as follows

  1. SCCP or SIP signaling between phone and CUCM
  2. RTP (real-time transport protocol) carries voice phone to phone (CUCM is not in traffic path for voice)

Call Flow – No DNS

Removing DNS reliance (demonstrated in ch9 notes) is recommended in CUCM. This allows phones to use IP to reach CUCM. The call flow becomes simiplified as the DNS step described above is not done and normal traffic flow occurs.

Phones signal to CUCM with SIP or SCCP, CUCM setups call, phones talk directly using RTP


Centralized Remote Branch Call Flow

Centralized deployment – CUCM servers are located at a main location (companies DC) with remote sites connecting over the WAN for both signaling and on-net voice.

Off-net calls could be routed out a local gateway at the site (PSTN or POTs lines installed in voice gateway at the branch)

Signaling remains the same, SCCP or SIP signaling traffic is sent to CUCM, CUCM setups the call between the 2 phones and the voice (RTP) traffic flows directly from phone to phone. The phones can be located in different sites (Branch to Branch, Branch to HQ, etc).

PSTN Backup Call Flow

If WAN fails, phones can no longer register with CUCM and no longer function. In this scenario SRST is recommended to provide local phone registration in the event of a WAN failure.

Survivable Remote Site Telephony (SRST) is a feature that allows branch routers to take over phone registration and call control if phones cannot reach CUCM. SRST provides on-net calling between phones within the branch. If the SRST routers dial plan is configured properly the branch can dial on-net extensions at another site and SRST will modify the dialed digits for PSTN routing.

CUCM will see the phones as unregistered. Reaching the phones over the PSTN is possible if CUCM is setup with an alternate path.

  • Call routing table has 2nd option to provide PSTN gateway and digit manipulation for PSTN dialed digits
  • Call Forward UnRegistered (CFUR) – destination number that calls will be forwarded if the phone is unregistered with CUCM. Used in conjunction with SRST. This is configured for each branch phone to configure the full PSTN number to reach the branch phones

These configurations in CUCM along with the dial plan in SRST will allow the branch site to still be able to call between sites during a WAN failure. When the WAN recovers, phones register back with CUCM and normal call flow resumes.

Centralized Deployment Considerations

CUCM v10 supports a max of 2000 locations and a max of 2100 H.323 or MGCP gateways per cluster.

• H.323 – Protocol created by ITU-T to allow multimedia communication over network-based environments

• MGCP – Media Gateway Control Protocol – Voice signaling protocol created by IETF. Allows voice gateways to be controlled by a centralized call agent (client / server)

There is no limit of number of phones at a branch site, however the number of phones supported by SRST is limited based on the router hardware at the branch site.

WAN’s must be configured with QOS and allocate bandwidth in the priority queue for voice traffic based on the number of concurrent calls that will happen at the site.

Call Admission Control (CAC)

A technique for monitoring the total remaining bandwidth available for voice traffic over a WAN circuit. The purpose of CAC is to prevent voice traffic in excess of what the circuit can support without overflowing the QOS priority queue and causing voice traffic to be dropped. CAC can be implemented using Locations in CUCM (shown in last chapter). RSVP (Resource Reservation Protocol) can also be used, a QOS mechanism.

Locations – track how many calls are between given locations and subtract bandwidth for each concurrent call. If no bandwidth is available, the call is dropped (default CAC behavior). The user gets a reorder tone.

AAR – Automated Alternate Routing – allows calls that would be dropped by CAC to be rerouted over the PSTN. AAR is triggered by CAC when CAC prevents a call over the WAN. AAR requires digit manipulation to retry the call over the PSTN


Distributed Call Flow

Distributed deployments of CUCM, one CUCM cluster signals another CUCM cluster over the WAN. Signaling flows from calling phone to local CUCM and from local CUCM to remote CUCM over the WAN. RTP traffic is setup directly phone to phone across the WAN.

CUCM used the following signaling protocols between CUCM clusters

  • ICT – Inter-Cluster Trunk
  • H.323
  • SIP

CUCM Call Routing – Sources

  • Phone – places a routing request through a dialed number
  • Trunk – Signals inbound calls from another CUCM, CME or call agent
  • Gateway – Signals inbound calls from SPTN to another call agent
  • Translation Pattern – Matches originally dialed digits and transforms them into a new dial string
  • Voicemail Port – Can be source of a call routing request if the application attempts to call, transfer or message notification on behalf of a users mailbox

CUCM Call Routing – Destinations

  • Directory Number (DN) – Unique on-net extension that can be assigned to a button on an IP phone
  • Translation Pattern – Matches a dialed string and transforms them into a new dialed string. This new string is analyzed and routed to a different target
  • Route Pattern – Matches a set of dialed digits and triggers a call routing process that can include one or more potential paths. Hierarchical set of call routing options
  • Hunt Pilot – Specific pattern of digits that can trigger a customizable call coverage system
  • Call Park Number – A pattern or range or patterns that CUCM can use to temporarily hold a call until a user dials the call park number to pick up the call
  • Meet-Me Number – Conference call initiator dials into a Meet-Me number to begin a conference

All destinations are a string of digits or a SIP URI (Uniform Resource Identifier) 

SIP URI – alphanumeric string – 1-555-860-5555@voice.cmpnetworking.com


Call Routing Configuration

Components of CUCM call routing: route patterns, route lists, route groups, gateways/trunks

Route Pattern

  • Matches a string of dialed digits
  • Pattern may be specific matching a single dialable number or general and match hundreds/thousands or possible numbers
    • Wildcards are used in the pattern
  • Required to provide PSTN access
  • Can be used to integrate with existing PBX dial plans
  • Associated with a route list or gateway
    • If the route pattern is directly associated with a gateway, the gateway can no longer be referenced by a route group. Gateway is locked to the specific route pattern

Route List

  • Ordered list of route groups
  • First entry is the preferred call routing path
    • If unavailable the 2nd in list will be used
    • Each new call uses the top-down order
  • This allows admins to choose which circuits get used for which type of calls

Route Group

  • List of gateways or trunks that are configured to support circuits to PSTN or remote CUCM clusters
  • Commonly configured to contain devices with similar signaling characteristics 
  • Distribution of the calls is configurable: top/down, circular

Gateways and Trunks

  • Physically terminate and support circuits to PSTN, digital/analog PBX and WAN circuits to remote clusters or IP-TSP circuits to service provider
  • CUCM supports peer to peer gateway protocols – H.323 and SIP
  • CUCM supports gateway control protocols – MGCP and SCCP

Call Routing Behavior

Dial analysis is performed by CUCM by matching dialed digits.

  • SCCP – Digits are collected digit by digit
  • SIP – Keypad Markup Language (KPML) and en-bloc (all at once as a set of digits)

Digit Analysis

CUCM selects a destination for the call routing request based on closest match

T.302 – Wait time, inter-digit timeout – default is 15 seconds. To wait for any additional digits to be dialed. After the timer is finished the call is routed.

Digit by digit analysis means CUCM collects digits one at a time as they are dialed. When collected patterns that no longer match as discarded as routing targets.

Hunt Groups

A hunt group is a set of phones (DN’s) that are reachable by calling a common number.

  • Line Group – contains DN’s that will ring sequentially. Allows for call distribution: top/down, circular, longest idle, broadcast
  • Hunt List – Contains top/down ordered list of line groups. Each call is routed to the first line in the list unless it is busy then the 2nd line will be rung. If the group is busy the next group in the hunt list will be used
  • Hunt Pilot – Matches a dialed string and targets a hunt list (call routing entry). Hunt pilot numbers can be on-net, E.164 or any format required.

Class of Control

Class of Control defines the ability to apply calling restrictions to a device. Configured using partitions and calling search spaces (CSS).

  • Prevent a phone from calling long distance
  • Routing the same called number to different targets depending on the time of day
  • Routing the same called number to different targets at different locations

Partition

Grouping of things with similar reachability characteristics. Assigned to things that get dialed.

By Default – one partition exists, null partition

75 additional partitions can be created

  • DN
  • Route Pattern
  • Translation Pattern
  • Voicemail Ports
  • Meet-Me Conference

Calling Search Space (CSS)

Top/down ordered list of partitions. Can be applied to device (phone or gateway) or to line on the phone

One CSS exists by default, contains null partition.

CSS’s are applied to things that make calls

Partitions and CSS

If the target dialed number does not exist in one of the partitions in the CSS, the call will fail

When a route pattern is moved from default partition, it is no longer available to the default CSS

Every CSS includes the default partition and the end of the list.

If both device and line CSS are applied, the partitions in both CSS are concatenated in sequential top/down order. The LINE CSS partitions are listed first followed by the device CSS partitions.

Line CSS overrides the device CSS

Best practice – Setup device CSS to allow full calling privilege to all patterns based on the devices location. The calling restrictions are applied using the line CSS which contain route patterns that match long distance but configured to block the call