Archive for the ‘CCIE VOICE’ Category

Collaboration update information videos

CCIE Collaboration Lab

February 20, 2014 Leave a comment

Lab Equipment:

  • Cisco Unified Computing System (UCS) C460 Rack Server
  • Cisco 3925 Integrated Services Routers Generation 2 (ISR G2)
  • Cisco 2921 Integrated Services Routers Generation 2 (ISR G2)
  • ISR G2 Modules and Interface Cards
    • - 1-Port 3rd Gen Multiflex Trunk Voice/WAN Int. Cards – T1/E1
    • - Cisco High-Density Packet Voice Digital Signal Processor Modules (PVDM3)
    • - Cisco Service Ready Module 710 Service Module with Cisco Unity Express
    • - 4-port Cisco Gigabit EtherSwitch 10/100/1000BASE-TX autosensing EHWIC with POE
  • Cisco Catalyst 3750-X Series Switch
  • Cisco Unified IP Phones 7965 and 9971
  • Cisco Jabber for Windows
  • Cisco Jabber Video for Cisco Telepresence*

*In backbone, no candidate configuration required

Software Versions
Any major software release which has been generally available for six months is eligible for testing in the CCIE Collaboration Lab Exam.

  • Cisco Unified Communications Manager 9.1
  • Cisco Unified Communications Manager Express 9.1
  • Cisco Unified Contact Center Express 9.0
  • Cisco Unified Communications Manager IM and Presence 9.1
  • Cisco Unity Connection 9.1
  • Cisco Unity Express 8.6
  • All routers use IOS version 15.2(4) M Train
  • Cisco Catalyst 3750 Series Switches uses 15.0(2) Main Train

Network Interfaces

  • Fast Ethernet
  • Frame Relay

Telephony Interfaces

  • T1/E1

Cisco Press Resources

 CCIE Collaboration Reading List:

     Cisco IP Telephony (CIPT) (Authorized Self-Study Guide), 2nd Edition

     Cisco Voice Gateways and Gatekeepers

     Fax, Modem, and Text for IP Telephony

     SIP Trunking

     Cisco Unified Presence Fundamentals

     Cisco Unity Connection

     Securing Cisco IP Telephony Networks

     Troubleshooting Cisco IP Telephony

Cisco Solutions Reference Network Design (SRND) and Design Guides

     Cisco Unified Communications System 9.x SRND

     Cisco Unified Contact Center Express Solution Reference Network Design Release 9.0(1)

     Design Guide for Cisco Unity Connection Release 9.x

     Cisco Unified CallManager Express Solution Reference Network Design Guide

Administration and Deployment Guides

     Cisco Unified Communications Manager Documentation Guide, Release 9.0(1)

     Cisco Unified CCX Administration Guide, Release 9.0(1)

     System Administration Guide for Cisco Unity Connection Release 9.x

     Deployment Guide for IM and Presence Service on Cisco Unified Communications Manager, Release 9.0(1)

Cisco Unified Communications Manager Express System Administrator Guide

Coming soon in Bsoft Bangalore 

#30, 1st floor,1st Main,BTM 2nd stage
Bennergetta Road,Bangalore,India

Mobile: 09886623909
Phone : 080 4146 5262

Categories: CCIE VOICE

CCIE Collaboration | 400-051

Cisco Collaboration Infrastructure

  • UC Deployment Models
  • User Management
  • IP Routing in Cisco Collaboration Solutions
  • Virtualization in Cisco Collaboration Solutions
    • UCS
    • VMware
    • Answer Files
  • Wireless in Cisco Collaboration Solutions
  • Network Services
    • DNS
    • DHCP
    • TFTP
    • NTP
    • CDP/LLDP
  • Power over Ethernet
  • Voice and Data VLAN
  • IP Multicast
  • IPv6

Telephony Standards and Protocols

  • SCCP
    • Call Flows
    • Call States
    • Endpoints types
  • MGCP
    • Call Flows
    • Call States
    • Endpoints types
  • SIP
    • Call Flows
    • Call States
    • SDP
    • BFCP
  • H323 and RAS
    • Call Flows
    • Call States
    • Gatekeeper
    • H.239
  • Voice and Video CODECs
    • H.264
    • ILBC
    • ISAC
    • LATM
    • G.722
    • Wide band

Cisco Unified Communications Manager

  • Device Registration and Redundancy
  • Device Settings
  • Codec Selection
  • Call Features
    • Call Park
    • Call Pickup
    • BLF Speed Dials
    • Native Call Queuing
    • Call Hunting
    • Meet-Me
  • Dial Plan
    • Globalized Call Routing
    • Local Route Group
    • Time of Day Routing
    • Application Dial Rules
    • Digit Manipulations
  • Media Resources
    • TRP
    • MoH
    • CFB
    • Transcoder/MTP
    • Annunciator
    • MRG/MRGL
  • CUCM Mobility
    • EM/EMCC
    • Device Mobility
    • Mobile Connect
    • MVA
  • CUCM Serviceability and OS Administration
    • Database Replication
    • CDR
    • Service Activation
    • CMR
  • CUCM Disaster Recovery
  • ILS/URI Dialing
    • Directory URI
    • ISL Topology
    • Blended Addressing
  • Call Admission Control
    • RSVP
    • SIP Pre-conditions
  • SIP and H323 Trunks
    • SIP Trunks
    • H.323 Trunks
    • Number Presentation and Manipulation
  • SAF and CCD
  • Call Recording and Silent Monitoring

Cisco IOS UC Applications and Features

    • SCCP Phones Registration
    • SIP Phones Registration
    • SNR
  • SRST
    • CME-as-SRST
    • MGCP Fallback
    • MMOH in SRST
  • CUE
    • AA
    • Scripting
    • Voiceview
    • Web Inbox
    • MWI
    • VPIM
  • IOS Based Call Queuing
    • B-ACD
    • Voice huntgroups
    • Call Blast
  • IOS Media Resources
    • Conferencing
    • Transcoding
    • DSP Management
  • CUBE
    • Mid-call signaling
    • SIP profiles
    • Early/Delayed offer
    • DTMF interworking
    • Box-to-Box Failover/Redundancy
  • Fax and Modem Protocols
  • Analog Telephony Signaling
    • Analog Telephony Signaling Theories (FXS/FXO)
    • Caller ID
    • Line Voltage Detection
    • THL Sweep
    • FXO Disconnect
    • Echo
  • Digital Telephony Signaling
    • Digital Telephony Signaling Theories (T1/E1, BRI/PRI/CAS)
    • Q.921 and Q.931
    • QSIG
    • Caller ID
    • R2
    • NFAS
  • IOS Dial-Plan
    • Translation Profile
    • Dial-peer matching logics
    • Test Commands
  • IOS Accounting

Quality of Service and Security in Cisco Collaboration Solutions

  • QoS: Link Efficiency (e.g. LFI, MLPPP, FRF.12, cRTP, VAD)
  • QoS: Classification and Marking
    • Voice vs Video Classification
    • Soft Clients vs Hard Clients
    • Trust Boundaries
  • QoS: Congestion Management
    • Layer 2 Priorities
    • Low Latency Queue
    • Traffic Policing and Shaping
  • QoS: Medianet
  • QoS: Wiress QoS
  • Security: Mixed Mode Cluster
  • Security: Secured Phone Connectivity
    • VPN Phones
    • Phone Proxy
    • TLS Proxy
    • 802.1x
  • Security: Default Security Features
  • Security: Firewall Traversal
  • Security: Toll Fraud

Cisco Unity Connection

  • CUCM and CUCME Integration
  • Single Inbox
  • MWI
  • Call Handlers
  • CUC Dial-plan
  • Directory Handlers
  • CUC Features
    • High Availability
    • Visual Voicemail
    • Voicemail for Jabber
  • Voicemail Networking

Cisco Unified Contact Center Express (UCCX)

  • UCCX CTI Integration
  • ICD Functions
  • UCCX Scripting Components

Cisco Unified IM Presence

  • Cisco Unified IM Presence Components
  • CUCM Integration
  • Cisco Jabber
  • Federation
  • Presence Cloud Solutions
  • Group chat and Compliance

Cisco Press Resources

Cisco Solutions Reference Network Design (SRND) and Design Guides

Administration and Deployment Guides


All about SIP

Session Initiation Protocol (SIP)
SIP is a signaling protocol widely used for voice and video.
Traditionally if your enterprise wants to connect to PSTN could, you would need BRIs, PRIs, PSTN gateways.
With SIP Trunking service you can avoid all the Network PSTN connections.

What is SIP Trunk:
Internet telephony service provider (ITSP) offer SIP trunk as a service for communication between enterprise PBX and PSTN.
IP Phone——-CUCM———–SIP Trunk————-ITSP——–PSTN
Each SIP trunk can have multiple voice session based on enterprise needs.
G.711 – 17 calls over T1
G.729a – 45 calls over T1
SIP trunk is not limited to voice, it can also help enterprise to setup instant messaging, real-time presence, video, etc.
For more details refer to RFC 3261

Benefits of having SIP Trunk:

  1. Like I said no need to invest in PSTN gateway and voice cards.
  2. Off course low cost compared to traditional PRI T1.
  3. Reach out to the world on cost of local call.
  4. Easy installation and maintenance.
  5. Optimal use of bandwidth as data and voice run on same connection.
  6. Move away from T1/E1 capacity limitations of 23/30 channels.
  7. SIP trunk normalization and transparency
  8. Up to 16 destination IP addresses per trunk
  9. Automated hunting from a primary PBX to a secondary
  10. Automated failover to PSTN with full trunk

For you to deploy SIP in your enterprise there are two things which needs to be there.

  1. SIP enabled PBX
  2. SIP enabled edge device

Bandwidth Utilization:
It is always good idea to reserve 27 Kbps with G.729  per call and 84Kbps for G.711

Minimum Bandwidth Requirement
Codec Voice Bit Rate
G.711 64 Kbps
G.729 8 Kbps


Cisco Recommendations:

Application H.323 MGCP SIP Preferred
Voice Mail Control of Individual Ports MGCP
Configuration Dial Peer Based Centralized in CUCM Dial Peer Based MGCP
Load on CUCM Least MGCP
Q.SIG Tunneling Only between PBXs Supported MGCP
Video H.320 ISDN H.320 ISDN H323/SIP
Fax & Modem Pass-through, T.38 Pass-through, T.38 Pass-through, T.38 H323/SIP/MGCP
Port Density High Density Cards MGCP
Redundancy Range of options with dial-peers Range of options with dial-peers H.323/Sip
Security IPSec and SRTP IPSec and SRTP TLS and SRTP SIP
Voice XML Supported Supported H.323/SIP


 Centralized SIP Trunk Design Limitations

MoH Centralized MoH Limited to 50
Central Site Device Pool Multiple Device Pools for Devices on Datacenters
Non Ported DIDs Requires a Different Call Flow and Different Call Routing
FAX Not Supported on SIP Trunk; Handled by Site GW
SRST Limited Access via FXO, PRI for Medium/Large Site
DTMF SIP Trunk and Check IP Phone

Basic SIP Trunk configuration in CUCM

  1. Step 1:
    1. Create a SIP profile (optional).
    2. Create a SIP trunk security profile (optional)
    3. Create a SIP trunk.
    4. Configure the destination address.
    5. Configure the destination port.
  2. Step2: Associate the SIP trunk to a Route Pattern or Route Group.
  3. Step3: Configure SIP timers, counters, and service parameters, if necessary.
  4. Step4: Reset the SIP trunk

Common SIP Requests

  1. REGISTER – to register a phone or line with a SIP Server
  2. INVITE – to set-up a call
  3. CANCEL- to cancel a call set-up
  4. BYE – to terminate a call

Common SIP Responses

  1. 100 trying
  2. 180 ringing
  3. 200 OK
  4. 401 not authorized
  5. 404 destination not found
  6. 486 busy

SIP Trunks vs. H.323 Trunks (Inter cluster)

H.323 SIP
Annex M1 Features / Q.SIG Tunneling NO NO
Signal Authentication No YES
Media Encryption YES YES
GK Support YES NO
SIP Proxy Support NO YES
iLBC and G.Clear Support No YES
G.722 Support YES YES
Multicast MoH YES YES
SIP Subscribe/Notify, Publish-Presence NO YES
Path Replacement NO NO
Call Completion to Busy Subscriber NO NO
Call Completion No Reply NO NO
Message Waiting Indicator (On/ Off) No YES
Alerting Name NO YES


SIP Trunking byChristina Hattingh, Darryl Sladden, ATM Zakaria Swapan Published by Cisco Press.

Cisco Unified Communications System 8.x SRND
BRKUCC-2006 – SIP trunk design and deployment
BRKCCT-2030 – SIP based Architectures for Cisco Contact Center Solutions & Collaboration
BRKUCC-2735 – SIP Trunk Design and Deployment Playbook for the Enterprise
BRKUCC-2450 – Planning for SIP trunking and dial plan centralization with SME

Categories: CCIE VOICE, RealTime DB, Troubleshoot Tags:

All about UC 500

February 21, 2013 Leave a comment

Technical Enablement Collateral Helps Partners


This document includes:

  1. Links to documents and Materials you need to be famaliar with
  2. Technical Enablement Labs that show you how to configure basic functions
  3. TEL Template

This is a community document open for editing as you see fit.


Links to Documents


For any Partner who wants to build a successful practice deploying UC 500 (the new UC 540/UC 560 or the legacy UC 520), there are certain materials you need to become famaliar with and as well some actions you need to take to have your Technicians and Engineers feel comfortable building and deploying UC 500.


Small Business University
Select Certification <– for both AM and SE


First Look SBCS UC540 2.0 Lab
Distance Learning Opportunity for Channel Partners <– for both AM and SE


Smart Designs
Design and implementation notes to show typical deployments <— for the SE


Integrating the UC 500 System with an Existing Customer’s Firewall


SBCS Feature Reference Guide (CCA configurable)

No longer requires login; accessible to partners and customers. <– for both AM and SE


Administration guide for CCA



The CCA Release Notes give you additional information about whats new as well as any known issues with each release

The CCA Prerequisite Checks must be performed on the PC you plan to run CCA on and will save you days of wasted time



Platform Reference Guides <– for the SE





make sure you understand the SKUs supported and the capacity of the system as defined in these great guides


Quick Pricing Tool (QPT) <— for the AM and SE (reduce Quoting time from Days to Minutes)



Use the Site Survey <— for AM & SE

Cisco Steps to Success for VoIP – UC500 SMB Version

Collect what you need to build the system (much of it in a staging environment) before you deploy on site.


SBSC System Test information (how it was tested, which code was used, what the results were)


Third Party Applications to be understood:



In addition to emmedded applications like SNR, WEBEX, TCV, IMAP Mail Integration, Video telephone, learn about third party integrations


Possible Milestones you can achieve as a measure of your ability to deploy.

  • Select Certification and identification of key individuals in for roles and responsibilities
  • get NFR Gear (UC500, ESW switches, Phones, AP541s, which you can demo to your customers)
  • Training touches – attend the virtual trainings and perform the TELs below in this document
  • Build your lab
  • Perform Demos


SBCS User Guides (including phones):


Get the Latest CCA Code:

Let the Latest UC500 Bundle:

Please take a look at the UC 500 Software Pack Roadmap posted at

Office Manager (OM) Information and SW

Office manager Installation Guide



Technical Enablement Labs (TEL) and Videos (VOD)

As SBCS moves forward, it does so with the UC 500 SWP, a given CCA version and the Office Manager (OM) version that work together to deliver new features and functions. This TEL lab area is updated by partner demand on the support community, many times to answer discussion threads that we feel should be answered more completely with lab demonstration. While we try to obsolete older labs to the bottom, there are still some older labs not yet replaced and so you should always consult the CCA OLH and Admin Guide as well as latest Smart Designs (which normally lag behind TELs but become more officially supported docs).


When we Announced CCA 3.0 and Office Manager 1.2, we delivered two Partner Webcasts and the .ppt used are here:


CCA 3.0 (new capabilities) Demonstrated (Webcast ppt from 1/13/2011)

Office Manager 1.2 technical Webcast/demo for partners 1/20/2011


Initial Maintenance and Configuration with CCA

Creating and Managing Customer Sites with CCA TEL


SW Upgrade: SWP 8.1.0 with CCA 3.0 TEL


Configuring a UC500 Demo System with CCA 3.0 TSW and Bulk User and Phone Import


UC 500 Localization using CCA 3.0 SW Upgrade Utility VOD


TSW for Bulk User and Phone Data Import using BulkDataImport.xls in CCA 3.0 VOD


Upload and Download files to UC 500 Flash using CCA 3.0 VOD


Upgrading or Installing a New License on a UC540/560 using CCA 3.0 TEL


Adding a VLAN to UC 500 TEL


Features and Functions Labs and VODs


The Remote Teleworker Router Configuration with CCASPA525G_SSL_Remote_TEL_19.pdf


SPA525G Remote SSL VPN Teleworker to UC 500


Users And Extensions Assignment in CCA 3.0 with Hold Alert and Speed Dial TEL Connectivity Status Diagnostics and Recovery with CCA VOD


Multiple WAN IP to LAN Static NAT Mappings with CCA 3.0 VOD


CCA Tele-worker Phone Support MTP Configuration VOD


Live Record Configuration and Beep Adjustments with CCA 3.0 TEL


Removing Live Record Beep Tones with CCA 3.0 VOD


Configuring Combined Paging Groups with CCA TEL


Paging Group Configuration with CCA VOD


SKYPE for SIP Configuration with CCA TEL


Configuring Remote Access to Administraton via Telephoine (AvT) for your AA with CCA TEL


Auto Attendant Script Customization (Alternate Greeting and Dial by First Name) with CCA TEL


Assigning FXO ports as “CO Lines” and assigning them to phones


Manual Override to AA TOD Schedule using Night Service


Configuring the UC500 to work with Call Accounting and Call Recording from ISI


SA500 Configuration with CCA TEL


Configuring and Operating FXO Trunk Groups with CCA TEL


Hybrid Key System Configuration with CCA


Webex Phone Connect configuration with CCA 3.0


Lab 3: UC540 EZVPN Configuration

Lab 4: SIP Trunk Configuration

Lab 6: SA500 Security Applicance in front of a UC500 with SIP trunking

Lab 7: B-ACD (Basic ACD) Configuration

Lab 8: Auto Attendant and Night Service

Lab 9: Multisite using CCA 2.1 Multisite Manager

Lab 10: Protect Link Gateway Service on SA 500 in front of UC540

Lab 11: Smart Application: TimeCardView

Lab 12: Smart Application – IMAP Unified Messaging

Lab 14: Unified CallConnector (UCC)

Lab K: Traffic Shaping and Max Connections (CAC),

Lab 15: SR520-T1 Configuration with CCA

Lab 16: Video Monitoring xVC2300 IP Cameras from SPA525G Phone

Lab 17: Smart Call Connector Attendant Console Operator

Lab 22: Disable Cisco IP Phone WEB Server access on UC500

LAB 24: NTP on SBCS (UC500 & SBCS)

Lab 25: TheOffice Demo configured using CCA 3.0 (Updated)

How to DEMO the OFFICE SBCS – Demo Guide

The actual DEMO Performed (54 minutes run time)

Lab 26: SNR Smart Application Configuration and Operation on UC500 with CCA

Lab 28: UC560 Voice Mail Expansion using upgrade of external CUE Flash to 8G

Lab 30  UC500  Access List Manager – An example on how to restrict Guest Network access

NOW OBSOLETE Labs Archive (based on older SWP and CCA version)

LAB 1: UC540 Initial Connection, Discovery, SW Upgrade, Phone Load Management, and Licensing

Lab 13: Smart Application – Live Record

Lab 20: Cisco Phone Firmware Upgrade (79xx models – manual)

Lab 23: Paging Groups on UC500

Lab 18: Auto Attendant custom script configuration for ‘no action transfer

Lab 21: SMTP Notification of UC500 Voice Mail to EMAIL

Lab 27: Configuring Alternate Greeting on UC500 Auto Attendant

Lab 2: UC540 Telephony Setup Wizard in Staging (staging without phones)

Lab 5: UC540 FXO PSTN Trunk Group Configuration

CCM 9.X New Features

February 18, 2013 Leave a comment

Pause in Speed Dial
Users can configure speed dials with FAC, CMC and post connect DTMF
Comma accepted in speed dial as delimiter and pause

Feature allows two methods of configuration:
-Method 1: Using comma as a pause and also as a delimiter
-Method 2: Dialstring/FAC/CMC/Post connect digits with no commas
Method 1: Command Delimiter for Pause
-Comma used to delineate dial string, FAC, CMC, and post connect digits
-For post connect digits, commas insert a 2 second delay
-Commas may be duplicated to create longer delays
-Preferred method for non-CUPC devices
Method 2: No Comma
-All digits to be used for dial string, FAC, CMC and post call digits entered as one string
-Once a digit string has been matched, CUCM moves on to next digit string
-Can be used on SCCP and SIP phones, but required for CUPC
Pause in Speed Dial Examples
-Will dial 914085551212, after connect, wait 8 seconds to dial 123456
-FAC for International Calls. Will dial 90114455612323# with FAC of 2244
-Will dial 91408551212, with a FAC of 6534 and CMC of 5656, wait 6 seconds, the dial the DTMF digits 9933
-Will dial 914085551212 with a FAC of 6534 and CMC of 5656, then immediately after connect, dial 9933
New Service Parameter allows configuration of interdigit delay
If the speed dial FAC or CMC is wrong
-Method 1: Call disconnects and an error is displayed
-Method 2: phone displays an error and allows user to manually enter information
Pause in Speed Dial Caveats
-Dial string is truncated in the calls history list (only dialed number)
-Feature may not work with CUPC client and variable length/overlapping dialplans (no comma delineation)
-This feature is not supported SRST

Codec Preference
Pre CUCM 9.0
-Administrator could only eliminate codecs (based on Maximum Audio Bit Rate)
-Could not prioritize G.711alaw over G.711ulaw, or G.729 codecs
With CUCM 9.0
-System default codec preference same as earlier versions
-Allow administrator to deterministically specify codec order
-Allow codec selection based on received offer
-Custom Codec list applied globally or on a GW/Trunk Level
-Can be applied to: SIP, MGCP, SCCP, H323 and EMCC

Codecs preference still choose by Regions

For SIP Devices/Trunk, can specify “Accept Codec Preference in received Offer”
Can change codec selection for EMCC logged in devices
Codec Preference Caveats
A common Codec Preference List must be the same on all clusters when using the following features:
-Extension Mobility Cross Cluster
-H323 Inter Cluster Trunks

Biggest challenge will be unexpected codec

-Check “Accept Audio Codec Preferences in Received Offer” settings
-Check at Device level and system level
When using non-pass through MTP, codec negotiated hop-by-hop

Native Call Queuing

Enables Hunt Pilot to queue callers

-Allow for redirection of calls based on different queue criteria
-Allow agents to participate in multiple queues
– Auto logout and call re-queue if agent does not answer
– Longest waiting call in all queues will be delivered first
– No ‘post call’ time or agent greeting options
– On phone ‘Queue Status’ display

Cisco Extend and Connect

What is the existing limitation?
– Using CTI (webex connect or CUCILync), user can monitor a calls, but not control the call
– No enterprise features for non-CUCM registered devices
– Cannot hold/resume, transfer, conference or park
– Remote devices ring and can be answered, but not mid-call features
What is Cisco Extend and Connect?
– A new device type, CTI Remote Device that represents all remote destinations for a user
– Anchors enterprise calls on the CTI Remote Device
– Allows a CTI application (like Jabber) 3rd party control of the remote connection to enable enterprise call features
Examples of a deployment scenario
Contact Center agent working from home
– Low bandwidth at house, VOIP not an option (hard phone or soft client) and cell phone is not an option
– Extend connect sends call to home phone and CAD agent allows enterprise features needed for contact center agents
Use Cisco Unified Communications with legacy PBX
– Customer has PBX under contract and not ready to move phones
– Customer wants UC for IM, Chat and messaging, but phones on PBX
– Extend Connect enables Jabber deployment for UC, but enterprise control of PBX phone (as remote device for Jabber)

New End User Webpages

CUCM 9.0 now has two types of end-user’s webpages
– One type of page is for core Users with one phone and one line
– The other page will be for users with multiple phones with one or more lines on each device
New User Page UI targeted towards core users
Cisco Mobility Updates
Simultaneous Ring in previous versions of CUCM
– CUCM 7.0 introduced the parameter “Reroute Remote Destination Calls to Enterprise Number”
– Calls direct to cell would ignore time of day settings and call the cell
– Calls would be anchor on the enterprise phone….but the line would not ring
New features in CUCM 9.0:
– Added “Ring All Shared Lines” service parameter
– Uses Boolean Setting
– True – all lines (including other remote destinations) ring
– False – only the dialed number (remote destination) rings
– Default and existing behavior is False

Single Number Reach Voicemail
The Problem:
– When a call is extended to a SNR destination, CUCM cannot determine if the call was answered by the user or VM
– Based on “Answer Too Soon”
– Time based mechanism is unreliable and requires tweaking for each service provider
New Solution
– CUCM 9.0 introduces a new parameter called “Single Number Reach Voicemail Policy”
– Can be either Timer Controlled or User Controlled
– Timer Controlled uses existing “Answer Too Soon” timer
– User Controlled requires the user to send a signal (DTMF) to accept the call

Hunt Pilot Connected Number Display
Hunt pilot DN display in previous versions
-Calls to a hunt pilot display the DN of the hunt pilot as the connected party ID
– Applies to both MGCP and SIP trunks
Hunt pilot DN display in CUCM 9.0
– This feature allows the connection to be updated with the answering party’s DN as the Called Party ID
– Applied on the Hunt Pilot Configuration page
– SIP: PAI and Remote PartyID are updated
– MGCP/H323: Connected Number sent to update the Called Party ID

RTCP Support
– RTCP provides out-of-band statistics and control info for RTP
– RTP sent on even port and RTCP is send over next higher odd port
– RTCP is supported between phones directly
RTCP not supported by:
– Trusted Relay Point (TRP)
– RSVP Agent
– DTMF Translator
– Passthru MTP

CUCM 9.0 RTCP New features:
– CUCM 9.0 supports RTCP through MTP in pass thru mode
– In non-pass thru mode, RTCP will still be blocked
– Only valid for SIP to SIP calls

BRI G.Clear
– CUCM v7.0 (1) first introduced G.Clear support for MGCP PRI
– G.Clear required for tandem ISDN bearer circuits in VOIP network
New features:
– CUCM 9.0 expands support for G.Clear to BRI interfaces
– Supported on MGCP BRI interface
– Supports G.Clear over SIP trunk with Early Offer and G.Clear

Security and OS Updates
– Red Hat Enterprise Linux 5.0 v7.0.2
– Host rename/reIP simplified (3 less steps to complete)
Optimized CLI commands:
– Utils dbreplication stop/dropadmindb/reset
– Utils dbreplication forcedatasyncsub
– Utils dbreplication status replicate
– Utils dbreplication runtimestate
Upgrade paths
– L2 upgrade from 8.6(1) and later to 9.0(1)
– Refresh Upgrade for 8.x (prior to 8.5), 7.1(5) and 6.1(5)

Security Feature Update
CTL Client Update
– Single installer for all Windows versions
– Supports Windows 7 (32 and 64 bit), Windows XP and Windows Vista

Updates to AXIS 2.0 (support .NET clients)

Assured Services for SIP Line side devices
– MLPP support for 99xx/89xx SIP phones and 3rd party SIP Phone
– TLS connections for 3rd party SIP phones

LDAP Enhancements
Custom User Fields
– Existing LDAP agreements sync 13 default attributes
– LDAP agreements will allow 5 Custom User fields
– Custom User Fields are common across all sync agreements
– Custom User Fields updated on 1 agreement are synched across all agreements
– Attribute will be validated at save time
– Error message thrown when saving and the attribute does not exist

LDAP and Manual User Support
Prior to CUCM 9.0
– Enabling LDAP sync would prohibit adding local users
– End user to be used by CUCM must be defined on AD and synched
– Extra users could trigger extra CAL’s on the MS AD
With CUCM 9.0
– Administrator can have both LDAP sync users and locally defined users
– Ability to modify local users and roles assigned to LDAP users
– Deleting LDAP synch will mark users synced for deletion (garbage collection)
– Administrator can convert an LDAP user to a local user
– User status field is used to differentiate between the Local user and LDAP Synchronized users
To convert LDAP synchronized user to the local user account:

- Check the box Convert User Account and Save changes
– After a user is converted to local CUCM user all the fields become editable

CUCM IM and Presence

Beginning with release 9.0, CUCM and CUP will start integration to be one product
– Includes common release and upgrade process
– Centralize administration
– Simplify licensing, now included as part of CUCM user licensing
– Deprecating IP Phone Messenger (IPPM) and CUPC 7.0
Through CUCM IM and Presence administration screens, configure UC Services for clients

UC Services that can be defined:
– Voice Mail
– Visual Voice Mail
– Conferencing
– Directory
– IM
– Presence

UC Services are used to build a UC Service Profile

UC Service Profiles assigned to users:
– Licensing for the feature handled at the user level
– Home cluster specified in the user page

When migrating to CUCM 9.0, existing service profiles and configuration in CUP will be migrated
– CUCM IM and Presence uses Templates and Layouts to speed up user creation
– BAT/AXL have been updated for CUCM/CUCM IM and Presence


Cisco Presence Integration with CCM 7.x,8.x

December 22, 2012 1 comment

Cisco Presence Integration with CCM

Summary steps of integrating Cisco call manager with Cisco presence

Step#1: Enable presence globally on Cisco Call manager

By default presence subscription is disabling on CCM.

System>Service parameter>Cisco Call Manager>

Search for “Inter-presence” key word and set “Allow Subscription”

Step#2: Create SIP trunk Security Profile in CCM

Special setting is required for SIP trunk which runs from CCM to Presence.

Copy “non Secure SIP Trunk Profile” to “Presence non-secure SIP trunk Profile”

Modify below parameters:

  1. Device security mode: Non-Secure
  2. Incoming Transport type: TCP+UDP
  3. Outgoing Transport Type: TCP
  4. IncomingPort 5060 (untick Enable digest authentication)
  5. Enable application Level Authentication UNTICK
  6. Accept Presence Subscription TICK
  7. Accept Out-of-Dialogue REFER TICK
  8. Accept Unsoliciliated Notification TICK
  9. Accept Replace header TICK
  10. Transforms security status UNTICK
  11. Save

Step#3: Add a SIP trunk now from CCM to Presence


Protocol = SIP

Fill below:

  1. Device Name : PRESENCE-TRUNK
  2. Description : blah
  3. Device Pool : DP_HQ
  4. Common Dev conf : None
  5. call classification : On-Net
  6. Media resource Grp : MRG_HQ
  7. Location : HQ_LOC
  8. AAR GROUP : HQ_AARG (if not using AAR leave empty)
  9. Packet Capture mode : None
  10. Packet Capture duration: 0
  11. MTP required : TICK
  12. Retry Video call as audio : TICK
  13. SIP information – Destination Add:
  14. DST is a SRV: UNTICK
  15. Destination port : 5060
  16. SIP PROFILE : Presence non-secure SIP trunk Profile
  17. Save

Step#4: Make your IP Phone presence capable

  1. Register a phone 2001 name it HQ-Phone1
  2. Create end user “test” and associate HQ-Phone1/2001 with the “test” user
  3. Make sure test user is a part of “Standard CCM End User” and “standard CTI enable”
  4. Make sure Primary extension “2001” is selected when you create the above “test” user

Also Make Physical phone DN2001 has “test” user associated with it. This is the last option in line 2001’s setting before “save” button. If this has not been done and you run presence diagnostic it will keep telling you that “No line appearance existed in CCM.

Step#5: Add an application user for IPPM and MOC CTI ports

This will be used by Presence server to initiate IP Phone services:

A) Go to > User Management>Application User>

  1. User ID : IPPM
  2. pass : blah
  3. Presence Group : Standard
  4. Select the Controlled devices for this feature
  5. Groups : Standard CCM End User
  6. Save

Repeat above “A” steps for MOC_USER as well. MOC_USER will be used by MOC CTI user in Presence. All users who want presence using Microsoft MOC client will be associated to this user.

Make sure all “accept” tick boxes are TICKED on MOC_USER.

B) Go to > SYSTEM>Application Server> Add NEW

Add Presence server IP address here i.e.

Step#6: Create IP Phone service URL

Go to> Device>Device Settings> IP Phone Service

  1. Service Name : IP PhoneMSG
  2. ASCII Service Name : IP PhoneMSG
  3. Service Description : Blah
  4. Service URL :
  5. Service Category : XML Service
  6. Service Type : Standard IP Phone Service
  7. Blank
  8. Blank
  9. Enable : TICK
  10. Save

****Then subscribe above service to HQ phone1/2001*****

Step#7: Enable presence licensing for each user

Go to> System>License>Capability Assignment>

Then Find the end user you want to assign the presence license.

Tick the user and hit <Bulk Assignment>

A new pop up window will come. Tick both check-boxes in that and save.

  1. Enable CUP – TICK
  2. Enable CUPC – TICK

That s all we needed to do on Call Manager. Now Jump on the Presence BOX.

Step#8: Presence box general configuration:

After installing basic presence, you’ll see presence post install setup screen on your web browser by typing presence Server IP address on your browser and supplying credentials to the login screen.

So you’ll see “Post Install Setup” screen with below options:

  1. CUCM Publisher IP address : (default, not changeable)
  2. AXL User : Administrator
  3. Axl password: blah…
  4. Confirm password : blah <then hit the “NEXT”>
  5. Security password : blah (whatever you supplied during installation)
  6. Then hit the “CONFIRM” (Ignore the warning)

Finally you will get 3 options:

A) Home B) Status C) TOPOLOGY

  1. Click on “HOME” you’ll see you are in a new home i.e. presence main admin page.

Step#9: Upload License and Activate presence Services

  1. First upload the license if you haven’t done that so far.
  2. GO to > Cisco Unified Serviceability>>Tools>Activate services
    Activate all services; it will take 2-3 minutes.

Step#10: Configure Presence

Jump straight on Presence Admin page>>Diagnostic>System Troubleshooter

Pay attention to RED crossed balls and yellow exclamation! Signs and fix them one by one.

  1. Under Presence Engine: Click on FIX under “no communication presence” this will take you to add presence gateway:

Add NEW>

Presence Gateway type : CUCM
description : blah

Presence Gateway: ← CCM IP

Double check the settings under below menus:

  1. SYSTEM> CCM Publisher : Check all parameter under this
  2. SYSTEM> Application Listener>Default class SIP TCP Listener (make sure its what you have defined in the SIP trunk on CCM – transport method TCP or UDP, both should have the same protocol/port) we are using:
    Protocol = TCP
    PORT = 5060
    Add NEW> description=blah/all address pattern=all

Step#11: Tune the Presence Engine’s Service parameter (same as we do with CCM)

SYSTEM>> Service Parameter>Select active CUPS Server> Select Presence Engine

SYSTEM>> Service Parameter>Select active CUPS Server> Select UP SIP Proxy(ver 7)


  1. Search “Proxy Domain” and set it to : domain name (Ex.
  2. Search “Transport Preferred Order” and set it to : TCP/UDP/TLS

Step#12: Configure IP Phone Messenger on Presence server

Application>IP Phone> Setting

  1. IPPM Application Status : ON
  2. Application user Name : IPPMSG (created in step 3A)
  3. Application Password: blah…
  4. confirm password : Blah
  5. Max Instant message : 25 default
  6. Subscription timeout : 3400 default   (3600 in ver 7)
  7. Publish timeout : 3600 default

Hit “SAVE”

Step#13: Select a SIP trunk between Presence to CCM

Tell presence which SIP trunk should be used for pumping calls to CCM.


  1. CUP CVP Support : UNTICK
  2. MAX Contact List Size : 200
  3. Enable Instant messaging : TICK
  4. Enable SIP Publish on CUCM TICK
  5. CUCM SIP Publish Trunk : <Select_Your_Trunk><– A MUST

Don’t forget to save after above. Above SIP trunk will be automatically listed in above “5”. This we is the one we created on CCM.

Step#14: Set TFTP address for IP COMMUNICATOR Clients

Application>Unified IP Personal Communicator>Settings

  1. Proxy Listener : Default Cisco SIP proxy TCP Listener
  2. Primary TFTP : (CCM pub tftp)
  3. Backup TFTP : (sub tftp) or whatever

LDAP – if you are using LDAP put LDAP parameters there. Else disable it.

Step#15: For MOC client define CTI Gateway

Application>>CUCM CTI Gateway>Settings

  1. Application Status : ON
  2. Application Username : MOC_USER (make sure its created on CCM as app user)
  3. Application Password : blah
  4. Confirmed Password : blah
  5. CUCM Address : (CCM address)

Now time to run the Presence troubleshooter again. This will tell you what’s remaining and how to fix it. Once those are done, activate the presence and other services. Still remaining:

  1. MOC integration
  2. Creating users and testing presence
  3. Voicemail integration with Presence
Categories: CCIE VOICE

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