Ccna Voice

CCNA VOICE - LAB SETUP

CUCM Software

GNS3

VMWare Workstation

Call Manager Express (CME)

Cisco Unity Connection/Cisco Unity Connection Express

Cisco 2600XM, 2801 Routers (ISR/VG)

NM2V WIC Card

CCNA VOICE INTRO CUCM can run in VMWare Google for GNS3 labs for CCNA Voice Voice/Data/Video Collaboration is the key Difficulty to integrate applications

Each area is its own world Cisco Goal: Unified Communications AVVID Acronym used prior to UC Bandwidth capabilities are increasing, therefore opportunities come to businesses and

homes. ISP see this opportunity in particular!

CCNA VOICE INTRO

Why VOIP?

Cost Savings:

MAC (Moves, Adds, Changes) $70-$200 per device)

Reduced wiring

Reduced telecommuter + branch office expenses

IT Staff + Application Consolidation

Toll Bypass (Long Distance in particular)

Soft Cost Savings:

Single Inbox for messages (Voicemail/Fax/Email)

Extension Mobility (Saves Office Space)

Internet Website Integration (Happy Client!)

Open Architecture (Multi Vendor Solution)

SIP is the TCP/IP of the voice world!

CCNA VOICE THE OLD TO THE NEW

Phase 1 Keep Existing PBX:

Requip existing routers for WAN + PSTN

Phase 2 Once all OK, removed PBX and use VOIP primarily.

FACT: 2 Million people in the USA still use rotary phones!

CCNA VOICE UNIFIED SOLUTIONS OVERVIEW

Core Products

Cisco Unified Communications Manager Express (Runs from router flash)

Cisco Unified Communications Manager (Runs from dedicated MCS appliance)

Cisco Unity Connection Express (Voicemail)

Cisco Unity Connection (Voicemail on dedicated MCS appliance)

Cisco Unified Presence (Tracking/IM)

CCNA VOICE - CUCME

Max of 450 IP phones, but 100 ideally!

Target Market: Enterprise Branch/Small Business Offices

Voicemail support added through CUE

Runs on Cisco ISRs (2800, 2900, 3800 etc)

Supports CLI + CCP configuration

CCNA VOICE - MODULES

VM Module required for CUE

AIM = Circuit Board/NM = Module with handle

AIM = Flash based/NM = HDD with Linux OS

3800 Series = CLI only

CCP = Nice and exam heavy as introductory

CCNA VOICE UNIFIED COMMUNICATIONS 500

8 to 48 phones

Integrated voicemail and auto attendant

External music on hold port

FXO/FXS Modules Analog connections

Routing and NAT features

VPN Users (10 max)

Optional 802.11 Wireless

Expensive!

CCNA VOICE CISCO UNIFIED COMMUNICATIONS BUSINESS EDITION

500 IP Phones

CCM Communications Manager

Cisco Unity Connection

Cisco Unified Mobility

NO REDUNDANCY!!

FULL CUCM BUT STRIPPED TO SINGLE SERVER

CCNA VOICE CISCO UNIFIED COMMUNICATIONS MANAGER FULL BEAST!

30,000 phones per cluster/shared database (60,000???)

Multiserver redundancy

Multisite support

Expensive!!

Flagship product for Cisco

CME supports up to 100 phones, CUCM exceeds 100 plus REDUNDANCY.

CCNA VOICE CISCO UNITY CONNECTION + UNIFIED PRESENCE Cisco Voicemail options: Cisco Unity Express Cisco Unity Connection Cisco Unity Cisco Unity Original version of unified messaging Runs on Windows OS Exchange + Domino integration PAINFUL to setup! Still has unique feature set, but is fading. (Direct tether to Exchange) 15000 users per server Cisco Unity Connection Linux Based Appliance Previously IMAP only now Exchange integration 20000 Users per server Unlimited number of telephone integration Featureiffic!!!

CCNA VOICE CISCO UNITY CONNECTION + UNIFIED PRESENCE CONT Cisco Unity Express Voicemail option with CME AIM + NM form factors (ISM and SM) 250 Users Max Basic interactive voice response (IVR) Auto attendant, email integration (Exchange) Cisco Presence Provides status information Integrates into nearly every IT facet! (CUCM/IP Phones/Unity/LDAP etc..) Uses industry standard SIP to collect data Integration with CUPC Cisco Personal Communicator Enterprise instant messaging

UNDERSTANDING ANALOG CONNECTIVITY

Pulse dialing around for 40 years (Rotary phones)

What is analog connectivity?

Transmission: Using some property of the transmission media to convey a signal.

Thomas Eddisons Phonograph in 1877-1900s Record players Braille Typical Home Telephones Lines Analog phone lines use the properties of electricity for voice transmission Phonograph: Store signals in a cylinder, bumps in tin foil (Stored and relayed

voice) magnetic fields/wave form signal.

UNDERSTANDING ANALOG CONNECTIVITY CONT..

Properties of electricity VOLTAGE X AXIS vs TIME Y AXIS As you speak into an analog phone, your voice is converted into electricity. The properties of the electricity are used to convey the properties of your voice. ANALOG: Loop and Ground Start Loop Start: PHONE has 2 wires that run in a complete circuit to a battery. PHONE ---------------------------RING WIRE--------------------------------BATTERY PHONE ---------------------------TIP WIRE------------------------------------BATTERY (CURRENT DETECT) When the receiver is ON-HOOK, the circuit is broken, when the phone is OFF-HOOK the circuit is complete. (Hence dialtone when phone is off hook)

UNDERSTANDING ANALOG CONNECTIVITY CONT.. Ground Start

Off hook signal temporarily accomplished by grounding the RING wire.

Grab outbound line and call inbound at the same time = GLARE

PHONES ---------- PBX ---------MULTIPLE LINES-----------CENTRAL OFFICE

Shoots ground signal down ring (Give me dialtone)

Supervisory Signal:

Used to send signals

ON HOOK/OFF HOOK/RINGING (Sent using AC current rather than DC)

Informational Signal

Dialtone/Busy/Ringback/Congestion/Reorder/Receiver Off Hook/No such number/Confirmation

UNDERSTANDING ANALOG CONNECTIVITY CONT.. Address Signal Dialing information over an analog line: PULSE BREAK/CONNECTED TIP/RING Dialtone Multi Frequency DTMF:

1209hz 1336hz 1477hz 697hz 1 2 3 770hz 4 5 6 852hz 7 8 9 941hz * 0 HASH *DIGITS REPRESENT FREQUENCIES*

UNDERSTANDING DIGITAL CONNECTIVITY

Problems with Analog

1. Distance limitations (Repeaters Layer 1) 2. Wiring limitations (Messy) *TIP AND RING FOR PAIRING KEY CONCEPT*

3. Digital voice eliminates distance issues

DIGITIZING VOICE Step 1 Sample the signal

If you sample the signal at twice the highest frequency, you can accurately reconstruct a signal digitally.

Common Frequencies:

Human Ear - 20-20,000hz

Speech - 200-9,000hz

Nyquist Theorum 300-4,000hz

Step 2 Perform quantization on the sample

Pulse Amplitude Modulation PAM

1. Take value of amplitude/voltage (Segments!) 2. Many samples are taken as low as possible in human speech range 3. PAM scale to line up samples with voltage level

Step 3 Convert to binary

Pulse Code Modulation PCM

A-LAW (OTHER PLACES) + N-LAW (USA) - A-LAW makes more sense!

Takes binary to represent POSITIVE and NEGATIVE 1 0 0 1 1 0 0 1 (2-4 bits SEGMENTS, 5-8 bits INTERVALS)

N-LAW is exact opposite! (Transcoding is where you convert between the 2)

Step 4 Optionally compress the samples

1. Send all 2. Just send changes 3. Build a codebook 4. Standard voice = 64Kbps Compressed value = 8Kbps with G.729

Example: 8000x8=64000 COW x 8000, most samples will be the same. Hence compression! Human voice/Codebook built as there are only so many frequencies used. G.729 codec used for 8Kbps voice.

MODERN VOICE: VOIP FOUNDATIONS

Call Processing Models Key Voice Protocols Deployment Models

DISTRIBUTED MODEL Phone Session RTP Connect Message Sent Bridge Phone and Phone for RTP communication

MODERN VOICE: VOIP FOUNDATIONS CENTRALIZED - Server and Client Model Faith in redundancy!

MGCP Protocol for Centralized model Routers and Phones are workhorses Simplicity SRST Backup/Failover/Mini Brain

KEY PROTOCOLS Signaling Protocols (Setup a call) H.323 Peer to Peer, Between VGs MGCP Server to Client, Between VGs SIP Long term option/victor SCCP Cisco Proprietary Streaming Protocols RTP Focus! Realtime Transport Protocol/

Sound of voice! RTCP Control/Stats for call

*SIP Supports proprietary extensions

CAMPUS IPT DESIGN Single Site

*G.711 Wideband Codec *ITSP No true QoS over WAN

MORE DESIGN..

WHAT IF NO WAN?

SRST When WAN fails, router takes on calls via PSTN.

PSTN should always be in place for backup.

TEHO Tail End Hop Off UK call via WAN to CHINA, then tailing off to a local call. FREE via WAN link and only then paying a local call fee.

DISTRIBUTED MULTI CLUSTER DESIGN

PREPARING THE INFRASTRUCTURE FOR VOIP

3 ROLES OF A CATALYST SWITCH

To provide Inline Power (Initially Cisco only) or Power Over Ethernet (802.3EF) Dual VLANs/Voice VLANs/Aux VLANs (Same thing..) Class Of Service CoS Layer 2 Markings How switch queues traffic.. + Quality

Of Service QoS Layer 3 Markings Prioritize traffic..

8 wires in standard network cable 4 used for Data transmission PoE uses opposite 4 cables

POWER

3 ways to power an IP Phone

Inline Power Cisco Pre Standard/IEEE 802.3AF Midspan Power Power Patch Panel (Cost wise might as well get PoE) Wall Power (Power supply/pack)

POE CONFIGURATION AND COMMANDS

Show power inline BIG ONE!

Switch is specd out to power every port with a phone

CDP communicates to the switch exactly how much power the phone is consuming.

Configuration

Conf t -> interface ______ -> power inline ->>> auto/delay/never

NORMAL SWITCHING WORLD One collision domain per port Broadcasts sent to all ports One subnet per lan Limited access control Vlans logically group users

Segments broadcast domains Subnet correlation Access control QoS VLANs traverse switches via trunks Switch adds tag with VLAN id

TAG is removed before hitting PC Only across trunks (To assist QoS) Flexability (VLANS)

Segmentation of users without routers (Layer 2) No longer limited to physical location Tighter control of broadcasts

VOICE/AUX VLANS General network design/security dictates voice and

data separation.

Seems impossible since IP phones have a built in switch.

VOICE VLANs always LOW VLAN ID! As STP will failover the lowest VLANs 1st!!!

PREPARING THE INFRASTRUCTURE FOR VOIP PART 2 IP Phone Boot Process

1. Cisco switch detects PoE capabilities. (Inline or 802.3af) 2. Switch sends voice VLAN via CDP to phone. 3. IP Phone sends DHCP discover and receives a DHCP offer including option 150 (IP

address of TFTP server)

4. IP Phone contacts TFTP server and receives configuration file. 5. IP Phone registers with CME router.

DHCP SERVICES ON A ROUTER

1. Excluded any necessary IP addresses (1-10 is best practice and/or 245-254) 2. Create DHCP pool 3. Define network 4. Define Default Router 5. Define DNS 6. Define any other options (150) 7. Configure IP helper addresses if needed *Option 66 = Option for TFTP by name rather than IP address

*IP Helper required for phones to obtain DHCP via highest Layer 3 interface

*show ip dhcp binding

NTP SERVERS

1. Configure NTP server 2. Optionally confogure one of more of your devices as NTP masters. Ntp server _ _ _ _

Clock timezone NAME hoirs offset from UTC

*show ntp associations

Designate CUCM as NTP master

Set from Stratom 1 Server

Ntp master

CISCO CALL MANAGER EXPRESS GETTING FAMILIAR WITH ADMINISTRATION CME Administration options CME command line CCP ADMIN - OS CLI (Jeremy Cioara preferred) CME GUI from router flash CCP Conf t Telephony-service ? LOTS OF OPTIONS! Show ephone registered Show ephone section ephone-dn SKY IS LIMIT! CLI Dial Peers/Phonebook/Route Plan CME GUI Phased out! Router GUI = RUBBISH Evolve to CCP!! CCP = Nextgen for SDM Features Wizard to setup router as CUCME Telephony settings have to be setup!

CME EPHONES AND EPHONE-DN Ephones-DNs are representations of directory numbers.

Can be single line or dual line (two simultaneous calls)

Configuration

CME Router

Show ephone MAC/SIGNALING PROTOCOL/ID

Show run include ephone

Conf t

Ephone-dn (1-150) tag single line/dual line

*You cant flip modes, you have to delete and reapply

Router provisions resources

Ephone-dn 1 number 1001

Secondary Numbers

Ephone-dn 1 number 1001 secondary 10001001

Ephones Representation of Cisco IP Phone Linked to device by MAC ADDRESS 1. Printed on box of phone 2. Printed on back of phone 3. Settings>Network Configuration Menu Show ephone *Auto registration by default Conf t Ephone 1 Mac-address ____ ____ ____ ____ Type _____ (OPTIONAL) *Add phone by MAC so router doesnt forget! Configure ephone + ephone DNs 1. Configure necessary EPHONE-DN 2. Configure necessary EPHONE 3. Associate EPHONE + EPHONE-DN using the

BUTTON COMMAND (Next slide..)

BUTTON COMMAND - BASICS

CIPC = Soft Phone (Cisco IP Communicator)

Under ephone:

Button ?

Button 1:2

1=BUTTON 2=EPHONE-DN 2

Then restart or reset.. (Restart = WARM BOOT, Reset = HARD RESTART)

Button 2:2

Button 3:3 etc

Button 1:1 2:2 3:3 One line of configuration.

At this point the phone is working on the network!

MORE BUTTON COMMAND MADNESS!

CISCO CME CISCO CONFIGURATION PROFESSIONAL Exam heavy!

**Service contract required with smartnet agreement**

Configuration document on Cisco website (Also prerequisites)

HTTP Based/Local Authentication user account

2 flavors of CCP:

Express running from the router flash CCP Full Suite on PC

Discover device before use!

Unfied Communications

Telephony settings to be setup 1st!

CISCO CME CISCO CONFIGURATION PROFESSIONAL

Steps via CCP

1. DNs and Phones (Any Order) 2. Links with user account (User unites DN + Phone)

Add Phone

Type

MAC

Autoline (Active Line)

Add DN

Primary Number - DN

Secondary Number DDI

Name

Description

*E.164 Registration Register with SIP Provider - ITSP

Add User User ID Name Display Name Pwd Generation: CUSTOM PIN Generation: Blank LINK PHONE+DN TO USER

CISCO CONFIGURATION PROFESSIONAL PART 2 - FEATURES CME Features

Phone Directory Forwarding Transfer Call Park Call Pickup Intercom Paging After Hours Restrictions Single Number Reach

CISCO CONFIGURATION PROFESSIONAL PART 2 - FEATURES Phone Directory Button press on phone -Personal Directory (Personal to user) -Corporate Directory -When an extension is created in CCP it is auto populated

into the Corporate Directory. -Advanced -> Directory Naming Schema -Telephony Settings/Directory Services Add manually

and limited to 100

CISCO CONFIGURATION PROFESSIONAL PART 2 - FEATURES Forwarding Extension -> Advanced -> Call Forwarding Toll Fraud Call forward/Max length (Stop international

calls on call forward) Transfer Transfer pattern CCP Advanced Telephony *Transfer to non Cisco phones* 9 _ _ _ _ _ _ _ _ _ _ (10 digits allowed only)

CISCO CONFIGURATION PROFESSIONAL PART 2 - FEATURES

Call Park -Cool! -Park call at phone number rather than extension. -Telephony Features -> Call Park -> Create -> Name -Number of slots for parking, reminders etc Lots of advanced features!!

CISCO CONFIGURATION PROFESSIONAL PART 2 - FEATURES Call Pickup Group of people in same team. Call can be picked up by any phone/

DN. Pickup group can be any number Searches group only Telephony features->Call Pickup Groups->Create->Add Extensions,

also Softkey on phone. *CUCM provides lots more configuration for pickup groups.

Intercom Bridge/button setup for intercom call. A directly to B. Used with directors/CEO etc Whisper Mode Intercom Auto Answer on speaker phone Can be set as a speeddial/Label Button Dedicated 2 way audio path between 2 phones

CISCO CONFIGURATION PROFESSIONAL PART 2 - FEATURES Paging Make an announcement using phone system to all phones! All phones will go into speaker phone mode CUCM never really had this feature, but CME has it! Paging Numbers Name/Description/Number/Members

(239.0.0.1:2000 UDP) Paging Groups Groups of Groups

After Hours Restrictions Not exam important.. Bonus! Allow or deny certain numbers during certain times. Telephony features->After Hours Toolbar Prefix Block/Schedules etc..

CISCO CONFIGURATION PROFESSIONAL PART 2 - FEATURES Single Number Reach People can reach you by dialing 1 number only. Extensions->Advanced->Single Number Reach Remote Number/Time (Seconds)/Timeout Value

GATEWAYS AND TRUNKS: UNDERSTANDING VOICE CODECS Digital conversion process

*Nyquist Theorum* Analog waveform/signals which are converted to binary.

How to turn spoken voice into bits with 4 steps. Step 1 Take many samples of the analog signal

Step 2 Calculate a number representing each sample (aka QUANTIZATION Pulse Code Modulation)

Step 3 Convert number to binary

Step 4 (Optional) Compress signal

GATEWAYS AND TRUNKS: UNDERSTANDING VOICE CODECS Common Audio Codecs

G.711 64Kbps MOS = 4.1 G.729 8Kbps MOS = 3.92 **NO 1 CODEC** G.729A 8Kbps MOS = 3.7 **NO 2 CODEC** G.726 32Kbps MOS = 3.85 G.728 16Kbps MOS = 3.61 ILBC Internet Low Bitrate Codec 15.2 Kbps MOS = 4.1 NEXT GEN + OPEN

SOURCE

MOS = Mean Opinion Score - BAD 1 5 GOOD Normal PSTN = MOS 4.0 4.1

*Each channel/DSO consumes BW value with all headers adds to 80Kbps.

GATEWAYS AND TRUNKS: UNDERSTANDING VOICE CODECS Choosing a codec and sample size

Sample size dictates the amount of audio included in each packet. (Default = 20MS of audio)

Larger samples = bandwidth samples

Larger samples = more delay

Bytes per sample = (Sample size * Codec Bandwidth) / 8

GATEWAYS AND TRUNKS: UNDERSTANDING VOICE CODECS Adding in data link/network overhead

Ethernet = 18 bytes

Frame Relay = 4-6 bytes

PPP/MLPPP = 6 bytes

-

IP = 20 bytes

UDP = 8 bytes

RTP = 12 bytes N + T = 40 bytes

Tunneling Bonus Overhead

GRE/L2TP = 24 bytes

MPLS = 4 bytes

IPSEC = 50-57 bytes

Adding it all together!

Total Bandwidth = Packet Size*Packet Per Second

Packet Size = 218 bytes

Packets Per Second = 50 x 218 = 10900 bytes per sec

10900 x 8 = 87200 bps/1000 = 87.2 Kbps of BW ---- WOOOOOOAH!

GATEWAYS AND TRUNKS: UNDERSTANDING VOICE CODECS

VOIP BANDWIDTH SAVINGS MEASURES 1. Voice Activity Detection (VAD): Suppresses the silence in the

conversation. Average of 35% BW savings. 2. Compressed RTP: Compresses network and transport layer

headers from 40 bytes to 2-4 bytes. Bandwidth savings are codec dependent. (Around 40% with G.729 CODEC) Option 2 is processor intensive!

GATEWAYS AND TRUNKS: UNDERSTANDING DIGITAL SIGNAL PROCESSOR RESOURCES Digital Signal Processors Offload media processing function from

voice processing equipment to dedicated hardware chips. - Coding - Transcoding (One CODEC to another) - Media Termination Point (MTP) - Conferencing Router = MIXER PSTN+VOIP into 1 stream.

VOICE TO PACKETS 2 forms of DSP = C549 and C5510 CODEC COMPLEXITY G.711 - MEDIUM G.726 - MEDIUM G.729A - MEDIUM G729AB MEDIUM

G.723 HIGH G.728 HIGH G.729 HIGH G.729B HIGH ILBC HIGH (DSP Calculator on Cisco Website)

GATEWAYS AND TRUNKS: UNDERSTANDING DIGITAL SIGNAL PROCESSOR RESOURCES RTP and RTCP

RTP is a QoS consideration

RTP carries audio payload between devices

RTCP carries call statistics between devices

RTP uses random, even numbered UDP ports between 16384-32767

RTCP uses random, odd numbered UDP ports between 16384-32767

GATEWAYS AND TRUNKS: CONNECTING CME TO OTHER VOICE SYSTEMS

CME to LAN CME to PSTN CME to PBX CME to PSTN VOIP

Voice Gateway Types

Analog voice gateway One call per port

Digital voice gateway Multiple calls per port

A voice gateway transitions between voice network types (VOIP/PSTN)

Same concept as a router separating networks.

GATEWAYS AND TRUNKS: FXO AND FXS

FXO+FXS

GATEWAYS AND TRUNKS: DIGITAL VOICE PORTS

Voice or data = VWIC 2MFT T1/E1 Card is beefy! 24 channels T1 32 channels E1 T1 and E1 Common Associated Signaling (CAS) Most common

RBS T1 and E1 Common Channel Signaling (CCS) Primary Rate

Interface (PRI) Basic Rate Interface (BRI) = 2 channels of voice, 1 for signaling CCS provides a dedicated channel for signaling.

GATEWAYS AND TRUNKS: VOICE GATEWAYS Gateways change

between VOICE and DATA.

Gateways bridge communications.

Gateway Control/Signaling Protocols

H.323 (Default/Old) Audio/Video Comms Suite

MGCP Used primarily by Cisco with Server + Client model

SIP Poised to be the universal VOIP standard

GATEWAYS AND TRUNKS: SIP

Designed as next generation H.323 Call Signaling and Call Setup Avaya use SIP all the time

CME DIAL PEERS: PART 1

Types of dial peers:

POTS Dial Peers

Connect to any traditional telephony network or devices

Defines number reachable through a given PORT (Keyword)

VOIP Dial Peers

Connect across any packet based network

Defines number(s) reachable at a given IP address

CME DIAL PEERS

CME DIAL PEERS

POTS + VOIP Example

CME DIAL PEERS Show voice port summary Ports->Sig Type->In Status->On/Off Hook Conf t Dial-peer voice tag type VOIP/POTS

Destination pattern 3301 Port 1/0/0 Dial-peer voice tag type VOIP/POTS Destination pattern 3302 Port 1/0/1

*This enables the 2 POTS phones to communicate through the CME router/FXS ports. Useful commands:

Debug voip dialpeer Show voice call summary

CME DIAL PEERS: VOIP

Conf t..

Dial-peer voice 330 voip

Destination pattern 330.

Session target ipv4: 10.1.1.2

*Default codec used is G.729*

1st Call Leg over IP to Voice Gateway

Show dial-peer voice summary

CME creates dial peers for all registered phones.

A target is required:


VAD = Voice Activity Detection


CME DIAL PEERS: WILDCARDS Period (.) = 1 digit

Plus (+) one or more proceeding digits

Brackets [ ] = Range of digits

Example [1-3] 1111, 2111, 3111

T = Any number of digits (0-32) *Generic wildcard

DP-9T (Anything up to 32)

Lazy dial plan 9.anything up to 32.

Dial-peer voice 10 voip

Destination-pattern 10.. (10XX)



1005 . Match 1005 dial peet tag id

*IMPORTANT NUGGET*


PSTN wildcards are out of CCNA Voice scope.. Phew!

Dial peers TO the PSTN are included!

CAS = Stealing bits/CCS = Dedicated

LEGACY VOICE ROUTER

Show voice ports summary

T1 = WIC Data or Voice

You need to tell the router which..

CONFIGURATION

Controller T1 1/0

Framing esf (USA)

Linecode b825

Ds0-group 5 timeslots 1-24 type fxo GROUND or LOOP *NOT IN CCNA


Show voice port summary

*Will display 24 FXO loop start ports

Pri-group timeslots DIGITAL PORT CFG

CCNA Voice you are only expected to know DIAL PEERS FOR PSTN

Destination-pattern 9T (PSTN WILDCARD DIAL PEER)

*Never know when done

*destination-pattern [2-9] (7 digit)

Wildcard for area code/local prefix destination-pattern [2-9]..

Dial-peer voice 9 POTS

Destination-pattern 9[2-9].. [2-9]

Port 1/0:5

Dial-peer voice 91 PORTS

Destination-pattern 91 etc (USA BIASED, NEED UK EXAMPLES)

CME DIAL PEERS: OUTBOUND DIAL PEERS

**The 555[1-3] Session target ipv4: 10.1.1.1 Dial-peer voice 2 voip Destination-pattern 5551 Session target ipv4: 10.1.1.2 Dial-peer voice 3 voip Destination-pattern 5551 Session target ipv4: 10.1.1.3 --------This dial peer wins. *Add a T . For 0-32 number of digits, also a # to process the call

immediately.

CME DIAL PEERS: INBOUND DIAL PEER MATCHING Next call leg has to have a dial-peer to know what to do. (IN and OUT)

MANIPULATING DIALED DIGITS

Auto stripping rule of POTS dial peers POTS dial peers automatically strip any explicit defined number

from the destination pattern before sending the call. Any non wildcard number specific only (EXPLICIT)

DIGIT MANIPULATION COMMANDS 1. Prefix (Add digits to left) 2. Forward-digits (How many digits?) 3. Digit-strip Turn off stripping 4. Num-exp (Match X and change to Y)

PSTN FAILOVER

PSTN FAILOVER

Dial-peer voice 6000 VOIP Destination-pattern 6 Session target ipv4: 10.1.1.2 *preference 0 Dial-peer voice 6001 POTS ----- PSTN BACKUP Destination-pattern 6 (STRIPPED) Port 1/0:1 No digit-strip Prefix 1512555 (USA LONG DISTANCE)

PSTN DIRECTING CALLS TO RECEPTIONIST

PSTN EMERGENCY CALLS

CLASS OF RESTRICTION Calling privileges ACLS for VOIP CCNA Voice COR concepts only not configuration Who can call what? PBX Realm Class of Service (NOT QOS) Call manager realm Class of Control Router realm = Class of Restriction Requires an in depth understanding of in + out dial peers Requires more detailed dial peers (No 9T for PSTN) Manually creating a PSTN dial plan EMERGENCY COR Dial-peer voice 999 pots Destination-pattern 999 No digit-strip Forward digits 3 (3 far right digits) INTERNATIONAL Dial-peer voice 12 pots Destination-pattern 9011T Prefix 011 (USA BIASED)

CLASS OF RESTRICTION

COR LISTS

Incoming dial-peer assigns incoming COR list

Outgoing dial-peer assigns outgoing COR list

If the OUTGOING COR list is a subset of the INCOMING COR the call IS forwarded

UNDERSTANDING COR *NOT FOR CCNA VOICE

*Bubble analogy

Dial-peer cos custom

Name 911 call

Name local call

Name ldcall

Name international

CLASS OF RESTRICTION

STEPS FOR COR

1. Define COR bubbles under cos custom NAME 2. Define outgoing COR lists (MEMBERS) 3. Define incoming COR lists (MEMBERS) 4. Assign COR to dial-peers WITHOUT INCOMING COR LIST YOU CAN DO ANYTHING

1. Lists and names 2. Outgoing bubbles 3. Incoming bubbles 4. Apply COR to dial-peer corlist incoming LD

CCNA VOICE FOCUSES ON CONCEPTS ONLY NOT CONFIGURATION

CISCO UNIFIED COMMUNICATIONS MANAGER OVERVIEW CUCME = CM on single voice gateway CUCM = Redundancy, scalable etc.. CUCM doesnt interface to PBX, hence voice gateway

requirement between digital and analog worlds.

Call Manager= The mind of the voice network

Major Functions: Call Processing, Signaling and Device Control, Dial Plan Administration, Phone Feature Administration, Directory Services and Link to External Applications.

CUCM HISTORY

Version 2.4 Cisco Made Own NT Based 4.0 2001 Install on any hardware Cisco blamed for faults!

Version 3.0 Only Cisco approved hardware Media Convergence Server BUT if not purchased from Cisco, NO end to to end support.

Version 4.x 2000 Version 4.3 2003 Version 5.x Linux Build/2003 Version 6.x Cisco stood ground on Linux based OS *MAINSTREAM*

CUCM - FEATURES

HTTP is 90% of all administration IE and Firefox only, doesnt like Chrome Navigation URL/cucmadmin (5 consoles)

System Menu = Global Configuration Mode equivalent Serviceability Menu = Monitoring/Alarms/Tools/Features/Services Control Center = Start/Stop Services (Features/Network) OS Administration = Tether to OS Disaster Recovery System = Backup/Restore CUCM database only OS Administration to update version of CUCM Cisco Unified Reporting = Reports/Data from CM/Sucks data from

all clustered CMs.

All services are installed by default, just activate and deactivate as required.

CUCM - CLI

You can SSH into the CUCM server

LAB CUCM with VMWare and CUCM ISO

SSH = Overlay of Linux OS (Restricted)

Utilities PING example

Database Replication

CUCM SUPPORTING END DEVICES

DEVICE POOL Assign settings to phone Assignment to IP Phone List of CUCM servers to use Codec to be used Time + Date information DEVICE POOLS group this configuration to a single assignment

CUCM SUPPORTING END DEVICES

REQUIRED DEVICE POOL ELEMENTS Device Pool NAME Cisco CM Group (Up to 3) Date/Time Group Region Softkey Template SRST Reference *DEVICE POOL is normally set as a LOCATION By default the CUCM group only contains the PUBLISHER

All auto registration devices will go to this PUBLISHER/CUCM GROUP

CUCM DEVICE POOL ELEMENTS

DATE/TIME CM Local = Default Greenwich Time Create Timezone REGION G.711 = Uncompressed 64Kbps per call G.729 = Compressed 8Kbps (20ms delay) G.729 offers human dictionary, MOS Scale/Score, Already covered! Different regions can use different codecs, dependent on their bandwidth capabilities. Relationships can be setup between regions Phones are added to the required region via DP membership.

CUCM DEVICE POOL ELEMENTS

SOFTKEY TEMPLATE Dictates what keys are available on the IP phone Device Device Settings Softkey Template You cannot change the default softkey template You can copy a softkey template Softkey Layout GO Undefined Key = BLANK SRST REFERENCE Disabled by default or use default voice gateway If router runs SRST, router supports phone and can talk to

PSTN/other site. Voice Gateway runs SRST System->SRST *2000 = PORT Pool is ready for devices to assign to Pool created per location

CUCM SUPPORTING END DEVICES PART 2

MANUALLY Enter MAC address and Directory Number for each phone. (UPC

code scanable) Selsius Ethernet Phone (Selsius are an organization) PHONE BUTTON TEMPLATE: controls line buttons DEVICE SECURITY PROFILE: Encryption settings Add live buttons: DN 2001 *HTTP Access to phone via HTTP server SECURITY CONCERN!! AUTO REGISTRATION CUCM hands out extensions to newly registered phones, similar to

DHCP. Default configuration file Static Assignment Good for new delivery

CUCM BULK ADMINISTRATION TOOL

BULK ADMINISTRATION TOOL Use an Excel spreadsheet to generate CSV file of devices. System->Unified CM->Enterprise Parameters->SCCP for auto

registration Device->Device Defaults>All Defaults for auto registration BAT Phones->Phone Template (Generic) Save Template (Sales Example) Add Line Template also Directory Number specified Spreadsheets Macros Enabled

LOCKING DOWN THE CISCO IP PHONE

Disable PC port Lock settings access Gratuitous ARP protect PC Voice VLAN access IP Phone HTTP access Product specific configuration layout GARP = ARP comes in that you didnt ask for. GARP sends a fake MAC address for your default gateway. Disabled in CUCM7. Phone conversation sent to PC and SWITCH. Call recording/monitoring etc.

CUCM SUPPORTING END USERS BENEFITS: Users can manage phone/Soft phones requires logins Advanced Features: Extension Mobility Tracking Per User Account There are 2 different types of users, END USERS and APPLICATION USERS End users can be linked to LDAP (Optional) CUCM can use 3 LDAP Options: 1.Local data only (NO LDAP) 2.LDAP Sync 3.LDAP Authentication LDAP Sync Disables bulk of CUCM User Management (Read Only) Passwords/CUCM specifics managed from CUCM LDAP Authentication Passwords managed in LDAP not in CUCM Authenticates directly against LDAP DATABASE

LDAP SUPPORT MS/NETSCAPE/SUN/IPLANET Must setup a SYNC AGREEMENT between CUCM and LDAP User Search Base and User ID LDAP attribute LOCAL LDAP User Management->App User->End User Add New Fields Associate END USER with device Now the user can login and manage associated devices.

CUCM BULK ADMINISTRATION TOOL PART 2

Used for large additions or changes to CUCM Database Phones/Users/Many tedious configurations Pre-integrated in CUCM Administration Export and reimport Exported data can be used for inplace migration or data

restore (Not possible with DRS) BAT COMPONENTS Template (Phone/Users) and CSV file -> BAT Engine Bulk Admin Excel Template/Upload or Download

Files/Download and open in Excel/Enable Macro/Run Macro and export to txt (CSV)

Create User Template Name Sales (Example)

CUCM MORE LDAP LDAP Supported Directory Active Directory Setup SYNC AGREEMENT Create Service/User Account in AD Create OU for sync Serviceability > Service Activation-> Enable Cisco DIR SYNC System->LDAP->LDAP System Enable Sync with Active Directory LDAP Attribute SamAccountName/Email etc.. LDAP = READY Add new LDAP directories: LDAP Manager = username@domain LDAP User Search Base - = OU (LDAP

SYNTAX=OU=CCMEndUsers,dc=home,dc=local) Add more than 1 DC. LDAP AGREEMENT in place, perform full sync. LDAP Admin done from AD. Change passwords from CUCM or AD. LDAP Authentication = Same setup as LDAP Sync. One in place, all password changes done from Active Directory.

CUCM: MANAGING GROUPS, ROLES AND PRIVILEGES

Delegate administrator rights Users assigned to groups Groups assigned to one or more roles Roles assigned to privileges *Important ordering!

CUCM UNDERSTANDING DIAL PLANS: CUCM ROUTE ARCHITECTURE

CUCM UNDERSTANDING DIAL PLANS

CUCM only knows about what is in the database No dialpeer required (CME configuration) CUCM only knows whats in cluster *1 PUBLISHER per cluster* ROUTE PLANS = Required for out of cluster communications.

CUCM DIAL PLANS CONFIGURATION

Add DEVICES VG and PSTN Call Routing -> Route Hunt -> GROUP/LIST/PATTERN Distribution Algorithm Circular or From Top (PREFERRED) Add Route List Ways to 2xxxx Groups to be used for calls When WAN call sent over PSTN requires TRANSFORMING *ROUTE

LIST LEVEL* PATTERN *LINK TO GATEWAY/ROUTE LIST Ties everything together 2xxx WILDCARD X = SINGLE DIGIT @ = North American Numbering Plan ! = One or more digits (32 digit cap) . = Access Code Termination HASH = Terminates Interdigit Timeout Provide outside dialtone Predot Strips before .

CUCM WILDCARD SAMPLES

[XYZ] [X-Y] [X-YZ] DIGIT SET [XYZ] [X-Y] [X-YZ] Negative Digit Set = Anything BUT!

Example of each: 38[2,4-6,9]3 = 3823 38[2-4]3 = 3813 9011!HASH = 9011_______________________HAS (32 digits then terminate)

CUCM PARTITIONS AND CALLING SEARCH SPACES Restricting devices from calling certain numbers

PARTITIONS Groups of dialable numbers - Lines/Route Patterns/Anything that has a number Examples: LOCAL-PT, INTERNAL-PT, INT-LD-PT

CALLING SEARCH SPACES A list of reachable partitions Assigned to any dialing entity Defines calling privileges Examples: INT-CSS contains INT-PT, LOCAL-PT partitions

Phone in the internal partition doesnt dictate calling privileges, this is where the CSS kicks in.

Partitions and CSS SIDE BY SIDE (GROUP and PRIVILEDGES) By default all phones+numbers are assigned to the NONE partition and CSS. Everything can call everything by default. Directory Number is in the partition not the phone. Bottom of all CSS is NONE partition. Best practice is to leave nothing in the NONE

partition or CSS.

CUCM PARTITIONS AND CALLING SEARCH SPACES Example 3 types of calling restrictions should exist in your organization:

1.Lobby/public phones: Internal Extensions only 2.Typical Users: Internal and Local PSTN 3.Management: Internal, Local PSTN and Long Distance PSTN

STEP 1 Create the partitions STEP 2 Assign numbers to partitions STEP 3 Create Calling Search Spaces STEP 4 Assign Calling Search Spaces to Devices

CUCM PARTITIONS AND CALLING SEARCH SPACES

CUCM FEATURE OVERVIEW Phone feature madness! Cisco IP Phone media streaming application IMPORTANT? CALL PARK Call Routing-> Call Park -> Add a range of numbers: 115x (0-9 Slots to Park call) Service Parameters -> Long List/Overview of a lot of odd parameters! Call Park in the list/

Settings for Call Park. CALL PICKUP Call Routing-> Call Pickup Group-> Add New Name + Number Group Call Pickup 0 Pickup SHARED PHONE LINES Add DN to 2 registered devices Multiple call waiting settings under DN Number of calls per device Max number can be set but 196 MAX Busy Trigger 2 people online, 3rd will be busy. Any change to DN impacts all devices. EDIT LINE APPEARANCE. DO NOT DISTURB Modify softkey template to enable DND feature Copy template to DND user Configure softkey layout On Hook Toggle DND Phone Configuration Menu On and Off Tickbox

CUCM FEATURE OVERVIEW CONT CALLBACK Lift handset and hungup User notifed that a user is available. Add Softkey under On hook Goes into effect on ring out of calling phone Crafty! Plays a chime when user is available. BARGE AND PRIVACY User can join a call (BARGE) Shared Lines Privacy button can prevent BARGING (DEFAULT) Device->Phone->Per Phone or Cluster Built in Bridge = BARGE On the fly, bridged conference call Phones handle conference call themselves Service Parameter Configuration = Global Settings for BARGE. SERVICES/EXTENSION MOBILITY Custom programs for phone. IP Phone Services Configuration Point to URL for apps Subscribe to this service under Device->Phone EXTENSION MOBILITY is covered in CCNP VOICE This is an XML service enabling users to login to the phone.

CISCO UNITY CONNECTION

FOCUS One of 5 LINUX VOIP appliances Integrates with legacy PBX via PIMG or TIMG USERS Manual, CSV, CUCM Import or LDAP CUC integrates with CUCM using SCCP or SIP PIMG Up to 8 digital or analog ports TIMG Digital T1 to SIP SCCP = Easier to setup than SIP (Jeremy opinion!)

CISCO UNITY CONNECTION CONT..

SIP Trunk used for CUCM and CUC Connection (Destination Address) SCCP requires message waiting (Integrated with SIP) CUCM Admin -> Voicemail Wizard based HOW CUC PROCESS CALLS: Call Handlers Scripting language for Unity System Greetings/IVR Equivalent/Series of Handlers Directory Type Users DN to reach them Interview Literally an interview, answer questions etc.. Info collector resource. Calls are identified as direct or forwarded DIRECT CALLS Messages button Calls Unity to collect VM FORWARDED CALLS CFNA, CFB, DND, Auto Attendant Forwarded when N/A

CISCO UNITY CONNECTION CONT..

Managing user and mailboxes in CUC User templates, make life easier! Lots of options BASIC ELEMENTS -User -Phone: Dialing Restrictions, CoS, Schedule -Location: Geographic location, language, time zone -CoS defines many options (Timers, Features, Restrictions) -You can create end users from the template+associate the phone

extension number. -Phone extension used to identify the user when they press the Messages

button -User template basics Core settings -If no VM box you can go to default greeting, Import Users->LDAP->Active Directory DV or AXL Remote (CUCM TO CUC) Service need activating for AXL (Serviceabilty Service Activation) LDAP and AXL

CISCO UNIFIED PRESENCE SERVER

Ccna Voice

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