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 it’s 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 OFFHOOK 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 770hz 852hz 1 4 7 2 5 8 3 6 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 Example: 8000x8=64000 COW x 8000, most samples will be the same. Hence compression! 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 Human voice/Codebook built as there are only so many frequencies used. G.729 codec used for 8Kbps voice. 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 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 spec’d 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. Ephones Can be single line or dual line (two simultaneous calls) Representation of Cisco IP Phone Linked to device by MAC ADDRESS Configuration 1. Printed on box of phone 2. Printed on back of phone 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 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 doesn’t 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 Add User User ID Name Display Name Pwd Generation: CUSTOM PIN Generation: Blank *E.164 Registration – Register with SIP Provider - ITSP 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: PART 1 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: Session target ipv4: 10.1.1.2 VAD = Voice Activity Detection Show dial-peer voice summary 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) Session target ipv4: 10.1.1.1 Show dial-peer voice summary 1005 …. Match 1005 dial peet tag id *IMPORTANT NUGGET* CME – DIAL PEERS: PART 2 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 CME – DIAL PEERS: PART 2 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 <DIGITS> (Add digits to left) • 2. Forward-digits <NUMBER> (How many digits?) • 3. Digit-strip – Turn off stripping • 4. Num-exp <MATCH> <SET> (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 doesn’t 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, doesn’t 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 didn’t 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 what’s 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  [ˆXYZ] [ˆX-Y] [ˆX-YZ] – Negative Digit Set  ˆ = Anything BUT!  Example of each:  38[2,4-6,9]3 = 3823  38[ˆ2-4]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 doesn’t 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 CISCO UNIFIED PRESENCE SERVER CISCO UNIFIED PRESENCE SERVER