Ccna Voice 640-461

CCNA VOICE 640-4611

CCNA VOIX Objectif et l'exigence. - Comprendre la certification Cisco voie. - Le monde la fin du 20me sicle. - Cisco Objectifs: communications unifies. - Pourquoi un organisme devrait utiliser VOIP. L'ancien vers le nouveau. Comment tirer le meilleur de cette srie. - Rfrences.* CCNA VOICE INTRODUCTION:

* CCNA- VOIX Objectif et exigences:CCNA VOIX publi en Juin 2008. Se dressant la (Gap) entre CCNA et CCNP. En Juin 2009, CCNA voix devient un pralable officiel pour CCNP.Objectif de CCNA VOIX:1- Conception. 2- Mettre en uvre. 3- grer un rseau vocal d'environ 200 utilisateurs.

* Comprendre La certification Cisco voix:CCENTCCNACCNA SecurityCCNAVoiceCCNPCCIE VoiceCCNPVoiceCCIE SecurityCCNPSecurityCCIER+SCCIEWirelessCCNPWirelessCCNA Wireless

* Le monde la fin du 20me sicle:VoiceVideoDataTrois rseaux distincts: Voice, Data, Video.Difficile d'intgrer des applications.- Chaque zone est un monde propre:: Infrastructure, staff.

* Objectifs: Cisco Unified Communications:B.W capacits ont augment pendant des annes. Fournisseurs de services voient l'opportunit. - Possibilit viens maintenant aux entreprises et les maisons.Voice Video

Data

* Pourquoi une organisation utiliserait VOIP:Traables et rduction des cots: Dplace, ajouter et changements (MAC).Rduire le cblage, Rduire les dpenses de tltravailleurs et de bureau de la branche.La consolidation de personnel informatique. La consolidation de la demande. Sans drivation.Souplesse et rduction des cots:Bote de rception unique pour les messages (messagerie vocale, fax, e-mail).La mobilit de poste (conomiser de l'espace de bureau).L'intgration de site Internet (client heureux).Une architecture ouverte (solution multifournisseur).

* L'ancien vers le nouveau:

* Comment tirer le meilleur de la srie:Rptition, rptition, la rptition. Prenez note, notez les informations cls que vous entendez.Achetez un routeur de laboratoire: -Cisco 2801 avec module CUE AIM, 2 FXS, 2FXO - Un couple Cisco IP Phone. - Logiciel de CUCM - GNS3Creuser plus profond. - Laissez vous sduire.

Le modle de structure Cisco Voix

The Cisco Voice Structure Model:extrmittraitement des appels

Call Processing: Cisco Unified Communication 500 ( UC500).

Cisco Unified Communication Manager Express (CME).

Cisco Unified Communication Manager Business Editions.

- Cisco Unified Communication Manager (CUCM).

traitement des appels: 1-Cisco Unified Communication 500 ( UC500):- Prise en charge de 8 48 tlphones IP. - Intgr dans le commutateur d'alimentation en ligne huit ports. - Musique externe sur le port d'attente (MoH). - (FXO) ports (pour les connexions RTC analogiques). - (FXS) ports (pour les connexions de tlphone / modem / fax analogiques). - Routage / (NAT) / Pare-feu de soutien. - (VPN) pour un maximum de 10 utilisateurs. -Optionnel Intgr 802,11 connectivit rseau sans fil. - Intgr messagerie vocale et Auto - Attendent.

traitement des appels2-Cisco Unified Communication Manager Express: CME est la prochaine tape de l'UC500. Soutien Amax de 450 IP Phone. March cible: Succursale Enterprise / PME. ajout Support Mail Voix grce Unity Express (CUE). Fonctionne sur Cisco ISR (2800, 2900, 3800, .......). Soutien ligne de commande et Cisco Configuration Professional (CCP).

Fournir volutivit 500 IP Phone. Combinez trois applications en un seul solution: - Cisco Communication Manager. - Cisco Unity Connection. - Cisco Unified Mobility. - Grande ...... Mais pas de redondance.Call Processing: 3-Cisco Unified Communication Manager Business Editions:MCS 7835

Call Processing: 4-Cisco Unified Communication Manager:chelle 30 000 tlphones IP par cluster Multi Server redondance. Soutien MultiSite. - Montrez-moi l'argent!!!!!!!!!!!!

* Cisco Call Processing Platforms:

Applications: Cisco Unity ( Voice mail).

- Cisco Unity Express. - Cisco Unity Connection. - Cisco Unity.

Cisco Unified Presence.

- Des applications supplmentaires.

Applications: 1-Cisco Unity Products: A-Cisco Unity Express: Jusqu' 300 des botes email. La solution la plus petite de l'unit, que vous pouvez ajouter votre rseau. AIM et NM facteurs de forme. Linux Based, la messagerie vocale de E-mail, Web, Tlphone. Rponse vocale interactive base (IVR).

Eng: Mohamed Saied AfifyApplications: 1-Cisco Unity Products: B- Cisco Unity Connection:Linux - bass sur les appareils (comme CUCM). Actuellement 20 000 des botes email max par serveur. Accs messages vocaux depuis ne importe o. L'intgration du serveur d'annuaire LDAP. - Le support de Microsoft Exchange.

Applications: 1- Cisco Unity Products: C-Cisco Unity :Fonctionner sur une plate-forme de Windows, intgre pleinement MS Exchange, Lotus Notes, systmes e-mail Novell Groupe Sage. Supporte totalement unifi de la messagerie. Actuellement 15 000 botes email max par serveur. Configuration Pnible. - A encore des caractristiques uniques, mais dclin rapidement.

Applications: 2- Cisco Unified Presence:Fournit l'tat des informations. Intgration avec Cisco Unified Personal Communicator (CCEP). Enterprise IM: (CUP incorpore Extensible Communication Platform japper (XCP)) La conformit des messages. Messagerie scurise.

Additional applications: - IVR / Auto attendant

Cisco Unified meeting place.

Cisco Unified Mobility.

- Cisco Emergency Responder

End Points: Tlphone IP.

- tlphones IP d'entre de gamme - Tlphone IP haut de gamme - 3- Tlphones IP avec cran tactile - Dispositifs spcialiss- Wireless / Cell Tlphone.

Video Tlphone.

- IM Client.

Eng: Mohamed Saied Afify1- tlphones IP d'entre de gamme Les tlphones IP suivants entrent dans cette catgorie: - Cisco 3911, Cisco 7906G, 7911G and Cisco 7931GCisco Unified IP Phone 7911Basic-caractristiques Cisco tlphones IP pour une utilisation faible moyenneInline power support- - Soutien des services de base de XMLCisco7931

Tlphones IP haut de gamme

Ces tlphones supporte deux ou plusieurs lignes - Des crans plus larges, haute rsolution, cran couleur - full-duplex Haut-parleur - Capacits d'alimentation en ligne. Example: Cisco 7941, 7942,7945 G Cisco 7960, 7961,7962,7965( six lines)

3- Tlphones IP Cisco avec cran tactile Cisco srie 797X Huit ligne tlphonique En couleur cran tactile 7975G (cran 5,6 pouces)

4-Appareil Spcial:Cisco 7937G Conference StationCisco 7937G Conference StationCisco7985 GCisco 7921G

Eng: Mohamed Saied AfifyCisco ATA 186/188 Les botes Cisco ATA sont de petits appareils qui peuvent convertir jusqu' deux tlphones analogiques (par ATA) dans les appareils VoIP. Cela peut tre la solution idale pour mettre existants tlcopieurs et appareils tout-en-un dans le rseau VoIP. - L'ATA 186 fournit une interface Ethernet unique.

Eng: Mohamed Saied AfifyCisco IP Communicator

Eng: Mohamed Saied AfifyCisco 7914/7915/7916 Expansion Modules

CCNA VOICE 640-4613

* Comprendre la Connexion analogique.

Cest quoi la connexion analogique?signaux lectriques: plus que vous ne le saviez.Comprendre Le signal analogique.

* Comprendre la Connexion numrique. Problme avec des connexions analogique. Conversion le signal analogique en numrique.

* Rsoudre le problme de cblage: TDMComprendre T1 et E1 CAS spcificits.Comprendre T1 et E1 CSC spcificits.

* Quelle est la connexion analogique?Transmission analogique: Utilisation de quelque proprit des moyens de transmission pour transmettre un signal. - Le phonographe de Thomas Edison en 1877. - Tourne-disques. - Braille pour les aveugles. - Les lignes tlphoniques de la maison typique.

- Lignes tlphoniques analogiques utilisent les proprits de l'lectricit pour la transmission de la voix.

* Proprits de l'lectricit:Comme vous parlez au tlphone analogique votre voix est convertie en lectricit.

Les proprits de l'lectricit sont utiliss pour transmettre les proprits de votre voix

Eng: Mohamed Saied Afify* Analog Signaling: 1- Loop and Ground start:a- Loop Start:

When the receiver is on-hook, the circuit is broken.

Eng: Mohamed Saied Afifyb- Ground Start:

Off-hook signal accomplished by temporarily grounding the ring wire.

Eng: Mohamed Saied AfifyAnalog Signaling:2- Supervisory Signal:

-Used to send the following signals over an analog line:

- On-hook - Off-hook - Ringing

- Ringing is sent using AC current rather than DC. 3-Informational Signal:

-Used to send the following signals over an analog line:

- dial tone - Busy - Ring back

- Congestion - No such number - Receiver off-hook

Eng: Mohamed Saied Afify

Eng: Mohamed Saied AfifyAnalog Signaling:

Analog Signaling:4- Address Signal:

- Used to send the dialing information over an analog line:

1- Pulse

2- Dual Tone Multi frequency (DTMF)

Eng: Mohamed Saied Afify* Problems with analog connections:1- Distance Limitation:2- Wiring Requirements:

Eng: Mohamed Saied AfifyHow to turn spoken voice into Bits?The 4 step recipe:

1- Take many samples of the analog signal. 2- Calculate a number representing each samples (AKA Quantization). 3- Convert that number to binary. 4- (Optional) Compress the signal

Eng: Mohamed Saied AfifyConverting Analog to Digital Connections:Digitizing Voice:Step 1: Sample the signal -The famed Dr: Nyquist formula - If you sample at twice the highest frequency, you can accurately reconstruct a signal digitally.Common Frequencies:

- Human ear 20 - 20,000 Hz Human speech 200- 9,000 Hz Telephone channels 300-3,400 Hz The Nyquist theorem 300-4,000 Hz

Eng: Mohamed Saied AfifyStep 2: Perform Quantization on the sample: Known as Pulse Amplitude Modulation (PAM)

Eng: Mohamed Saied AfifyStep 3: Convert the (Quantized) signal to binary: = 1 0 0 1 1 0 0 1Know as Pulse Code Modulation (PCM)2 PCM method: A-Law, Law

- A Law makes more sense

Eng: Mohamed Saied AfifyStep 4: Optionally Compress The signal: You can : 1- Send all these samples 2- Send just the changes 3- Build a codebook

* Standard voice sample: 64 Kbps

* Common compressed value: 8 Kbps (G.729)

Eng: Mohamed Saied Afify* Moving from Analog to Digital:Each number represents a sound that some one made while speaking in to telephone handset

Eng: Mohamed Saied AfifyTime Division Multiplexing Voice Channels:

Eng: Mohamed Saied AfifyThe problem of signaling :* In our analog world, signals = Electrical frequencies

* Once voice is digitized, signals must be 1s, 0s.

* Two methods of handle signaling in the digital realm: 1- Channel Associated Signaling. (CAS) 2- Common Channel Signaling. (CCS)

Eng: Mohamed Saied Afify1- Channel Associated Signaling: (CAS)The least significant bit in every 6th frame is signaling.

-T1 CAS steals bits from voice channel to transfer signaling information. Called (RBS)

Eng: Mohamed Saied Afify2- Common Channel Signaling: (CCS) CCS dedicates a signaling channel in T1 and E1 lines.

Most common signaling protocol is Q.931. ( used for ISDN circuit)

Most popular connection between voice system worldwide.

- More flexibility with signaling message. - Higher security. - More B.W for the voice bearer channels.

CCS also allows PBX vendors to communicate proprietary messages (and features) between their PBX systems using ISDN signaling, whereas CAS does not offer any of these capabilities.

When using CCS configurations with T1 lines, the 24th time slot is always the signaling channel.

When using CCS configurations with E1 lines, the 17th time slot is always the signaling channel.

- Other signaling protocols used with CCS configuration is SS7 ( communicate between COs)

CCNA VOICE 640-461Eng: Mohamed Saied Afify . 4

Eng: Mohamed Saied Afify* Pieces of the PSTN.

* Difference between PBX and Key systems.

* Connection to and between the PSTN.

* PSTN Numbering Plans.

* Understanding Voice Codecs.

* Roles of Digital Signal Processor.

* Choosing Acodec and Sample Size.

* VOIP B.W Saving measures.

Eng: Mohamed Saied Afify * Pieces of the PSTN:1- Analog Telephone2- local Loop3- Co Switch4- Trunk 5- private Switch6- Digital Telephone

Eng: Mohamed Saied Afify* Different between PBXs and Key Systems:PBX Systems: - Typically have digital PSTN connections. - Provide each users a unique extension. - Support a large number of feature.

Key Systems: - Typically have analog PSTN connections. - Uses shared lines between phone. - Support a smaller number of feature.

Eng: Mohamed Saied Afify* Connection to and between PSTN:

Eng: Mohamed Saied Afify* PSTN Numbering Plans:- PSTN Numbering Plans managed under the ITU (E.164) Standard. - Country Code. - National destination code. - Subscriber number.

Ex: North American Numbering Plan (NANP) - Country Code. - Area Code. - Central office code. - Station code.

Eng: Mohamed Saied Afify* Understanding Voice Codecs:- The powers that be created a measurement system known as a

Mean Opinion Score (MOS) to rate the quality of the various voice codecs. - The MOS is expressed as a single number in the range 1 to 5, where 1 is lowest perceived audio quality, and 5 is the highest perceivedaudio quality measurement.

Eng: Mohamed Saied Afify* Table shows how each audio codec fared in MOS testingYou will hear two codecs continually repeated: G.711, G.729.

- G.729 comes in two different variants: G.729a, G.729b

Eng: Mohamed Saied Afify* Roles of Digital Signal Processor:- Cisco designed its routers with one primary purpose in mind: routing. In the realm of VOIP, the network requires the router to convert the loads of voice in to digitized, packetized transmission.

This task would overwhelm the resources you have on the router.

DSPs offload the processing responsibility for voice related task from the processor of the router.

DSP is a chip that performs all the sampling, encoding, and compression functions

on audio coming into your router.

Eng: Mohamed Saied Afify PVDM2-8 : Provides .5 DSP chip

PVDM2-16: Provides 1 DSP chip

PVDM2-32: Provides 2 DSP chips

PVDM2-48: Provides 3 DSP chips

PVDM2-64: Provides 4 DSP chipsBased on the DSP requirements given by the Cisco DSP calculator, you can

purchase one or more of the following PVDM:

Eng: Mohamed Saied Afify* Choosing A codec and sample size:- G.711 G.729 G. 726 GSM FR

G.722 G.723 G.728

Sample size dictates the a mount of audio included in each packet (default 20 ms)

Larger samples = B.W saving

Larger samples= more delay

Bytes per sample = (Sample size * Codec B.W) / 8

Ex: ??? = ( 0.02 * 64000 ) / 8

= 160

Eng: Mohamed Saied Afify* Adding in data link and network overhead: Ethernet 18 bytes

Frame relay 4-6 bytes

PPP/ MLPPP 6 bytes

IP 20 bytes

UDP 8 bytes

RTP 12 bytes Layer 2Network + Transport layer = 40160 + 18 + 40 = 218 Bytes/ Packet

Eng: Mohamed Saied Afify* Tunneling: Bonus overhead GRE/ L2TP 24 bytes

MPLS 4 bytes

IPsec 50-57 bytes * Adding it all together:Total B.W = Packet Size * Packet per second

?? = 218 * 50

= 87200 bps

= 87.2 kbps

Eng: Mohamed Saied Afify* VOIP B.W saving measures:1- Voice Activity Detection (VAD): Suppress silence in the conversation

Average 35% B.w saving

2- Compressed RTP : Compresses Network and Transport layer header from

40 bytes to 2-4 bytes

B.w saving codec dependent ( a round 40 % with G.729 codec)

Eng: Mohamed Saied Afify* Modern Voice: VOIP Foundations Call Processing model.

Deployment model.* Cisco Voice Gateway Types Analog Voice Interfaces (FXS-FXO-E&M).

Digital Voice Interfaces.* Voice Gateway Communication Protocol H.323 - SIP - MGCP - SCCP.* Voice and Video Transport Protocol RTP -RTCP.

Eng: Mohamed Saied Afify* Call Control Models: (Distributed)

Eng: Mohamed Saied Afify* Call Control Models: (Centralized)

Eng: Mohamed Saied Afify* Campus IPT Design:

Eng: Mohamed Saied Afify* Central Site Deployment Design:

Eng: Mohamed Saied Afify* Distributed, Multi cluster Design:

Eng: Mohamed Saied Afify* Cisco Voice Gateway Types: Analog Voice gateway ( one call per port).

Digital Voice gateway ( Multiple call per port).

Eng: Mohamed Saied Afify1- Analog Voice Ports: FXS

Eng: Mohamed Saied Afify2- Analog Voice Ports: FXO

Eng: Mohamed Saied Afify3- Analog Voice Ports: E&M

Eng: Mohamed Saied Afify* CME voice gateway supporting analog voice connections:

Eng: Mohamed Saied Afify* Digital Voice Ports:T1 and E1 CAS

T1 and E1 CCS ( Also called PRI)

BRI ( Basic Rate Interface)

ISDND Channel 16 kb/s (Signaling)2 B channels (Voice)ISDN BRID Channel 64 kb/s (Signaling)30 B Channels (Voice)ISDN E1 PRID Channel 64 kb/s (Signaling)24 B Channels (Voice)ISDN T1 PRI NFAS23 B Channels (Voice)

Eng: Mohamed Saied Afify* Connecting CME to other Voice Systems:

Eng: Mohamed Saied Afify* Voice Gateway Communication Protocol:

PSTN

PSTN

Eng: Mohamed Saied Afify The Unified Communication System supports four methods of Gateway Communication:

1- H.323

2- MGCP

3- SIP

4- SCCP

* Cisco Unified Communications Manager Signaling and Media Paths:Cisco Unified Communications Manager Express(SCCP / SIP)Media Exchange (RTP)(SCCP / SIP)- Cisco Unified Communications Manager performs call setup using a Signaling Protocol (SCCP/SIP).- Media exchange occurs directly between endpoints using RTP.Eng: Mohamed Saied Afify

Eng: Mohamed Saied Afify Voice and Video Transport Protocols:

1- RTP

2- RTCP

Eng: Mohamed Saied Afify* Beginning With Vision.

* Three key roles of catalyst switches.

* Connecting and Powering Cisco IP Phones.

* VLAN Concepts and Configuration.

* Understanding Voice VLANS.

* Understanding the Cisco IP phone Boot Process.

Eng: Mohamed Saied Afify* Beginning With Vision:

Eng: Mohamed Saied Afify* Three key roles of catalyst switches: 1- Inline power / Power Over Ethernet .

2- Dual VLANs / Voice VLANs / Auxiliary VLANs.

3- Class Of Services / Quality Of Services.

Eng: Mohamed Saied Afify* Connecting and Powering Cisco IP Phones: RS 232

10/100 SW

10/100 PC

Eng: Mohamed Saied AfifyThere are three ways of providing power to your phone:

- Power Brick (Wall Power).

- Midspan Power (Powered Patch Panel/ Power injector).

- Power over Ethernet (POE) Switch.

Eng: Mohamed Saied Afify1- Power Brick (Wall Power): NetworkNon-POE SwitchEthernetConnectionPower Brick

Eng: Mohamed Saied Afify2- Powered Patch Panel / Power injector: NetworkNon-POE SwitchPower andEthernetEthernetOnlyProvides Power to entire patch panel

Eng: Mohamed Saied Afify3- Power over Ethernet (POE) Switch: Cisco Inline Power

Cisco IEEE 802.3af NetworkPOE SwitchPower and Ethernet

Eng: Mohamed Saied Afify* IEEE 802.3af Classifications:

Eng: Mohamed Saied Afify* 802.3af Power Classes:

Eng: Mohamed Saied Afify* VLAN Concepts and Configuration:- The Normal Switching World:

- One collision domain per port. - Broad cast sent to all port - One subnet per LAN. - Very limited access control.A usual SwitchPC 2PC 1

Eng: Mohamed Saied Afify - There are many benefits to using VLANs in an organization:

- Logically group users.

- Segments broad cast domain.

- Subnet correlation.

- Access control.

- Quality of Service.

Eng: Mohamed Saied Afify* VLAN Trunking / Tagging:

Eng: Mohamed Saied Afify* Understanding Voice VLANS:

Eng: Mohamed Saied Afify* Understanding the Cisco IP phone Boot Process:1- Cisco Switch detects POE Capabilities.

2- Switch sends Voice VLAN via CDP.

3- IP phone receives DHCP request including option 150.

4- IP phone contacts TFTP server, receives configuration file.

5- IP phone registers with CME.

Eng: Mohamed Saied Afify*Preparation the infrastructure for VOIP:

- Beginning With Vision. - Configuring IP Phone DHCP support.

- Configuring NTP.

Eng: Mohamed Saied Afify* Beginning With Vision:Voice 172.16.1.0Data 172.16.2.0

Eng: Mohamed Saied Afify* Configuring a Trunk to the CME Router:

Eng: Mohamed Saied Afify* Configuring Inter-VLAN Routing:

Eng: Mohamed Saied Afify* Configuring DHCP Services on a router:1- Exclude any necessary IP addresses.

2- Create DHCP pool. - Define Network. - Define default router. - Define DNS setting. - Define any other option (150).

3- Configure IP helper- address, if necessary

Eng: Mohamed Saied Afify* Configuring a Router-Based DHCP Server:.10.10

Eng: Mohamed Saied Afify* Setting the Clock of a Cisco Device with NTP: It allows Cisco IP Phones to display the correct date and time to your users.

It assigns the correct date and time to voicemail tags.

- It gives accurate times on Call Detail Records (CDR), which are used to track calls on the network.

It plays an integral part in multiple security features on all Cisco devices.

It tags logged messages on routers and switches with accurate time information.

Eng: Mohamed Saied Afify* Configuring NTP:Configure NTP Server.

- Optionally, configure one or more of your devices as NTP master.

Eng: Mohamed Saied Afify* Licensing and Model for Cisco Unified CME.

* Installing Unified CME on Cisco router.

Eng: Mohamed Saied Afify* CME It is not free:1- IOS license. . Licenses the IOS software on the router.

2- Feature license. . License CME for specific number of users.

3- Phone user license. . Licenses the IP Phone to interact with CME or CCM.

Eng: Mohamed Saied AfifyCisco Unified Communications Manager Express Feature Licenses

Eng: Mohamed Saied AfifyISR bundle examples

Eng: Mohamed Saied Afify* CME design model / configuration model: 1- PBX model.

2- Keyswitch model.

3- Hybrid model.

Eng: Mohamed Saied Afify1- PBX model :

Eng: Mohamed Saied Afify2- Key switch model :

Eng: Mohamed Saied Afify3- Hybrid model :

Eng: Mohamed Saied Afify* Getting the necessary files:1- Basic files.

2- GUI files.

3- XML Templates files.

4- MOH files.

5- Script files.

6- Miscellaneous files.

Eng: Mohamed Saied Afify* Installing CME:1- Get the necessary files from the Cisco website.

2- Place the files on an accessible TFTP server.

3- Copy the files to the router flash memory.

- Using the ( Copy ) command ( hourly). - Using the ( archive ) command (salary).

Eng: Mohamed Saied AfifyGetting Familiar with CME administration

Eng: Mohamed Saied Afify* CME administration options:

- Command line.

- Graphic User Interface (GUI).

Eng: Mohamed Saied Afify* Managing CME using the command line: More flexible.

Support all CME feature.To access the command-line interface (CLI) of the CME router, use one of three methods:

1- Console Port.

2- Telnet access.

3- SSH access

Eng: Mohamed Saied AfifyTelephony service.

ephone.

ephone-dn.

Dial-peer.

Eng: Mohamed Saied Afify* Managing CME using GUI:1- Integrated CME GUI.

2- Cisco Configuration Professional (CCP).

- CCP Express. - CCP.

Eng: Mohamed Saied Afify* Integrated CME GUI

Eng: Mohamed Saied Afify* Creating a CCP Community of Managed Devices

Eng: Mohamed Saied Afify* Connecting to the CME Router Securely

Eng: Mohamed Saied Afify* Unified Communications Initial Configuration

Eng: Mohamed Saied Afify1234

Eng: Mohamed Saied Afify* Ensuring the foundation.

* CME: Three Key commands.

* Specifying the necessary load information.

Eng: Mohamed Saied Afify* Ensuring the foundation:

1- Know the IP Phone boot process!

2- Voice Vlans configured, CDP enabled.

3- DHCP services configured, option 150 enabled.

4- TFTP server configured with correct files.

.Phones are now coming in droves to CME.

Eng: Mohamed Saied Afify* CME: Three Key commandsIssued from Telephony Service configuration mode

- Max-DN.

- Max-Ephones.

- IP Source Address.

Eng: Mohamed Saied Afify* Specifying the necessary load information: CME extracts firmware (load files) into flash for supported devices.

Newer CME versions organize these into sub directories.

Firmware must be specified using the (load) command under Telephony-Service mode.

These file should also be made available via TFTP.


* Understanding the purpose.

*Configuring Ephone DNs.

* Configuring Ephone.

Ephones and Ephone-DNsPart (2)* Possibilities with EPhone-DNs.

* Shared line configuration.

* Auto- Registration.


- Understanding the purpose.

- Configuring Ephone DNs.

- Configuring Ephone.

Eng: Mohamed Saied Afify* Understanding Ephone-DNs: Ephone - DNs are representations of directory numbers.

Can be Single-Line or Dual-Line.

Single-Line

. Only able to handle asingle active call at atime.

. If on active call, incoming call receive busy signal. Dual-Line

. Handles two simultaneous Calls.

. Necessary for call waiting, Conference calling, Consultative transfers.

Eng: Mohamed Saied Afify* Understanding Ephone: Ephone are representations of Cisco IP Phones.

Are linked to the device by the mac address

- Printed on the box of the Cisco IP Phone.

- Printed on the back of the Cisco IP Phone.

- From the setting > network configuration menu of the IP Phone.

Eng: Mohamed Saied Afify* Configuring Ephone and Ephone DNs: Create necessary Ephone DNs.

Create necessary Ephone.

Associate Ephones and Ephone DNs using the Mystical Button Command.

Eng: Mohamed Saied Afify* Why the Button Command is Mystical:All the possible button arguments:Example: button 1:5 button 2m6 3m7 4m button 5f10

Eng: Mohamed Saied Afify Possibilities with EPhone-DNs.

- Shared line configuration.

- Auto- Registration.

Eng: Mohamed Saied Afify1001Ephone1Ephone 2Button 1 m 2Button 2 w 2

Button 1 : 1Button 2 s 2Button 3 b 3Button 4 f 4

100210031004

Eng: Mohamed Saied Afify* Understanding Shared Lines and Button overlay:- The following creates a shared line configuration:- Problem: One call per Ephone DNs, one active user.Ephone 8

Ephone 9

10101010

Eng: Mohamed Saied Afify* Understanding Shared Lines and Button overlay:- This also creates a shared line configuration:Problem: Incoming calls randomly distribute.Ephone 8

1010Ephone 8

1010 Can be helped using the preference and huntstop commands.

Eng: Mohamed Saied Afify* Understanding Shared Lines and Button overlay:

Eng: Mohamed Saied Afify* Understanding Shared Lines and Button overlay Commands: Button Overlay allow you to associate multiple Ephone-DNs with a single line instance.

The two primary button overlay commands: - O Separator- Overlay line with no call waiting. - C Separator- Overlay line with call waiting.

Eng: Mohamed Saied Afify* Understanding Auto- registration and assignment: By default, CME registers and Phone ( you can disable this).

Ephone registration are not saved.

Auto- assignment associates Ephone-DNs to new Ephones.

Auto- registered phones need to be restarted to auto assign.

Eng: Mohamed Saied AfifyCisco CME: Voice Productivity Features Part (1)

Eng: Mohamed Saied Afify Configuring a Voice Network directory.

Configuring Call forward.

Configuring Call Transfer.

Configuring Call Park.

Configuring Call Pickup.

Eng: Mohamed Saied Afify* Configuring Network directory:

Eng: Mohamed Saied Afify* Configuring Call Forward:

Eng: Mohamed Saied Afify5

Eng: Mohamed Saied Afify* Configuring Call Transfer:

Eng: Mohamed Saied Afify* Configuring Call Park: Allow you to Park a call on hold.

Configured using the park slot command, which has many options

- Reserved for DN. - Time out Seconds. - Limit Count. - Notify DN. - Only. - Recall. - Transfer DN. - Alternate DN. - Retry Seconds.

Eng: Mohamed Saied Afify* Configuring Call Pickup: Answer another ringing phone in the network.

Three types of call Pickup.

- Directed Pickup. - Local group Pickup.

- Other group Pickup. Group 1101Group 1102

Eng: Mohamed Saied Afify110111011102110111021102Group 1101Group 1102100210011003100610051004

Eng: Mohamed Saied AfifyCisco CME: Voice Productivity Features Part (2)

Eng: Mohamed Saied Afify Configuring Intercom.

Configuring Paging.

Configuring After-Hours Call Blocking.

Configuring Single Number Reach.(SNR)

Configuring Music on Hold.(MOH)

Eng: Mohamed Saied Afify* Configuring Intercom:303134 : 3052 : 31AssistantManagerA100A101

Eng: Mohamed Saied AfifyThere are three other arguments you can use with the intercomcommand to tune the functionality:

barge-in: Automatically places an existing call on hold and causes the intercom to immediately answer.

no-auto-answer: Causes the phone to ring rather than auto-answer on speakerphone.

no-mute: Causes the intercom to answer with unmuted speakerphone rather than muted. Although this is beneficial to allow immediate two-way conversation, you run the risk of one side barging into existing conversations or background noise.

Eng: Mohamed Saied Afify* Configuring Paging:1- Paging is a one way speaker phone based announcement.

2- Accomplished by creating a paging number and assigning IP Phones to the paging number.

3- Each IP Phone can only be assigned one paging number.

4- Supports unicast or multicast mode.

5- Supports multiple paging groups.Paging Group 5510Paging Group 5511

Eng: Mohamed Saied Afify- Configuring Unicast, Single-Group PagingTo multicast paging, you could modify the paging command with the following syntax:3255102323232

Eng: Mohamed Saied Afify- Configuring Multiple-Group Paging323232333332,335510551155123334

Eng: Mohamed Saied Afify* Configuring After-Hours Call Blocking:After-hours call blocking has three major steps of configuration:

Step 1: Define days and/or hours of the day that your company considers off-hours.

Step 2: Specify patterns that you want to block during the times specified in Step 1.

Step 3: Create exemptions to the policy, if needed.

Eng: Mohamed Saied Afify- Configuring After-Hours Time Ranges and Dates

Eng: Mohamed Saied Afify- Configuring After-Hours Block Patterns

Eng: Mohamed Saied Afify- Configuring After-Hours Exemptions

Eng: Mohamed Saied Afify* Configuring Single Number Reach: Allow you to link an additional device to a parent number.

SNR in CME is a lightweight version of Mobile Connect, which

is a CUCM feature allowing you to assign multiple devices to ring

simultaneously.

- Using SNR may cost you additional voice trunk to PSTN.

Eng: Mohamed Saied Afify* Configuring Music on Hold:- Allow you to stream a .Wav or .au file stored in routers flash memory.

Supports unicast or multicast mode.

Supports G.711 or G.729 Codecs.

Dont get caught playing copyrighted stuff !

Eng: Mohamed Saied AfifyUnderstanding The CME Dial - Plan ( Part 1)

Eng: Mohamed Saied Afify Configuring Physical Voice Port Characteristics.

Configuring a T1 CAS PSTN Interface.

Configuring a T1 CCS PSTN Interface.

Understanding Dial Peers.

Understanding Call legs.

Configuring POTS and VOIP Dial-Peers.

Dial-Peer wildcards and digit manipulation.

Configuring a North American PSTN Dial-Plan.

Understanding outbound and Inbound Matching

Eng: Mohamed Saied Afify* Configuring Physical Voice Port Characteristics:- Configuring Analog Voice Ports: - Foreign Exchange Station Ports (FXS)

- Foreign Exchange Office Ports (FXO)

- Configuring Digital Voice Ports

Eng: Mohamed Saied AfifyFXS ports have three common areas of configuration:

Signaling.

Call progress tones.

- Caller ID information.

Eng: Mohamed Saied Afify module 1 module 02621 XMFa 0/0Fa 0/1VIC-2fxVWIC-IMFTT1NM-HD-2VE1/0/101

Eng: Mohamed Saied AfifyForeign Exchange Office PortsTwo additional commands are of note:

- dial-type

- ring number

Eng: Mohamed Saied Afify* Configuring a T1 CAS PSTN Interface:

Eng: Mohamed Saied Afify* Configuring a T1 CCS PSTN Interface:

Eng: Mohamed Saied Afify* Understanding Dial Peers:-Types of Dial Peers

1- POTS Dial Peers

Connect to any traditional telephony network or devices.

Define number(s) reachable through a given port.

2- VOIP Dial Peers

- Connect across any packet-based network.

- Define number(s) reachable through at a given IP address.

Eng: Mohamed Saied Afify* Understanding Call legs: Call legs define the voice route path.

Every change in network (type) requires anew call leg.

Call legs are define inbound and out bound.

Eng: Mohamed Saied Afify* Configuring POTS Dial Peers:330133013301330233021/0/01/0/1Router B

Eng: Mohamed Saied Afify* Wildcards You Can Use with the destination-pattern Command

Eng: Mohamed Saied Afify* Configuring VOIP Dial Peers:330330.1010..

Eng: Mohamed Saied Afify* Configuring a North American PSTN Dial-Plan:

Eng: Mohamed Saied Afify* Understanding outbound and Inbound Matching:

Eng: Mohamed Saied Afify Outbound Dial Peers:

The most specific destination pattern always wins

Eng: Mohamed Saied AfifyUnderstanding The CME Dial - Plan ( Part 2)

Eng: Mohamed Saied Afify Understanding outbound and Inbound Matching.

Digit Manipulation.

Understanding and Implementing CME Class of Restriction.

Eng: Mohamed Saied Afify* Understanding outbound and Inbound Matching:

Eng: Mohamed Saied Afify Outbound Dial Peers:

The most specific destination pattern always wins

Eng: Mohamed Saied Afify1- Match the dialed number (DNIS) using the incoming called-number dial peer configuration command.

2- Match the caller ID information (ANI) using the answer-address dial peer configuration command.

3- Match the caller ID information (ANI) using the destination-pattern dial peer configuration command.

4- Match an incoming POTS dial peer by using the port dial-peer configuration command.

5- If no match has been found using the previous four methods, use dial peer 0.

* Inbound Dial Peer:

Eng: Mohamed Saied Afify* Inbound Dial Peer Matching Examples:

Eng: Mohamed Saied Afify Any Voice codec. No DID Support. No IVR. VAD enabled. No DTMF Relay. No RSVP.Dial Peer 0

Eng: Mohamed Saied Afify* Digit Manipulation: -The Auto Stripping Rule of POTS Dial Peers

Eng: Mohamed Saied Afify*Digit Manipulation Commands: Prefix < Digit >

Forward Digit < Number >

Digit Strip

Num- exp < match > < set >

Voice Translation profile

Eng: Mohamed Saied AfifyPractical Scenario 1: PSTN Failover Using the prefix Command

Eng: Mohamed Saied AfifyIP WANPSTN

Eng: Mohamed Saied AfifyPLAR1

Eng: Mohamed Saied AfifyPractical Scenario 2: Directing Operator Calls to the Receptionist5000

Eng: Mohamed Saied AfifyPractical Scenario 3: Specific POTS Lines for Emergency Calls

Eng: Mohamed Saied AfifyDigit Manipulation Order of Operation for POTS Dial Peers

Eng: Mohamed Saied Afify* Understanding and Implementing CME Class of Restriction: Prevent standard employees from making international calls, but allow management to place international calls without restriction.

Block certain high-cost numbers (such as 1-900 numbers in the United States).

Prevent certain internal phones from reaching executive office directory numbers.

Eng: Mohamed Saied AfifyX 3000X 4000X 5000

Eng: Mohamed Saied Afify- Emergency Calling- 10 Digit Dialing- 11 Digit Dialing- International* Manually Creating A PSTN Dial Plan ( In North America)

- 7 Digit Dialing


* Understanding COR:- First, Define COR Under COR Custom- Second, Define Outgoing COR Lists- Third, Define Incoming COR Lists- Fourth, Assign CORs to Dial - Peers

Eng: Mohamed Saied AfifyX 3000X 4000X 5000


* How Routers Implement Class Of Restriction:


Ex: 240000Eme-LocaEme-Loca4000

Eng: Mohamed Saied Afify*There are a couple important rules of COR lists that you should know:

Rule 1: If there is no outgoing COR list applied, the call is always routed.

Rule 2: If there is no incoming COR list applied, the call is always routed.

Eng: Mohamed Saied AfifyCisco CUCM

Administration Overview, Supporting End Device

Eng: Mohamed Saied Afify Cisco CUCM: Administration Overview.

Cisco CUCM: Supporting End Device.

Eng: Mohamed Saied Afify Reviewing the place of CUCM.

- Getting familiar with CUCM Web management.

- CUCM Command line interface.* Cisco CUCM: Administration Overview.

Eng: Mohamed Saied Afify Cisco Unified Communications Manager Administration.

Cisco Unified Serviceability.

- Disaster Recovery System.

Cisco Unified Operating System Administration.

- Cisco Unified Reporting.

- Command-Line Interface (CLI).

Eng: Mohamed Saied Afify* Cisco CUCM: Supporting End Device.- Understanding Key Device Pool Setting.

- Methods of Adding Phone to CUCM.

Eng: Mohamed Saied Afify* The purpose of device pools:IP Phones have a number of criteria that must be assigned to them:

- List of CCM Servers to use.

- Codec that should be used.

- Time and date information.

- Device Pool group this configuration in to a single assignment.

Eng: Mohamed Saied Afify* Required Device Pool Elements: Device Pool name.

Cisco Call Manager group.

Date / Time group.

Region.

SRST Reference.- Softkey Template

Eng: Mohamed Saied Afify* Methods to add Cisco IP Phones to CCM: Manually: Entering the MAC address and extension of each IP Phone through the CCM admin web page.

- Automatically: CCM hands out extension numbers to newly registered phones like DHCP addresses.

- Bulk Administration Tool: Use Excel spread sheet to generate CSV files of devices.

Eng: Mohamed Saied AfifyCisco CUCM: Supporting End Users

Eng: Mohamed Saied Afify- Bulk Administration Tool.

- Locking down the Cisco IP Phone.

- Cisco CUCM: Supporting End Users.

Eng: Mohamed Saied Afify* Methods to add Cisco IP Phones to CCM: Manually: Entering the MAC address and extension of each IP Phone through the CCM admin web page.

- Automatically: CCM hands out extension numbers to newly registered phones like DHCP addresses.

- Bulk Administration Tool: Use Excel spread sheet to generate CSV files of devices.

Eng: Mohamed Saied Afify* The Cisco Bulk Administration Tool: Help in making large additions or changes to CUCM Database:

- Phone. - Users. - Many (tedious) Configuration.

Now pre- integrated in CUCM administration.

Supports Data export and re- import.

Exported data can also be used in place migration or data restore ( not possible with DRS).

Eng: Mohamed Saied Afify*Understanding BAT Components:- When making Bulk additions, two pieces need to exist:

Eng: Mohamed Saied Afify* Locking down the Cisco IP Phone: Variety of simple to configure security options:

- Disable PC port.

- Lock setting access.

- Gratuitous ARP protection.

- PC voice Vlan access.

- IP Phone web browser access.

Eng: Mohamed Saied AfifyCisco CUCM: Supporting End Users.

Eng: Mohamed Saied Afify* Benefits of CCM User Accounts: Users have the ability to manage their own phone.

Most soft phone devices require user logins.

Gives way to advanced features such as extension mobility.

More sophisticated tracking: per user account.

Eng: Mohamed Saied Afify* Understanding User Accounts: Two categories of users: End users / Application users.

End user can be added to the CUCM database via three main method: - Manual. - BAT. - LDAP.

1- Manual Entry:

- User ID. - Last name. - Presence group. - Remote destination limit. - PIN. - Password.

Eng: Mohamed Saied Afify3- LDAP Lightweight Directory Access Protocol:- LDAP is a standards-based system that allows an organization to create a single, centralized directory information store.

- LDAP holds information about user accounts, passwords, and user privileges.

- CUCM supports LDAP integration with several widely used LDAP systems, including the following: - Microsoft Active Directory (2000, 2003, 2008) - iPlanet Directory Server 5.1 - Sun ONE Directory Server (5.2, 6.x)

- CUCM can interact with LDAP in two ways: - LDAP Synchronization

- LDAP Authentication

Eng: Mohamed Saied Afify* LDAP Synchronization:- When LDAP Sync is enabled, user accounts must be created and maintained in LDAP and cannot be created or deleted in CUCM.

Some user data (but not all) is maintained in LDAP and replicated to the CUCM database.

- The user password must be maintained in both the LDAP system and in CUCM.

Some user attributes are not held in LDAP and are still configured in CUCM because those attributes exist only in the CUCM database.

Eng: Mohamed Saied Afify- LDAP Authentication redirects password authentication requests from CUCM to the LDAP system.

- End-User account passwords are maintained in the LDAP system and are not configured, stored, or replicated to CUCM. * LDAP Authentication:

Eng: Mohamed Saied Afify* User account information: User account information is divided into three categories, with fields for specific data in each category:

1. Personal and Organizational Settings:

- User ID - First, Middle, Last Name - Manager UserID - Department - Phone Number, Mail ID

2. Password Information: Password

3. CUCM Configuration Settings:

- PIN - SIP Digest Credentials - User Groups and Roles - Associated PCs, controlled devices and DNs

- Application and feature parameters (Extension Mobility, Presence Group, CAPF)

Eng: Mohamed Saied Afify* LDAP Sync Mechanism: - The first time the synchronization happens, the following events take place:

- All existing end-user accounts in the CUCM database are deactivated (not deleted).

- Accounts whose CUCM User ID exactly matches a user in LDAP are reactivated, and any settings from LDAP are updated or applied in the CUCM database.

- Accounts that exist only in LDAP are created in the CUCM database.

- Any accounts that remain deactivated (meaning they do not exist in LDAP) are deleted from the CUCM database after 24 hours.

Eng: Mohamed Saied AfifyThe basics steps to set up LDAP Sync are as follows:

1. Activate the Cisco DirSync service.

2. Configure the LDAP system.

3. Configure the LDAP directory.

4. Configure LDAP Custom Filters.* Configure LDAP Sync:

Eng: Mohamed Saied Afify* Managing Groups, Roles and Privileges: Users assigned to groups.

Groups assigned to one or more Roles.

Roles assigned to privileges.UsersGroupsRolesApplicationPrivileges

Eng: Mohamed Saied AfifyCisco CUCM

Understanding Dial Plan

Partitions and Calling Search Spaces

Eng: Mohamed Saied Afify* Understanding Dial Plan: CUCM Route Architecture.

- Building Route Group.

- Building Route List.

- Building Route Pattern.

Eng: Mohamed Saied Afify* CUCM Route Architecture:

Eng: Mohamed Saied AfifyRoutePatternRoute GroupDeviceDeviceDeviceDevice* Call Manger Route Plan Architecture:Route ListRoute Group

Eng: Mohamed Saied Afify* Route Pattern Wildcards:X : Single Digit

@ : North American Numbering Plan

! : One or More Digit

. : Access Code Termination / Digit Formatting

# : Terminates Interdigit Timeout

Eng: Mohamed Saied Afify[xyz] [x-y] [x-yz] = Digit Set

[^ xyz] [^x-y] [^x-yz] = Negative Digit Set

Example:x123

195xx

[139]11

[1-5]00

[13-59]1

[^1-3]11

38[^2-4]3

9011!#

Eng: Mohamed Saied AfifyRoutePatternRoute GroupDeviceDeviceDeviceDevice* Call Manger Route Plan Architecture:Route ListRoute Group2xxx1-WAN2-PSTN1-WAN2-PSTN10.1.1.110.1.1.2WANPSTN10.1.1.110.1.1.22xxx1xxx

Eng: Mohamed Saied Afify- Cisco CUCM: Partitions and Calling Search Spaces Define Partitions and Calling Search Spaces.

Assigning Calling privileges using Partitions and CSS.

Eng: Mohamed Saied Afify* Partitions and Calling Search Spaces:- Partitions: Groups of dialable numbers. - Lines. - Route Patterns. - Any thing that has a number.

Internal-PTLocal-PTINT-LD-PT200120022003..29999.[2-9]xxxxxx9.011!9.1[2-9]xx[2-9]xxxxxx

Eng: Mohamed Saied AfifyInternal- CSS

Internal-PT

2010Out- CSS

INT-LD-PTLocal-PT

- Calling Search Spaces: A list of reachable Partitions.

- assign to any dialing entity.

- Define calling privileges.

Eng: Mohamed Saied Afify* Understanding Partitions and Calling Search Spaces. By default, All numbers assign to the (none) Partitions.

By default, All devices assign to the (none) CSS.

All numbersPartitions: none(none) CSS

None

Eng: Mohamed Saied Afify* Understanding Partitions and Calling Search Spaces. Numbers moved into new Partition become unreachable from the (none) CSS devices.

All numbersPartitions: none(none) CSS

None

9.@EXT- CSS

External-PT

- None2010External-PT

Eng: Mohamed Saied Afify* Practical Partitions and Calling Search Spaces. Three types of calling restrictions should exist in your organization:

- Lobby / Public Phones: Internal extension only.

- Typical users: Internal and Local PSTN.

- Management: Internal , Local and Long distance PSTN.Step1: Create the Partitions.

Step2: Assign number to Partitions.

Step3: Create CSSs.

Step4: Assign CSS to devices.

Eng: Mohamed Saied AfifyStep1: Create the Partitions.- Plan your Partitions a round the calling restrictionsInternal-PTLocal-PTLD-PTStep2: Assign number to Partitions.200120022003..29999.[2-9]xxxxxx9.1[2-9]xx[2-9]xxxxxxInternal-PTLocal-PTLD-PT200120022003..29999.[2-9]xxxxxx9.1[2-9]xx[2-9]xxxxxx

Eng: Mohamed Saied AfifyStep3,4: Create / Assign CSS:Lobby-CSS

-Internal-PT

Local-CSS

-Internal-PTLocal -PTManger-CSS-Internal-PTLocal-PTLD-PTLobby PhoneNormal Users PhoneManager Phone

Eng: Mohamed Saied Afify* Try it yourself:

Eng: Mohamed Saied AfifyStep1: Create the Partitions.- Plan your Partitions a round the calling restrictionsInternal-PTLocal-LD-PTINT-PTStep2: Assign number to Partitions.100210039.[2-9]xxxxxx9.1[2-9]xx[2-9]xxxxxx9.011!Manager-PTInternal-PTLocal-LD-PTINT-PTManager-PT1001

Eng: Mohamed Saied AfifyStep3,4: Create / Assign CSS:Lobby-CSS

-Internal-PT

Employee-CSS-Internal-PTLocal-LD-PTManger-PTManger-CSS-Internal-PTLocal-LD-PTManger-PTINT-PTLobby PhoneEmployee PhoneManager Phone

Eng: Mohamed Saied AfifyCisco CUCM:CUCM Feature Overview

,

Understanding QOS


Call Park.

Call Pickup.

Shared Lines.

Do Not Disturb.

Call Back. Barge and Privacy. Services / Extension Mobility* Cisco CUCM:CUCM Feature Overview

Eng: Mohamed Saied Afify* Quality of Service (QoS): Understanding the enemy.

-The three current models of QoS.

- QoS queuing method.

Eng: Mohamed Saied Afify Understanding the enemy:1- Lack of B.W.

2- Packet Loss.

3- Delay.

- Fixed delay. - Variable delay.

- Jitter.LAN5 MbpsT11.5 Mbps

Eng: Mohamed Saied Afify* Network Requirements For Voice and Video:1- B.W.

2- End - to End delay. (150ms or Less)

3- Jitter. (30ms or Less)

4- Packet Loss. (1% or Less)

Eng: Mohamed Saied Afify* The three current models of QoS: Best Effort.

Integrated Services. (IntServ)

Differentiated Services. (DiffServ)* QoS ToolBelt: Classification: Involves identifying and grouping different traffic types.

Marking: Tages or colors the packet so it can be quickly recognized elsewhere in the network.

Eng: Mohamed Saied Afify* Queuing Algorithms: Weighted Fair Queuing. (WFQ)

Class-Based Weighted Fair Queuing. (CBWFQ)

Low Latency Queuing. (LLQ) (PQ-CBWFQ)50 Packet http20 Packet ftp10 Packet telnet

Eng: Mohamed Saied AfifyCisco Unity Connection (CUC)

,

Cisco Unified Presence (CUPs)

Eng: Mohamed Saied Afifywww.youtube.com/MohamedSaiedAfify

www.facebook.com/VoipLovers

VoiceLover

www.twitter.com/VoipLovers

Eng: Mohamed Saied Afify* Overview of Cisco Unity Connection: CUC is one of 5 VOIP linux appliances.

CUC integrates with legacy PBX systems via PIMG or TIMG.

CUC integrates with CUCM using SCCP or SIP.

- CUC users ( Manually, CSV, CUCM import, or LDAP ).

Up to 2000 mail boxes per servers.

Access voice mail from anywhere.

VPIM support.

Active / Active high availability.

Eng: Mohamed Saied Afify* How CUC Integrates with CUCM?SCCPSIP

Eng: Mohamed Saied Afify* How CUC Processes Calls?- All inbound calls to CUC are handled by a series of call handlers:

1- System call handlers.

2- Directory handlers.

3- Interview handlers. Call Routing:Two primary call routing are built in CUC:

1- Direct call.

2- Forward call.

Eng: Mohamed Saied Afify* Managing Users and Mailboxes in CUC: User templates make configuration easier.

User templates basic:

1- Name. 2- Phone. 3- Location.

COS defines many options ( timers, features, restriction).

User creation options:

1- Manually. 2- Bulk administration. 3- Import from CUCM. 4- Import from LDAP.

Eng: Mohamed Saied AfifyCisco Unified Presence (CUPs)

Eng: Mohamed Saied Afify* What is the point of CUPs? CUCM include basic presence feature (ON/OFF hook).

CUPs add in: - Additional state ( Available, A way, Busy). - Additional method ( Desk, Mobile Phone, IM, Conf). - Calendar integration (Status Auto- Updates). - Enterprise instant message (IM).

Brings Cisco Unified Personal Communicator (CUPC) to the table.

CUPC supports Desk phone, Soft phone modes.

Eng: Mohamed Saied Afify-Desk phone mode:

Eng: Mohamed Saied Afify- Soft phone mode:

Eng: Mohamed Saied Afify* How CUPs / CUPC Communicate: Simple / XMPP- Presence information.

SOAP- CUCM database access.

CTIQBE office communicator integration.

LDAP user database integration.

Eng: Mohamed Saied Afify* CUCM Configuration of Unified Presence:1- Assign user license in CUCM.

2- Associate users with phone / lines in CUCM.

3- Enable CTI on phone / line.

4- Create CSF device or UPC device in CUCM.

5- Associate user with CSF device.

Eng: Mohamed Saied Afify* CUPs Configuration of Unified Presence: 1- Create link to voice mail server.

2- Create link to CUCM server ( CTI gate way).

3- Add LDAP server.

4- Define the CCMCIP profile.

Eng: Mohamed Saied Afifywww.youtube.com/MohamedSaiedAfify

www.facebook.com/VoipLovers

VoiceLover

www.twitter.com/VoipLovers

Eng: Mohamed Saied AfifyVoice Troubleshooting

Eng: Mohamed Saied Afify* Voice Troubleshooting: Cisco CME Getting in Troubleshooting groove.

Troubleshooting registration issues.

Troubleshooting Dial-Plan issues.

Eng: Mohamed Saied AfifyTroubleshooting: Getting in Troubleshooting groove. Step 1: Define the problem.

Step 2: Gather the facts.

Step 3: Consider the possibilities.

Step 4: Create an action plan.

Step 5: Implement the action plan.

Step 6: Observe results.

Step 7: If resolved, document.

Eng: Mohamed Saied Afify-Troubleshooting registration issues:1- Switch detects un powered device, Supplies power.

2- Switch provides VLAN information to the IP Phone.

3- Phone sends DHCP request, receives IP and TFTP info.

4- Phone contracts TFTP servers, retriever configuration file.

5- Phone contracts CME route listed in configuration file.

Eng: Mohamed Saied Afify* Troubleshooting Dial-Plan issues: Understanding how Dial-Peers work.

Use some key troubleshooting commands: - Show IP interface brief.

- Show Dial- Peer voice summary.

- debug VOIP Dial- Peer.

Eng: Mohamed Saied Afify* Voice Troubleshooting: Cisco CUCM CUCM common issues.

CUCM additional troubleshooting utils.

Eng: Mohamed Saied Afify* Troubleshooting registration using reports: Registration problems . Where to look:

- Device configuration.

- Route plan report.

Understanding reports:

- Finding the CUCM reports.

- CDR analysis and reporting (CAR).

Eng: Mohamed Saied Afify* Additional CUCM utilities to save the day:1- Cisco Real Time Monitoring Tool ( RTMT).

2- Cisco Disaster Recovery System (DRS).


**********************ISDN Non-Facility Associated signaling (NFAS) allows multiple ISDN PRIs to be controlled by a single D channel with an associated D channel for backup. ***********************************************************************************************************************************************************************************************************************************************************************

Ccna Voice 640-461

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