CCNA VOICE

CCNA Voice

Abdul Jaseem VP
CCNA VOICE Guide 1.0 (43 Pages)

Warning!
Reproduction or copying exact content from this guide strictly prohibited.

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 Traditional Voice System

 3 Types of Analog Signaling
 2 Types of Signaling Methods (Loop Start, Ground Start)
 Understanding PSTN, its Components and Connection to PSTN
 Understanding PBX
 --PSTN Numbering Plan (Country Code, Area Code, Site code, Subscriber Number)
 Why VoIP?
 Process of Converting Analog Signals to Digital Signal
 Codecs
 DSPs
 Understanding components of Unified Communication
 CUCM Version history
 Function & Features of CUCM
 CUCM Hardware requirements
 CUCM Installation (CUCM7)
 CUCM CLI
 CUCM Web Administration
 Understanding IP Phones
 --Other VoIP Devices (ATA, VG224, VG248, VT Advantage, VT Camera)
 IP Phone Registration
 Phone Button Template
 Softkey Template
 --CUCM Cluster
 --CUCM Call Processing
 Device Pool
 IP Phone Boot up Process
 CUCM Call Flow
 User Accounts
 End User Configuration
 LDAP
 CUCM User Phone Features
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 SIP Phone Registration

 DNA
 CAR
 Cisco RTMT
 IP Phone Web Access

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Traditional Voice System

 Traditional telephone network has been placed in 1900s.
 An analog phone uses properties of electricity used to convey audio
information over cables.
 When you speak in to an analog phone, the sound that come out of your mouth
are converted to a form of electricity.
 For a voice call to occur, certain informations must be passed between the
telephony device and the initiator such as on-hook, off-hook, availability,
dialing, Ringing etc. These are referred as SIGNALING.

3 Type of Analog Signaling

 For a voice call to occur, certain informations must be passed between the
telephony device and the initiator such as on-hook, off-hook, availability,
dialing, Ringing etc. These are referred as SIGNALING.
1. Supervisory Signaling: Involves the detection of the change of status of a
loop or trunk. (ON HOOK, OFF HOOK, RING)
2. Address Signaling: Involves passing dialed digits (PULSE, DTMF)

3. Informational Signaling: Provide audible tone to user, which indicates

certain conditions (Dial tone, Busy, Ring Back, Congestion, Conformation)

FXO, FXS Signaling Methods

FXO & FXS indicates pass signaling via two methods.
Loop Start (PSTN)

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 When handset picked up (off-hook) connects the 48V circuit that draws
current from the device (Router, PSTN, etc.) & indicates the change in status
 Router or PSTN provide dial-tone to the end device
 User dial the number and destination Rings
 Destination off hook, (Router, PSTN) bridge 2 calls each other
 (Glare issue: when user dial & other used off-hook same time)

Ground Start (PBX)

 It requires ground voltage detection to create loop.
 Used in trunk lines
(Note: Loop start & Ground start are less important in VoIP since these are too old
technologies, we have to configure only what matches the other side)

Analog Connectivity Issues

 Distance limitation (Signal strength low, need repeater, amplify noise)
 Wiring limitation (1 Call per line)

Digital Connections and Signaling

 It uses TDM to transmit voice traffic.
 T1 (24 channels) and E1 (32 Channels) digital circuit
 CAS, CCS, ISDN
 (Refer CVOICE Guide)

Understanding PSTN
 When phone system was originally created, individual phones were wired
together to allow communication. If you wanted to connect more than one
person you need multiple cables and ports. PSTN eliminates this difficulty.
 Public Switched Telephone Network, PSTN is a huge voice network having dial
plan to all over the world.
 PSTN is a network of networks, which connects multiple telecom providers
together in to a massive worldwide network.
 All the telecom service provides in the world communicate together via a
common protocol called SS7 (Signaling System 7)

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 When user makes a call, the first CO performs an SS7 lookup to locate
the destination, once found SS7 responsible for call routing through the
voice network

Components of PSTN

 Analog Telephone: Convert audio in to electrical signal. Able to connect

directly to PSTN
 Local Loop: Link between customer premises and telecom service provider
 CO Switch: Central Office Switch provides services to the local loop,
includes Signaling, Digit Collection, Call routing, Call setup & tear down
 Trunk: Provide connection between CO Switches or Private switches (PBX)
 Private Switch (PBX): Allows business to operate miniature PSTN inside the
company. Cost saving for nearby calls.
 Digital Telephone: Typically connects to PBX systems. Convert audio in to
binary values.
(Note: Many believe that PSTN will eventually be absorbed to Internet)

Understanding PBX
 Many organizations have thousands of phones internally, if they purchase
direct PSTN connections the cost would be huge! PBX (Private Branch Exchange)
eliminates this issue. PBX allows to internally manage in-house telephones.
 It gives flexibility to internal users to make calls without using PSTN
resources (free internal extension to extension call)
 External calls handled by Trunk lines.

PBX Components
 Line Card: Provide connection between telephone and PBX
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 Trunk Card: Provide connection from PBX to PSTN or to other PBX
 Control Complex: Intelligence behind PBX system. Call Routing, setup etc.

PSTN Connections

--PSTN Numbering Plan (Country Code, Area Code, Site code, Subscriber
Number)

Why VoIP?
 Convert analog voice signal to digital and then placing the binary values as
data packet with IP Addressing headers and finally sending over data network.
 Reduced cost of communication: VoIP calls are absolutely free of cost
(internet or WAN charges are applicable)
 Reduced cost of cabling: Single Ethernet cable carries voice & Data
 Take your Phone with you (Mobility)
 IP Soft Phone
 Feature-rich communication: Voice, Data & Video combined together (E.g.: IVR)
 Open Standard: PBX are basically proprietary locked. VoIP uses ICT/IP
standard

Process of Converting Analog Signals to Digital Signal

1. Sampling
Sampling is the process of cutting an audio stream

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Human ear : 20 – 20,000 Hz
Human Speech : 200 – 9,000 Hz
Telephone Channel : 300 – 3,400 Hz (Enough to convey mood of the caller)
Nyquist Theorum : 300 – 4,000 Hz (2x4000 = 8000 Samples per sec)

2. Quantization
Assigning values to sampled values.
A sample is a numeric value consumes a single byte (8 Bit) of information.

3. PCM (Pulse Code Modulation)

Convert the quantized value to digital value of 8 bits
8 Bit can represent 0 – 255, even though Quantization limited to +127 to -127
1st bit used to represent the polarity
a Law (Everywhere) : 1 = +ve; 0 = -ve
µ Law (US, Japan, Canada) : 0 = +ve; 1 = -ve
8000 Samples x 8 bits = 64000 = 64Kbps (Band Width Standard Voice Call)

4. Compress (Optional)
Send a single sound sample and tell the remote device to continue playing that for
certain time. It reduces the bandwidth.
Codecs based on compression technique
Common compression: G.729 = 8 Kbps

CODEC (COder-DECoder / COMpression-DECompression)

An audio codec is a device or computer program capable of coding or decoding a
digital data stream of audio. The way our voice is coded to packet
CODEC Sampling B/W MOS Payload Sample Size (ms)
G.711 8000 Hz 64Kbps 4.3 240/160 30/20 Including TCP/UDP header
total ≈ 80Kbps
G.722 16000 Hz 64Kbps 4.5 Wideband Codec
G.729 8000 Hz 8Kbps 3.92 30/20 30/20 More DSP used, high
complexity codec
G.729A 8000 Hz 8Kbps 3.7 Less DSP requirement
iLBC 8000 Hz 15.2Kbps 4.1 50/38 30/20 Industry standard, Open
source
G.726 : 32Kbps MOS = 3.85
G.728 : 16Kbps MOS = 3.61
Default VoIP Sample Size = 20mSec

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DSPs (Digital Signal Processors)

Basic function of a router to route traffic and it is not a processor intensive
task. VoIP is processor demanding task, here comes DSPs to offload the processing
responsibility of voice related task from the processor of the router.
A DSP chip performs all Sampling, Encoding and compressing functions.
VIC + DSPs = Processing Voice to Packets
Cisco bundles these chips to PVDMs (Packet Voice DSP Modules)
PVDM2-8 Provides 0.5 (Half) DSP chip
PVDM2-16 Provides 1 DSP chip
PVDM2-32 Provides 2 DSP chip
PVDM2-48 Provides 3 DSP chip
PVDM2-64 Provides 4 DSP chip
DSP consumption is based on the Codecs

Packetizing an Audio (Audio Payload)

1 Sec Audio → 8000 Samples →
1 Sec Audio → 20mSec Packets → 50 Packets Per Sec
8000/50 = 160 Samples in a Packet
Such a packet is called Audio Payload

RTP (Real Time Transport Protocol)

 Protocol to send the codec (Protocol of voice) carry Audio Payload between
devices
 RTP operates at the transport layer OSI model.
 RTP is one way stream only. In a 2 way conversation, there will be dual RTP
stream between the devices.
 RTP Port Numbers: Random even UDP ports from 16,384 to 32,767

Audio Payload UDP RTP IP Data Link

Payload Type Sequence Number Time Stamp

1. Payload Type: Specify Audio or Video communication

2. Sequence number: Allows remote device to put the packet back in the order
3. Time Stamp: Sampling Size 20msec or 30mses
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RTCP (Real Time Transport Control Protocol)

 At the time RTP starts, RTCP also engages. Its primary job is statistics
reporting which includes Packet Count, Packet Delay, Packet Loss, Jitter
(delay variations)
 RTCP Port Numbers: Random odd UDP ports from 16,384 to 32,767
 RTCP packets will be exchanged between every 5 seconds

--CUCM Cluster

--CUCM Call Processing

Understanding Components of Cisco Unified Communication (CME, CUCM-BE,
CUCM)
Cisco Call Manager Solutions/ Voice Management Products (IP PBX)
1. CME (450 Limited end devices, no redundancy, router hang, CLI Based also GUI-CCP)
2. CUCMBE (1000 devices, Software in as server platform, No redundancy)
3. CUCM (30,000 devices, Dedicated Call Processing solution)

Understanding CUCM (Cisco Unified Communication Manager)

CUCM Version History
Selsius developed call Manager later Cisco purchased → Cisco Call Manager V2.4
Version Platform Hardware
V2.4 Windows NT Any Hardware
V3.X Windows 2000 MCS Servers only
(Cisco Approved Windows
Hardware IBM/HP)
V4.X Windows 2000 MCS
V5.X Package installation on Linux MCS
V6.X Linux Based OS MCS Linux
V7.X Virtual Machine Appliance on MCS
VMWare ESXi
V8.X Virtual Machine Appliance on Cisco Unified Virtual Machine
VMWare ESXi Computing System
(UCS)

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Features of CUCM

 Mind of Voice network

 Full support for Audio/ Video telephony
 Built in disaster recovery system
 Directory support (IBM IDS to Windows ADDS)
 Server based cluster deployment
 Device Control
 Dial plan & Permisions
 Phone feature and control
 20 Server per cluster (Publisher, Subscribers (more), TFTP, DHCP, DNS,
Application (Voice Mail), Media resources)
Functions of CUCM
Call Processing
Signaling end devices
Phone feature Administration
Directory Services
Link to external applications
CUCM Hardware Requirements
CUCM 7.X CUCM 8.X CUCM 9.X CUCM 10.X CUCM 11
2 GHz Processor 2.4 GHz Processor
2 GB RAM 6 GB RAM
72 GB Hard Disk 72 GB Hard Disk

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CUCM Installation
1. Setting Virtual Machine (Virtual Hardware)
Create New Virtual Machine → Typical Next → Select ISO → Credentials (Linux Root) →
VM Name, Location → 80 GB → Next → Customize Hardware (RAM 2GB, Bridge) → Finish
2. Installing CUCM
Media Check No → Agree → OK → Install → Proceed → Upgrade No → Windows data Import
No → Continue → Time Zone (Asia/ Culcutta) → Automatic Negotiation of NIC Yes →
Change MTU Size NO → Use DHCP NO (Host Name, IP, Mask, Gateway) → DNS NO →
Credential (Platform Administration) → Certificate Info → 1st Node YES → NTP (Real
time YES/ LAB NO) → Clock Info → Database Access Password (For DB replication,
Communication between nodes) → SMTP NO → Web Page Credentials → Platform
Configuration Complete OK

CUCM CLI (10%)

CLI is accessed via SSH protocol using terminal programs (Putty, Secure CRT, Tera
Term, etc.)
1. admin:show network eth0 - for getting the IP of CUCM
2. admin:set network ip eth0 142.100.69.10 255.255.0.0 - Change IP
3. admin:set network gateway 142.100.69.11 - Change Gateway
4. admin:set network status eth0 up - Enable network Adapter
5. admin:set network status eth0 down - Disable network Adapter
6. admin:utils network ping 142.100.0.1 - Ping
7. admin:utils network traceroute - Tracert
8. admin:set network domain TEST.COM - Set Domain Name
9. admin:set network hostname TEST_CUCM - Set Host name for CUCM
10. admin:set network dns primary 142.100.69.11 - Set Primary DNS
11. admin:set network dns secondary 142.100.69.12 - Set Secondary DNS
12. admin:utils service restart Cisco Tomcat - Restart web Services
13. admin:utils service list - List all Services
14. admin:utils system restart - Reboot CUCM
15. admin:utils system shutdown - Shutdown CUCM

CUCM Web Administration (90%)

Support IE, Firefox. Doesn’t Support Google Chrome
Go to browser https://[IP ADDRESS_OF_CUCM]/ccmadmin

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5 Consoles in CUCM
1. Cisco Unified CM Administration: Core/ Day to day Administration
2. Cisco Unified Service Ability: For Services, Issues Debug
3. Cisco Unified OS Administration: Like an OS (Change IP, Ping, etc)
4. Cisco Unified Disaster Recovery: Backup & restore DB
5. Cisco Unified Reporting: Report, Status report, Call Duration, etc

1. CUCM Admin Interface

This is the main administration of CUCM
System Menu:
Includes tasks for the configuration of Subscribers, CM Group, Device Mobility
Group, Device Pool, Region, Location, SRST, Service Parameter, Enterprise Parameter
Call Routing Menu:
Includes tasks to define call routing system, Call hunting, Class of Control,
Features (Intercom, Call Park, Call Pickup, etc.)
Media Resources:
For configuring MOH, Annunciator, Media Termination Point, Transcoder etc.
Advanced Feature:
For configuring Voice Mail Integration, Extension Mobility, VPN Features etc.
Device Menu:
For configuring Gateways, Gate Keeper, Trunks, IP Phones, Remote Destination,
Device Settings, Phone button and Soft key templates etc.
Application Menu:
CUCM Assistant configuration wizard and Plugin downloads
User Management Menu:
For configuring Application User, End User, User Group and Roles
Bulk Administration Menu:
Perform bulk operation for repetitive tasks
Help Menu:
Access to local searchable help files
2. CUCM Service Ability Interface
For activating and deactivating some basic services in CUCM and troubleshooting
Alarm Menu:
Option for system monitoring and performance, health monitoring
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Trace Menu:
Configuring and troubleshooting trace settings
Tools Menu:
Access to CDR Analysis, Service Activation, Control Centre Feature Service, System
logs
SNMP Menu:
Configuring SNMP (Simple network Management Protocol) and its authentication
Help Menu:
Access to local searchable help files
3. CUCM OS Administration Page
Here administrator can monitor and interact to the Linux Based OS such as
 Monitor hardware resource (CPU, Disk Speed) utilization
 Check upgrade software version
 Configure NTP (Network Time Protocols) server IP
 Manage server security, Digital Certificate
 TAC Remote Assistance account
 PIG to other devices
4. Disaster Recovery System Interface
Provides DRS backup and restore capability
5. CUCM Reporting Interface
Method to access system reports, logs, issue logs.

Understanding Cisco IP Phones

IP Phone Front Panel
Display Standard display for IP Phone
MWI Message Waiting Indicator: Ringing alert, MSG alert (Blink)
Line Buttons To assign DNs or Speed dial, Special functions etc. Line 1
must be a DN always (Right side to display)
Soft Keys To assign special features (Redial, Transfer, Hold, etc.)
Selector Button To navigate through different options
Dial Pad 0 to 9 and *, # numbers
Directory Button To access cooperate directory
Message Button To access messages
Service Button To access services offered by the company
Settings Button Check IP, Reset, Factory restore, etc.
Help (Info) Button Help topics
Volume UP/ Down To adjust volume
Headset Button To enable headset

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Mute Button To Mute

Speaker To activate speaker phone
IP Phone Back Panel
Auxiliary Port (RS232) To connect expansion module
10/100 SW To connect IP Phone to the switch
(Network)
10/100 PC To connect co-located PC (IP Phone
acts as a mini switch)
Power Port 48V DC Power (If there is no PoE)
Headset Port To connect Cisco Headset (RJ11)
Handset Port To connect receiver handset
Types of IP Phones
7905 Single Line button, No display, No Auxiliary & PC Port
7906 Single Line button, Small display, No Auxiliary & PC Port
7910, 7911, 7912 Single Phone button, Small display, PC Port, No Auxiliary
Port
7920, 7921, 7922 Single Phone Button, IP Based Mobile phones (Wi-Fi)
7935, 7936 1 Line buttons, Auxiliary & PC port, For Conference
7940, 7941, 7942, 7945 2 Line buttons, Auxiliary & PC port, Big Screen
7960, 7961, 7962, 7965 6 Line buttons, Auxiliary & PC port, Big Screen, 7965 is
Color
7970, 7971, 7972 8 Line buttons, Auxiliary & PC port, Big Screen, Color
7985, 7986 8 Phone Button, Auxiliary & PC port, Big Screen, Color,
Support video
Exp Modules 7914, 7915, These are not phones, but Expansion modules. 14 Line
7916 buttons, Uses Auxiliary port to interface. We can add 2
exp module to an IP Phone
8800 Smart Phones
8900 Smart Phones, Video Support, Inbuilt camera
9900 Smart Phones, Video Support, Inbuilt camera

Protocols & Services used by an IP Phone

IP Phones requires set of Protocols and services
 NTP (Network Time Protocol)
 CDP (Cisco Discovery Protocol)
 DHCP (Dynamic Host Configuration Protocol)
 TFTP (Trivial File Transfer Protocol)
 PoE (Power Over Ethernet)
 DNS (Domain Name System)
NTP (Network Time Protocol):

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IP standard provides network based time synchronization. It ensure same
time on all devices based on time different time zones. Time synchronization is
important for many functions to operate
CDP (Cisco Discovery Protocol):
Is a cisco proprietary L2 protocol provides network mapping information for
directly connected Cisco devices
DHCP (Dynamic Host Configuration Protocol):
DHCP is faster, easier, widely accepted method to distribute IP information to the
clients. It provide following information to the IP Phones
 IP Address
 Subnet Mask
 Default Gateway
 DNS Server
 TFTP Server IP
DHCP can be provided by an existing DHCP server or Local router or CUCM itself
TFTP (Trivial File Transfer Protocol):
IP Phones utilize TFTP to download their configuration files, firmware images etc.
Normal TFTP server can’t fulfill IP Phones requirement, hence we must have CUCM
TFTP server. Server uses Port 67 and client uses port 68.
DORA Process in DHCP
Discover: Broadcast: Can somebody give me an address?
Sender 0.0.0.0 UDP 68, Receiver 255.255.255.255 UDP 67
Trying to discover whether there is any DHCP Server
Offer: Broadcast: I can give you 192.168.1.2, Mask, G/W, DNS, Lease Duration
Sender 192.168.1.1 UDP 67, Receiver 255.255.255.255 UDP 68
Request: Broadcast: I want to accept the offer, allow me to use 192.168.1.2 for
3600 seconds. Client takes the address offered by the Server pool
Sender 0.0.0.0:68, Receiver 255.255.255.255:67
ACK: Unicast to confirm the acceptance of provided IP info
PoE (Power Over Ethernet):
Provides DC power over Ethernet cabling. Helps less cabling efforts and cost saving
(Wiring, No Power supply needed)
DNS (Domain Name System)

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DNS provides IP to Domain mapping and vice versa. It is not critical to IP
Phones. ADNS server must be external to the CUCM Cluster; DNS is not a service that
CUCM can offer.

---Other VoIP Devices

1. ATA
2. VG224: 2 Ethernet Ports + 24 FXS Ports
3. VG248:
4. Cisco VT Advantage
5. Cisco VT Camera

IP Phone Registration in CUCM

IP Phone registration (Manual)

Voice Enabled
Switch

CUCM
IP: 142.100.69.10/16

User PC
IP: 142.100.69.5/16

Step 1: Service Activation:

Cisco Unified Services Ability → Tools → Service Activation
• Cisco CallManager
• Cisco Tftp
• Cisco DHCP Monitor Service
→ Save
Go to Tools → Control Center – Feature Services check the service activation
status. It should be 0 days always. We can Stop, Start, Refresh those services from
here.
Step 2: Configure CUCM DHCP Server (if there is no DHCP Server)
Cisco DHCP Monitor Service must be enabled

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Cisco Unified CM Administration → System → DHCP → DHCP Server
Find → Add New
Host Server (Select) → Save

System → DHCP → DHCP Subnet

Find → Add New
DHCP Server* <Select_SERVER_NAME>
Subnet IPv4 Address* <142.100.0.0>
Primary Start IPv4 Address* <142.100.0.50>
Primary End IPv4 Address* <142.100.0.100>
Primary Router IPv4 Address <GATEWAY:142.100.69.254>
IPv4 Subnet Mask* <255.255.0.0>
Primary TFTP Server IPv4 Address (Option 150) <142.100.69.10>
→ Save
IP Helper Address for DHCP Broadcast
Router(config)#interface ethernet 1/0
Router(config-if)#ip helper-address 142.100.0.254
Step 3: IP Phone Registration (Manual)
Go to Device → Phone → Find → Add New
Phone Type* (Select type) → Next → Select the device protocol: SCCP → Next
Device Information
Device Name* <MAC_ADDRESS(SEP005056C00001)>
Description <Cisco IP Communicator>
Device Pool* <SELECT_POOL_OR_DEFAULT>
Phone Button Template* <Standard CIPC SCCP_OR_SELECT>
Softkey Template <SELECT_FROM_LIST>
Common Phone Profile* <Standard Common Phone Profile>
Protocol Specific Information
Device Security Profile* <Standard SCCP Non-Secure Profile>
→ Save
Association Information
Line [1] - Add a new DN → Directory Number Information
Directory Number* <1000> → Save
Alert Name : Called party Name when the phone is Ring back. If Phone A calling
Phone B, “Calling NAME B” is displayed on Phone A when alert name
configured on Phone B.
Display Name : Internal caller ID, appear on the remote phone
Go to Device → Phone to see registered phones

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IP Phone Registration Automatic

Step 1 & 2 is same
Remove Unassigned DN from Call Routing → Route Plan Report → Unassigned DN → Find →
Select All → Delete
Cisco Unified CM Administration → System → Cisco Unified CM → Find
Select Server
Under Auto-registration Information
Starting Directory Number* <1000>
Ending Directory Number* <1010>
Remove check mark from
Auto-registration Disabled on this Cisco Unified Communications Manager
→ Save
Go to Device → Phone to see registered phones
-----------------------------------------------------------------------------------
• Cisco Unified CM Group → Check → Auto-registration Cisco Unified Communications
Manager Group (All the auto reg phones comes under this group)
• Enterprise Parameters → Auto Registration Phone Protocol* → SCCP
• Device → Device Settings → Device Defaults (We can change the ‘Device Pool’ and
‘Phone Button Template’ for Auto reg phones)
-----------------------------------------------------------------------------------
IP Phone Registration (BAT) (Check after Device Pool & Button Template)
BAT is a fast way to add, remove, modify database entries for almost every part of
CUCM Database.
1. Service Activation
Cisco Unified Serviceability → Tools → Service Activation → Cisco Bulk Provisioning
Service
2. Creating Phone Template
Cisco Unified CM Administration → Bulk Administration → Phone Template → Find → Add New →
Phone Type → <Cisco 7960> → Next → Select the device protocol: SCCP → Next →
Template Name* <7960_SCCP_FINANCE_DEPARTMENT_PHONES>
Device Pool* <BANGALORE_DP>
Phone Button Template* <Standard 7960 SCCP>
Common Phone Profile* <Standard Common Phone Profile>
Device Security Profile* <Cisco 7960 - Standard SCCP Non-Secure Profile>
→ Save
Click Line [1]

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Line Template Name* <LINE_7960_SCCP_FINANCE_DEPARTMENT_PHONES>
→ Save
Related Links → Configure Device (7960_SCCP_FINANCE_DEPARTMENT_PHONES) → Go
→ Save
3. Creating Phone File Format
Bulk Administration → Phone → Phone File Format → Create File Format → Find →
Add New (or copy ‘Simple Phone Format’ then edit)
Format Name* <7960_SCCP_FINANCE_DEPARTMENT_PHONE_FILE_FORMAT>
Device Fields: Device Name, Description
Line Fields: Directory Number
IP Phone Service Maximums: Maximum Number of Lines: 1
→ Save
4. View File Format
Bulk Administration → Phone → Phone File Format → Add File Format →
Format File Name* <7960_SCCP_FINANCE_DEPARTMENT_PHONE_FILE_FORMAT>
Then click (View File Format) link, copy the content to a notepad file.
DEVICE NAME,DESCRIPTION,DIRECTORY NUMBER 1
Sample: (7960_SCCP_FINANCE_DEPARTMENT_PHONES_BAT.txt)

DEVICE NAME,DESCRIPTION,DIRECTORY NUMBER 1

02004C4F4F51,SALES1,1001
02004C4F4F52,SALES2,1002
02004C4F4F53,SALES3,1003
02004C4F4F54,SALES4,1004
02004C4F4F55,SALES5,1005
5. Upload File .txt file
Bulk Administration → Upload/Download Files → Add New →
File: → Browse → 7960_SCCP_FINANCE_DEPARTMENT_PHONES_BAT.txt
Select The Target* <Phones>
Select Transaction Type <Insert Phones - Specific Details>
(Optional ‘Overwrite File if it exists.**’)
→ Save
6. Validate Phones
Bulk Administration → Phones → Validate Phones →
Validate Phones Specific Details
File Name* <7960_SCCP_FINANCE_DEPARTMENT_PHONES_BAT.txt>
Phone Template Name* <7960_SCCP_FINANCE_DEPARTMENT_PHONES>
Job Description <7960_SCCP_FINANCE_DEPARTMENT_PHONE_VALIDATE>

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Bulk Administration → Job Scheduler → Find → Click Job ID → Log File Name and
check the status
7. Insert Phones
Bulk Administration → Phones → Insert Phones →
File Name* <7960_SCCP_FINANCE_DEPARTMENT_PHONES_BAT.txt>
Phone Template Name* <7960_SCCP_FINANCE_DEPARTMENT_PHONES>
(Optional: Override Options)
Job Information
Job Description <7960_SCCP_FINANCE_DEPARTMENT_PHONES_INSERTING>
Run Immediately <Check>
→ Submit
Bulk Administration → Job Scheduler → Find → Click Job ID → Log File Name and check the
status
Go to Device → Phones to see registered Phones.

Phone Button Template

Defines the behavior of Phone Buttons. Two buttons are assigned for DN by default,
but we can customize our own.
Device → Device Settings → Phone Button Template → Find → Copy Standard 7960 SCCP → Button
Template Name* <Type 1 Standard 7960 SCCP> → Save →
Button Information
Button Feature Label
1 Line** MY NUMBER
2 Speed Dial <SPEED DIAL>
3 Privacy <Privacy>
4 Service URL <Service URL>
5 Speed Dial BLF <Speed Dial BLF>
6 Call Park BLF <Call Park BLF>
→ Save
Device → Phone, Select phone by clicking in the device Name
Device Information
Phone Button Template* <Type 1 Standard 7960 SCCP> - The one we created!
→ Save
New Button template has been applied; we can apply this template while registering IP
Phones also.
Configure Template for Particular Phone
Device → Phone, Select phone by clicking in the Device Name again

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Association Information
Select buttons
2. Speed Dial Settings
Speed Dial Settings
Number Label ASCII Label
1 1003 MANAGER MANAGER
→ Save → Close
Abbreviated Dial Settings
Number Label ASCII Label
2 1004 USER1 USER1
AbbrDial softkey is used to dial Abbreviated speed dial
4. Add a new SURL
5. Add a new BLF SD (Busy Lamp Field / Speed Dial)
Destination Directory Number Label Label ASCII
1001 <None> MANAGER MANAGER
→ Save → Close
6. Add a new BLF Directed Call Park (Busy Lamp Field/Directed Call Park Button Settings)
Directory Number:Partition Label Label ASCII
PARK1 PARK1
→ Save → Close

Softkey Template
Controls what Softkey button functions are available to the user (Redial, Hold,
Conference, Transfer, Park, Mobility etc.)
Device → Device Settings → Softkey Template → Find
Copy ‘Standard User’
Name* <SPECIAL_USER>
Description <FOR_SPECIAL_USERS>
Related Links: Configure Softkey Layout → GO
Select a call state to configure (ON HOOK, OFF HOOK, ON HOLD, CONNECTED, etc.) and assign
Required Softkeys.
→ Save

Device Pool
It provides a set of common configuration to a group of devices. Typically we are creating
one devoice pool per an area (but we can create more if needed).
System → Device Pool to see the existing device pool
The components of device pool is given below.

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 CCNA VOICE

1. Cisco Unified CM Group

It defines a top-down ordered list of redundant subscriber servers to which the
phone can register. It includes maximum of 3 subscriber server plus optional SRST
reference. The goal of CUCM Group is to provide server redundancy for IP Phones.
SUB-SERVER1 Primary Call Manager
SUB-SERVER2 Secondary Call Manager
SUB-SERVER3 Tertiary Call Manager
Phone initially register with Primary Call Manager if it goes down, IP Phone
contact Secondary Call Manager and so on.
Configuration
System → Cisco Unified CM Group → Find → Add New
Put Name to the Group
Cisco Unified Communications Manager Group Members
Available Cisco Unified Communications SUB-SERVER1
Managers SUB-SERVER2
SUB-SERVER3
SUB-SERVER4
▼▼▼▼▼▼ (Select & Pull-down)
Selected Cisco Unified Communications Managers <SELECTED_SUB-SERVER1>
(CUCM Group 1) <SELECTED_SUB-SERVER2>
<SELECTED_SUB-SERVER3>
Selected Cisco Unified Communications Managers <SELECTED_SUB-SERVER2>
(CUCM Group 2) <SELECTED_SUB-SERVER1>
<SELECTED_SUB-SERVER3>

NB: If we check Auto-registration Cisco Unified Communications Manager Group, all the Auto
Reg phones comes under this group

2. Date & Time Group

If the devices are not in NTP server’s time zone, the displayed time will be wrong.
Date/Time Group offsets the correct time learned via NTP to match local time zone where
the device is located. Also we can specify the format of time zone.
Configuration
System → Date and Time Group → Find → Add New
Group Name* BANGALORE_TIME_1
Time Zone* Indian Standard Time - (GMT+5:30)
Separator* / (Slash)
Date Format* D/M/Y
Time Format* 12-hour

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 CCNA VOICE

3. Region:
It used to set bit rate for a call. Same region uses High Bit Rate Codec and region
to other region uses low bit rate codec. ( Same region call: G.711, Region to Other
region: G.729)
Configuration
System → Region → Find

Add New
Name* BANGALORE_REGION → Save
Modify Relationship to other Regions
BANGALORE_REGION G.711
→ Save

Add New
Name* CHENNAI_REGION → Save
Modify Relationship to other Regions
BANGALORE_REGION G.729
CHENNAI_REGION G.711
→ Save

Add New
Name* DELHI_REGION → Save
Modify Relationship to other Regions
BANGALORE_REGION G.729
CHENNAI_REGION G.729
DELHI_REGION G.711
→ Save

4. Location:
To set Bandwidth, defines maximum amount of band width used by calls from a
location. Each call is tracked and bandwidth used is deducted from the total of
that location. If bandwidth of the particular location exceeds the call may dropped
or routed to PSTN.
Configuration
System → Location → Find → Add New

5. SRST Reference
System → SRST → Find → Add New
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 CCNA VOICE
Name* <BANGALORE_GATEWAY_SRST>
Port* 2000
IP Address* <142.100.69.254>
→ Save

6. MGRL
7. Auto Registration CSS (Auto Call to Admin)
8. LRG
Configure Device Pool
System → Device Pool
Device Pool Settings
Device Pool Name* BANGALORE_DP
Cisco Unified Communications Manager Group* BANGALORE_GROUP1
Roaming Sensitive Settings
Date/Time Group* <BANGALORE_TIME_1>
Region* <BANGALORE_REGION>
SRST Reference* <BANGALORE_GATEWAY_SRST>
→ Save
Applying Device Pool to Phones
Device → Phone → Phone Configuration page → Device Pool <BANGALORE_DP>
→ Save

IP Phone Boot up Process

1. Obtain power
Sources
 Cisco PoE (10W)
 IEE802.3AF (15.4W)
 Mid-span Power
 Electrical Power
Process
From CDP message switch can be able to analyze how much power needed for an IP Phone
Cisco switch send FLP (Fast Line Pulse 147KHz), phone receive and send back FLP. Then
switch provide required power. 3rd party phone will take full power.
2. Run Boot Strap Loader
IP Phone runs a bootstrap loader and expand the firmware image that is stored inside the
flash memory of phone. This will initiate the software and hardware of IP Phone.

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 CCNA VOICE

3. Obtain VLAN information

Phone send CDP request message to get VLAN (VOICE) information. If voice VLAN information
not configured, there will be a delay in this stage.
To avoid the delay use #switchport voice vlan untagged
4. Obtain IP Address
IP Address can be obtained from DHCP server.
 IP Address
 Subnet Mask
 Gateway IP
 TFTP Server IP
If DHCP server in a remote location #ip-helper address <IP> command is used to change DHCP
broad cast to Unicast.
5. Download Configuration File
Phone contacts TFTP server and request its configuration file. Each phone has customized
configuration created by CUCM and uploaded to when administrator creates and modifies the
phone.
For SCCP Phones : SEP<MAC-ADDRESS>.cnf.xml
For SIP Phones : SIP<MAC-ADDRESS>.cnf.xml
If above file is not present xmldefault.cnf.xml will be downloaded (For auto reg also)

6. Registering with CUCM

Phone register with primary CUCM listed in the configuration file. CUCM server will send
other configurations such as DNs, Softkeys, etc. via SCCP messages in the last stage of
registration process.

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 CCNA VOICE

TCP Connection
TCP Response
SCCP Keep alive (30s)
SCCP Response
PRIMARY SERVER

SECONDARY SERVER

TERTIARY SERVER

CUCM Call Flow

3. CUCM CHECK DIAL PLAN (DN)

6. PROPERTY NEGOTIATION
(IP, Port No:, Codec)

5. PHONE B ATTENDED THE CALL

Phone A: 1001 7. END TO END RTP Phone B: 1002

Key Protocols
 SCCP (Signaling Protocols)
 SIP (Signaling Protocols)
 RTP
Call Flow
1. ‘A’ sends SCCP OFF HOOK signal to CUCM
2. CUCM sends DIAL TONE to ‘A’

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 CCNA VOICE
3. ‘A’ dials DIRECTORY NUMBER (DN) [Digit by Digit (SCCP) or En block (SIP)]
4. CUCM check DIAL PLAN (DN, Manipulation, Call Privilege, Call Coverage)
5. When match found CUCM send SCCP RING signal to ‘B’ and SCCP RING BACK signal to ‘A’
6. ‘B’ attended the call
7. Property negotiation (IP, Port No:, Codec)
8. Establish end to end RTP media streaming + RTCP
Maximum Number of Calls and Busy Trigger
Maximum Number of Calls: = Active calls + outgoing calls
Busy Trigger: = Active Calls + Incoming Calls
Maximum Number of Calls >= Busy Trigger

User Accounts
Why would we create End users?
• Next evolution of phone system (Like Skype)
• Soft Phones requires user logins (Call center setup)
• SIP Phone need user account
• User can manage their own phones (Reducing administrator load - Speed
Dial, Change Language)
• Extension mobility (Like Roaming Profile in Windows)
• Tracking (All international Phone calls report per user)

1. End User
End users are typically people who type User Name & Password in a login screen
(usually web page) to access some features or control.
• Associated with actual person
• For individual interactive login
• Used to assign user features and admin rights
• Create locally & Can be integrated with LDAP (Light Weight Directory
Access Protocol- Microsoft, Netscape, SUN, IPLANET)

2. Application User
Application user is typically applications which send authentication information to
read or write something on the system.
• Associated with an Application
• For non-interactive logins
• Used for application authorization
• Create locally, unable to integrate with LDAP

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 CCNA VOICE

Structure of User Accounts

User accounts in a CUCM server contains 2 sections (General & CCM Data)
General Data
 User ID/ First/ Last/ Middle Name
 Department
 Phone Number/ Mail ID
 Password
CCM Data
 PIN Number
 CUCM Administration Privileges
 Associated IP Phones/ DN
 Feature Access

End User Configuration

1. Manual Local data (No LDAP)
2. BAT
3. LDAP Sync: General info managed from LDAP server, Password managed from CUCM
4. LDAP Authentication: Password managed by LDAP

1. End User Manual Configuration Local Data

Cisco Unified CM Administration → User Management → End User → Add New
User ID : Jaseem
Password : Ciscovoice123
PIN : 123 (Extension Mobility)
Last Name : vp
Associated Devices : <Select Phones> → Save
User Management → User Group → Standard CCM End Users → Add → Find → Add Selected
http://[IP_of_CUCM]/ccmuser
Now user can login to user page and manage (limited access) the associated phones.
User Options → Device →
Line Settings, Speed dial, User Settings, Password, PIN, etc.

Managing User Groups, Roles, Privileges (Sub Administrators)

1. User assigned to Group
2. Group assigned to Roles

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 CCNA VOICE
3. Roles assigned to Privileges
1. Creating Groups
User Management → User Group → Add New
Name : Read_Only_Administrator_Group
Add End User to the Group → Select User → Add Selected → Save
2. Creating Roles
User Management → Roles → Find → Copy/ Add New
Role Information: Application →
Cisco Call Manager Administration
Cisco Call Manager Service Ability, etc. → Next
Name : Read_Only_Administrator_Roles
3. Select Privileges according to our requirement
4. Assign Roles
User Management → User Group → Roles (!) → Assign Role to Group → Select Roles
(Read_Only_Administrator_Role)
Standard CCM Admin Users (To get the Web Page Access)
→ Add Selected → Save

2. End User BAT Configuration

Step 1. Service Activation
Cisco AXL Web Service
Cisco Bulk Provisioning Service
Cisco TAPS Service
Step 2. Create Template
Bulk Administration → User → User Template → Find → Add New
User Template Name* : Sales_Users_Template
Check : Default Password to user ID
User Group : Standard CCM End User
→ Save
Step 3. Create CSV File (Use .xlt file to generate .txt CSV file)
Fill the text file with required info
Step 4. Upload File
Bulk Administration → Upload/ Download → Add New
Browse,
Select The Target* : Users

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 CCNA VOICE
Transaction Type : Insert Users
Step 5. Insert Users
Bulk Administration → Users → Insert Users

LDAP
LDAP allows organizations to create single, centralized directory information
store. LDAP holds informations about User Account, Passwords, Privileges, etc.
Informations in LDAP available to other applications & softwares hence no need to
maintain separate directories. It simplifies user administration because only need
to store user information in a single place.
LDAP Sync: some user data is maintained in LDAP Server & replicated to CUCM.
General data obtained from LDAP will be read only. But some user attributes can be
edited from CUCM. User password is local to the CUCM only.
LDAP Sync: General info managed from LDAP server, Password managed from CUCM
LDAP Authentication: It redirects the password authentication from CUCM to LDAP
System. Centralized password system. It problem in LDAP authentication, when LDAP
server fails nobody can login.
LDAP Mechanism
1. All existing User account in CUCM database are deactivated (not deleted)
2. Accounts whose CUCM User ID exactly match with user ID in LDAP are
reactivated & its attributes updated from LDAP to CUCM
3. LDAP accounts are replicated to CUCM database
4. CUCM deactivated accounts will be deleted after 24 hours

3. Adding End User by LDAP (LDAP Sync)

We are discussing LDAP integration using Windows Active Directory. Windows ADDS
perform full sync of all records every time, leads server busy and affect
performance, hence schedule to off hours. Here Password managed by CUCM
Windows Server 2012 Side Configuration
1. Installing Windows ADDS
i. Server Manager → Add Roles and Features → Server Roles
 Active Directory Domain Services
 Active Directory Lightweight Directory Services
Install → Restart Server
ii. Promote Windows server 2012 to a Domain Controller (Configure ADDS)

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 CCNA VOICE
Action Center flag → Promote this server to a domain controller
Add a new forest → Domain name <smart.com> → Next
Directory Service Restore Mode (DSRM) <********Ciscovoice123>
→ Next → Next → Next → Next → Next → Install
2. ADDS Configurations
Server Manager → Tools → Active Directory Users and Computers
Right click on the domain name <SMART.COM> → New → Organizational Unit
Give a name <CUCM Users> → OK
Expand the OU and create users (Last Name is used for SYNC)

CUCM Side Configuration

Cisco Unified Serviceability → Tools → Service Activation → Cisco DirSync → Save
Cisco Unified CM Administration → System → LDAP → LDAP System
Check Enable Synchronizing from LDAP Server
LDAP Server Type <Microsoft Active Directory>
LDAP Attribute for User ID <sAMAccountName> → Save
System → LDAP → LDAP Directory → Find → Add New
!!!** Existing End Users found in the corporate directory will be deleted**!!
LDAP Directory Information
LDAP Configuration Name* <CUCMLDAP1>
LDAP Manager Distinguished Name* <[email protected]>
LDAP Password* <Ciscovoice123>
Confirm Password* <Ciscovoice123>
LDAP User Search Base* <ou= CUCM Users, dc=smart, dc=com>
LDAP Directory Synchronization Schedule
Perform Sync Just Once <Check>
LDAP Server Information
Host Name or IP Address for Server LDAP Port
142.100.128.0 389
→ Save → Perform Full Sync Now
Check status from User Management → End User → Find

4. CUCM LDAP Authentication

LDAP Authentication: Password managed by LDAP (Password not replicating, entire
Authentication redirected to LDAP server)
Cisco Unified CM Administration → System → LDAP → LDAP Authentication
Check Use LDAP Authentication for End Users
LDAP Manager Distinguished Name <[email protected]>

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 CCNA VOICE
LDAP Password <Ciscovoice123>
Confirm Password <Ciscovoice123>
LDAP User Search Base <ou= CUCM Users, dc=smart, dc=com>
Host Name or IP Address for Server LDAP Port
142.100.128.0 389
→ Save

CUCM User Phone Telephony Features

CUCM supports wide range of features.
Some basic annunciator will can be activated using
Cisco Unified CM Administration → Tools → Service Activation → Cisco IP Voice Media
Streaming App → Save
It is also called Call Coverage Features, that refers several feature and mechanism
that all incoming calls are answered

1. Shared Line
Same Directory Number for many phones, all those phones will ring same time.
Cisco Unified CM Administration → Device → Phone → <Select Phone> → Line [2] - Add
a new DN
→ Directory Number* : 1111 → Save

NB: All the phones will be ringing simultaneously, when connected to one, others
will be in ‘Remote in Use’ mode. When the connected phone goes to ‘Hold’ others can
‘Unhold or Resume’ the call (Less Privacy)

To overcome the issue, enable privacy in service parameter

Cisco Unified CM Administration → System → Service Parameters
Server : <Select Server>
Service : Cisco Call Manager (Active)
Enforce Privacy Setting on Held Calls : True

2. Conference on Shared Line (Barge & Privacy)

If 2 phones have shared line configured & one of the hone is using that line, the
second phone can force a three-way conference by using Barge feature. The
conference is hosted on first phones ‘built in Conference Bridge’. It doesn’t use
CUCM resources for conferencing. Phone itself have some DSPS (Built in Bridge)

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 CCNA VOICE
Create a Phone Button Template with ‘Privacy Button’ and apply to the
shared line phones.

Enable ‘Barge Soft Key’ while ‘Remote in Use’ state, or use ‘Standard Feature Soft
key template’
(Single Phone)
Device → Phone → <Select Phone>
Built In Bridge : On
Privacy : Off

Or (Cluster wide)
Cisco Unified CM Administration → System → Service Parameters
Server : <Select Server>
Service : Cisco Call Manager (Active)

Builtin Bridge Enable : On

Privacy Setting : False

3. Join Conference
This uses CUCM resources to perform conference.
Enable ‘Join’ soft key while connected state
Answer call → put on hold → Answer next call → Put on hold → Join button
3 call on hold press Join Button

4. Call Back
Monitor the availability of another user when he come back.
Add soft key ‘Call Back (CallBack)’ template while ‘on hook & ring out’ state.
Dial a Number + CallBack

5. Call Forward
All or some calls to be forwarded to a destination number by the User or
Administrator either at the phone itself or at CUCM User/Admin pages. During call
forward, the CSS in device & Line are ignored, hence Call Forward CSS must be
configured.
1. Forward All from the Phone
Set Call forward All (CFwdALL) all soft key while ‘On Hook’ state

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 CCNA VOICE

2. Conditional Call Forward

Go to line page → Call Forward and Call Pickup Settings
Forward All
Forward Busy Internal
Forward Busy External
Forward No Answer Internal
Forward No Answer External
Forward No Coverage Internal
Forward No Coverage External
Forward on CTI Failure
Forward Unregistered Internal
Forward Unregistered External
No Answer Ring Duration (seconds) : 10
Create End User & Associate a phone, then he can also configure Forwards.

6. Transfer
Transferring is a kind of forwarding done after speaking between two parties.
1. Consulting Transfer
There will be a consulting between 2 parties before transferring. We have to press
the Transfer Softkey 2 times, one to start consulting and again to transfer call.
Also user can Press Transfer Softkey + DN + Transfer Softkey without consulting.
Enable ‘Transfer (Trnsfer)’ soft key while Connected state
2. Direct Transfer
Call 1 : Hold
Call 2 : Active
Select Held call the Press DirTrfr
DirTrfr : Call 1 & Call 2 get connected
Enable ‘Direct Transfer (DirTrfr)’ soft key while Connected state

7. Call Park
Allows user to temporarily attach a call to a Park Slot number. It is a special
kind of holding.
Call Routing → Call Park →
Call Park Number/Range : 222X

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 CCNA VOICE
Cisco Unified Communications Manager : CUCM-SUB01
Enable ‘Park’ soft key while connected state

System → Service Parameters →

Call Park Display Timer* : 10 (Shows message “Call Park At 222X”)
Call Park Reversion Timer* : 60 (Ring back the initial phone)

8. Direct Call Park

Call Routing → Direct Call Park →
Number : 333X
Reversion Number : 1005
Retrieval Prefix : *
→ Save
Use ‘Transfer (Trnsfer)’ and transfer to 333X (eg: 3330). To retrieve call dial
*3330 from any phone. If not answered 1005 will ring.

9. Call Pickup
Ability to pick someone else ringing phone.
Define a soft key template while ‘on hook’ state consisting following keys
Pick Up (PickUp)
Group Pick Up (GPickUp)
Other Pickup (oPickup)
Create Pickup Groups
Eg. Group 1:
Call Routing → Call Pickup Group
Call Pickup Group Name : SALES_GROUP
Call Pickup Group Number : 1999
Description : SALES_GROUP_PICKUP_NUMBER
→ Save
Eg. Group 2:
Call Routing → Call Pickup Group
Call Pickup Group Name : PURCHASE_GROUP
Call Pickup Group Number : 2999
Description : PURCHASE_GROUP_PICKUP_NUMBER
→ Save
Associate Phones to Pickup Group
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 CCNA VOICE
Device → Phones 1 → Line → Call Forward and Call Pickup Settings →
Call Pickup Group : SALES_GROUP

Device → Phones 2 → Line → Call Forward and Call Pickup Settings →

Call Pickup Group : SALES_GROUP

Device → Phones 3 → Line → Call Forward and Call Pickup Settings →

Call Pickup Group : PURCHASE_GROUP

Device → Phones 4 → Line → Call Forward and Call Pickup Settings →

Call Pickup Group : PURCHASE_GROUP
1. Auto Pickup
Phones in same group can be picked up with a single soft key. When Phone 1 is
ringing, hit ‘PickUp’ soft key and ‘Answer’
To get single button answer
System → Service Parameters →
Auto Call Pickup Enabled* : True
2. Group Pickup
Ringing Phones in a group can be picked up from other group. When Phone 3 or Phone
4 ringing, Phone 1 or 2 can pick up by hitting ‘GPickUp’ + GROUP_NUMBER (eg: 2999)
3. Direct Pickup
‘GPickUp’ soft key followed by Directory number in same group. Or associate all the
groups.
4. Other Pickup
Call Routing → Call Pickup Group → Select a Pickup Group from ‘Available Call
Pickup Groups’ and add to ‘Selected Call Pickup Groups’ → Save
Do the same for other Pickup groups that is associate all Groups
Now just press ‘oPickup’ soft key to pick other group calls

10. Do not disturb

Add soft key ‘Toggle Do Not Disturb (DND)’ while on hook
When we press ‘DND’, under CUCM phone the ‘Do Not Disturb’ filed will be checked.
DND Option : Ringer off
DND Incoming Call Alert : Flash only/ Beep only/ Disable

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 CCNA VOICE

11. Services
Future topic

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 CCNA VOICE

SIP Phone Registration

1. Creating SIP End User
Create an END USER
Name : sipuser
and Add user to Group →
 Standard CCM End Users
 Standard CTI Enabled
→ Add Selected → Save
2. Registration
Device → Phone → Add New →
Phone Type* : Third-party SIP Device (Basic) → Next
MAC Address* : AAAABBBBCCCC <DUMMY_MAC_ADDRESS>
Description : SIP_PHONE1
Device Pool* : Default
Owner User ID* : sipuser
Phone Button Template : Third-party SIP Device (Basic)
Device Security Profile* : Third-party SIP Device Basic - Standard SIP Non-
Secure Profile
SIP Profile* : Standard SIP Profile
Digest User : sipuser <SELECT_END_USER>
→ Save

Line [1] - Add a new DN → Directory Number : 1111 → Save

3. Login to SIP Phone (3CX Soft Phone)
Settings → Accounts → New
Account Name : <OPTIONAL>
Caller ID : <OPTIONAL>
Extension : 1111
ID : sipuser <END_USER_ID>
Password : 123
Specify IP Address of SIP Server
I am in the office- local IP : 142.100.64.11 <IP_OF_CUCM>
Associate end user to sip phone

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 CCNA VOICE

DNA (Dialed Number Analyzer):

Cisco Call Manager Dialed Number Analyzer enables you to enter in calling and
called numbers and receive a report that describes the following:
 How CallManager intends to route the call
 How CallManager changes the calling party number
 How CallManager changes the called party number
Service Activation
Cisco Unified Crevice Ability → Tools → Service Activation (PUBLISHER) → Cisco
Dialed Number Analyzer → Save

Access DNA Page

Cisco Unified Crevice Ability → Tools → Dialed Number Analyzer
or https://<CUCM_IP_ADDRESS>/dna

Perform Analysis
Analysis → Phones → Select Phone → Enter Dialed Digit → Analyze
Analysis → Trunks → Select Trunk → Calling Party Number, Called Party Number →
Analyze

CUCM Reporting
Cisco Unified Reporting tool pulls informations from a range of sources related to
troubleshooting and maintaining & system analysis of CUCM.
Cisco Unified Reporting → Go or
https://<IP_ADDRESS_OF_CUCM>/cucreports

CAR (Call Detailed Record Analysis & Reporting)

 Every call that CUCM processes can be logged. It contain information about
Call, Voice Clarity, etc. These logs are called CDR (Call Detailed Record) &
CMR (Call Management Record).
 CDRs stored in Subscribers & uploaded to CDR/CAR Database of Publisher Server
at regular interval (this interval can be administratively set).
 CDR Database can be used by 3rd party billing application to prepare internal
or external phone billing reports.
To activate CAR,

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 CCNA VOICE
Cisco Unified Crevice Ability → Tools → Service Activation (PUBLISHER) →
Cisco CAR Web Service → Save
Then go to https://<IP_ADDRESS_OF_CUCM>/car

If external billing server is to be used, activate Cisco SOAP CDRonDemand Service

must be enabled. (SOAP – Simple Object Access Protocol)
CDR (Call Detailed Record)
It contain Call Starting time, ending time, Call Duration, etc.
After activating CAR Service, you should configure CDR to load data.
CM Administration Page → System → Service Parameter → Select Server →
CDR Enabled Flag*: True: It determines whether CDRs will be generated. This must be
set on all servers. Default is false
CDR Log Calls with Zero Duration Flag: True: CUCM records failed calls or calls
which duration less than 1 second.
Display FAC in CDR: Controls whether FAC used to make a call will be included in
CDR.
CMR (Call Management Record):
Packet sent, Received, Packet loss, Jitter
CM Administration Page → System → Service Parameter → Select Server →Call
Diagnostics Enabled: Enabled Regardless of CDR Enabled Flag

Exporting CDR & CMR

Generating CDR Reports
The following reports are available
 Bills: Individual, Department
 Top N: By Charge, By duration, By Number of Calls
 Manager Call Usage, Assistant call usage
 IP Phone Services: Number of users subscribed to the services, and
utilization percentage of each

Cisco Unified RTMT (Real Time Monitoring Tool)

RTMT allows Administrator to collect, view, interpret, and monitor various
counters, trace files, and log files generated by CUCM.

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 CCNA VOICE
It is a software installed on Administrators work station. It is available
to download from CUCM plugins menu itself.
Cisco Unified CM Administration → Applications → Plugins → Find → Cisco Unified
Real-Time Monitoring Tool – Windows → Download
Capabilities of RTMT
 Monitor system health
 Generate e-mail alerts for objects that fall below or exceed defined
threshold value
 Collect, view different trace files
 View Syslog messages
 Configure & Monitor Performance counters
 Gateway Activity (H.323, MGCP, FXO/FXS, T1/E1, SIP/IC Trunks) informations

IP Phone Web Access

It gives all device information (MAC, Host Name, Model, DN, Firmware, Network
Configuration & statistics, logs, etc.)
Device Page → Web Access*: Enabled
Click IP Phone IP Address to get web page
User can do this from Security settings of the Phone

Dependency Record
Step 1 Choose System > Enterprise Parameters.
Step 2 Scroll to the CCMAdmin Parameters area of the window.
Step 3 From the Enable Dependency Records drop-down list box, choose False.
A dialog box displays with a message about dependency records. Read the information
carefully before clicking OK.
Step 4 Click OK.
The field displays True.
Step 5 Click Update.

Step 6 Close the browser that you are using; then, reopen the browser. This
makes the parameter take affect for the entire system.

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 CCNA VOICE

CCNA VOICE REAL LAB

DHCP: 142.100.64.0/24
CUCM: TFTP: 142.100.64.11/24
142.100.64.211/24

A: 1001 B: 1002 C: 2001 D: 2003

SIP: 2004
PICKUP_GROUP_AB: 1000 PICKUP_GROUP_CDS: 2000
1. Configure DHCP Server & POOL (142.100.64.X/24)
2. Register Phone A & B Manually and assign numbers given in the scenario
3. Register Phone C & D Automatically and specify the Auto registration range
4. Configure a SIP Phone by adding another end user (sipuser)
5. Configure shared line (5001) in A, B, C and D
6. Enable Barge in B only, assign privacy button also
7. Configure SPEED DIAL BUSY LABP FIELD to 2004 SIP phone by adding Phone button
template.
8. Configure 2 device pool (US, INDIA) and put A, B in US-DP and C, D in INDIA-
DP & set codecs
9. Configure Call Forward No Answer (20 Sec) to SIP Phone 2004
10. Configure a Park Slot 111X and Directed Call Park at 222X with * Prefix
11. Assign A, B in to a PICKUP_GROUP_AB with group number 1000
12. Assign C, D and SIP Phone in to a PICKUP_GROUP_CDS with group number 2000
13. Configure all kind of PICKUP options
14. Add a read only Administrator (readadmin)
15. Enable call back feature in all Phones
16. Configure 2 end user (user1, user2). Associate A, B to User1 & C, D to User2

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