Cisco ccna voice

Legal notice and disclaimer .......................................................................................................................... 5
Introduction ..................................................................................................................................................
6
Definitions ................................................................................................................................................ 6
Well Known Ports ..................................................................................................................................... 6
Miscellaneous commands ........................................................................................................................ 7
POTS Technologies ....................................................................................................................................... 8
Analogue Connections ..............................................................................................................................
8
PSTN Signalling ......................................................................................................................................... 8
E1 / T1 Signalling ...................................................................................................................................... 9
IP Voice Technologies .................................................................................................................................
11
Cisco Voice Infrastructure Model ........................................................................................................... 11
Signalling .................................................................................................................................................
12
IP Transport ............................................................................................................................................ 13
IP Overhead ............................................................................................................................................ 13
Compressed RTP .....................................................................................................................................
14
Problems with Digital Voice ....................................................................................................................
14
Causes of Delay .......................................................................................................................................
14
QoS ......................................................................................................................................................... 14
AutoQoS ..................................................................................................................................................
15
MQC – Modular QoS CLI .........................................................................................................................
15
Analogue to Digital Conversion / Codecs ................................................................................................... 17
Conversion ..............................................................................................................................................
17
Codec Summary ......................................................................................................................................
17
G711 ....................................................................................................................................................... 17
Numbering Plans ........................................................................................................................................ 19
PSTN Numbering Plan .............................................................................................................................
19
Phones ........................................................................................................................................................ 20
Phone Range ...........................................................................................................................................
20
Phone Boot Process ................................................................................................................................
20
Powering .................................................................................................................................................
21
Basic Configuration .....................................................................................................................................
22
Switch configuration ...............................................................................................................................
22
Configuring DHCP ................................................................................................................................... 22
Configuring NTP ......................................................................................................................................
23

CME Communications Manager Express ....................................................................................................
24
Licensing ................................................................................................................................................. 24
CME Files ................................................................................................................................................ 24
Installing ................................................................................................................................................. 24
Basic CME Configuration ........................................................................................................................ 25
Phone Loads / files ................................................................................................................................. 25
Phone configuration files ........................................................................................................................
26
Ephone-dn .............................................................................................................................................. 26
EPhone ....................................................................................................................................................
26
Additional functions ................................................................................................................................... 29
Voice network Directory (Local Directory on phone) ............................................................................. 29
Call forwarding ....................................................................................................................................... 29
Call transfer ............................................................................................................................................ 29
Call Park .................................................................................................................................................. 30
Call Pickup ...............................................................................................................................................
31
Intercom ................................................................................................................................................. 31
Paging ..................................................................................................................................................... 32
After hours call blocking ......................................................................................................................... 32
Music on Hold .........................................................................................................................................
33
CME GUI ..................................................................................................................................................
33
Gateways .................................................................................................................................................... 34
Analogue gateways – Single call per port ...............................................................................................
34
Digital gateways – Multiple calls per port .............................................................................................. 34
Dial Peers ................................................................................................................................................
34
Call Legs .................................................................................................................................................. 35
Digit Manipulation ......................................................................................................................................
37
POTS Auto stripping ................................................................................................................................
37
Example PSTN Failover ........................................................................................................................... 37
Example 0 for operator........................................................................................................................... 37
Configuring Voice Ports .............................................................................................................................. 38
Configuring VWIC T1 & E1 cards .............................................................................................................
38
Configuring FXO/FXS ports ..................................................................................................................... 38
Unity ........................................................................................................................................................... 40
Unity Range ............................................................................................................................................ 40

Unity Express .......................................................................................................................................... 40
CUE Features .......................................................................................................................................... 40
Troubleshooting ..................................................................................................................................... 41
Setup Process ......................................................................................................................................... 41
Initial Engine Setup .................................................................................................................................
41
Controlling / Connecting to the module .................................................................................................
42
Initial Configuration of the Module ........................................................................................................ 42
Upgrading CUE ........................................................................................................................................
42
Configure CME to access CUE .................................................................................................................
43
CUE Web Interface ................................................................................................................................. 44
Initialisation Wizard ................................................................................................................................
45
Smart Business Communication System .................................................................................................... 48
Typical UC520 Models ............................................................................................................................ 48
Typical CE520 Models .............................................................................................................................
48
CCA Communities ................................................................................................................................... 49
Cisco Configuration Assistant Tabs .........................................................................................................
49
Additional Resources ..................................................................................................................................
50

Version 1.0
Copyright © 2010 Michael Morgan.
All rights reserved. Any redistribution or reproduction of part or all of the contents in any form is
prohibited other than printing for personal use. This publication may be used free of charge, selling
without prior written consent prohibited. You may not, except with our express written permission,
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pirated copy and must be destroyed immediately.
StudyShorts guides are intended to provide enough information for last minute exam preparation and
reference, and are not a substitute for other training material. They were prepared to assist my studies
and passing the associated exam and as such may contain errors and some facts may have been
summarised or removed.

Term Definition
FXO Foreign Exchange Office – Connects to a Telco central
office
FXS Foreign Exchange Station – Connects to a local
analogue phone or a fax
CO. Telco Central Office
Key Switch Typically uses analogue PSTN connections, uses shared
lines between phones and limited feature sets. Phones
tend to have line buttons matching the incoming PSTN
lines rather than extension numbers
PBX Private Branch Exchange - Typically uses digital PSTN
trunks, provides unique telephone extensions and have
a large feature set
Local call A call between to local ports
Off net call A call terminated outside of a local port (PSTN)
DNIS Dialed Number Identification Service. A service
provider by the Telco to signal the number dialled by
the calling party (Direct Inward Dial)
ANI Automatic Number Identification. Signals the
telephone number of the calling party (Caller ID)
Integrated Messaging A subscriber can access both an email box and a voice
mail box using a single client
Unified Messaging A subscribers can access both email and voice mail
from a single mail box
VAD Voice Activity Detection. Allows the phone system to
reduce / stop sending packets during silent periods of a
voice call resulting in a bandwidth saving of about 35%
H.450 Avoids hair-pinning forwarded and Transferred calls
TDM Time Division Multiplexing
DS0 A single timeslot / channel. Carries 64kb/s
T1 1.544mbps. 1.536mbps actual data, .008mbps framing.
24 x DS0 channels.
E1 2.048mbps - 32 DS0 channels
CAS Channel Associated Signalling. Signalling is placed in
data carrying DS0 channels. Typically called Robbed Bit
Signalling
CCS Common Channel Signalling. A dedicated DS0 timeslot
is used for signalling. Commonly called Primary Rate
ISDN
ITU-T International Telecommunication Union,
Telecommunication Standardization Sector
IETF Internet Engineering Task Force
RTP Real-time Transport Protocol. Carries the media stream
(even UDP port)
RTCP Real-time Transport Control Protocol. Carries statistic
information (odd UDP port)
ACD Automatic Call Distributution. Usually used in a call
centre environment
CoS Class of Service – Layer 2 process for prioritising traffic
QoS Quality of Service
ToS Type of Service – Layer 3 process for prioritising traffic
TCL Scripting language allows advanced functionality for
Auto attendant etc
T.37 Fax transmission by transporting the image file using
SMTP (store and forward)
T.38 Fax Relay over an IP network
Definitions
Well Known Ports

Protocol Port IP
FTP 20, 21 TCP
SHH 22 TCP

Telnet 23 TCP
SMTP 25 TCP
DNS 53 TCP, UDP
DHCP /
BOOTP
67 UDP
TFTP 69 UDP
NEWS 119 TCP
NTP 123 UDP
SNMP 161, 162 UDP
Mode Description Command
# Show layer 1 & 2 info on all
interfaces
Show interfaces
# As above but on specific interface Show interfaces interface
# Show layer 3 info Show ip interfaces
# As above but on specific interface Show ip interfaces interface
# Show brief interface status Show ip interface brief
# Clear all counters on one or all
interfaces
Clear counters
(config) Turn off domain lookups No ip domain-lookup
Telnet / Session Management
# Show open sessions from this router Show sessions
# Show open sessions to this router Show users
# Kills one of the open sessions from
this router
disconnect
# Kills one of the open sessions to this
router
Clear line <x>
(config-line) Timeout on the particular line
connection
Exec-timeout minutes seconds
Logging & Debugging
# Redirect status messages to the
current session
Terminal monitor
# Turn off all debugging u all / undebug all / no debug all
# Show log buffer memory stats and
messages
Show logging
(config) Allocate buffer memory for log
messages
logging buffered 32000
(config-line) Stop debug messages corrupting
input field
Logging synchronous
CDP
# Show basic info on connected
neighbors
Show cdp neighbors
# Show detailed info on connected
neighbors
Show cdp neighbours detail
# As above but with wildcards Show cdp entry <name wildcard / *>
(config-if) Disable CDP broadcast on an
interface
No cdp enable
(config) Disable CDP entirely No cdp run
Lower Limit (Hz) Upper Limit (Hz)
Human Ear 20 20000
Human Speech 200 9000
Telephone Channel 300 3400
Miscellaneous commands
Frequencies of Audio Signals

Nyquist Theorem – Frequency sample must be twice the maximum frequency to accurately reconstruct
the original wave form.

Analogue Connections
Two connections-
Ground / Tip – 0v
Battery / Ring – -48v
PSTN Signalling
Signalling
 Ground Start – The station/PBX will ground both ring and tip to request a dial tone.
 Loop Start –When a phone is on hook the loop is open, when taken off hook the station will
close the loop to the exchange to request a dial tone. Typically used in home environments as this
is susceptible to glare.
 Glare – If an incoming call happens at the same time as an outgoing line is requested in a PBX
environment, they can become connected causing confusion to the outgoing caller.
Supervisory Signalling
 On-hook – When the phone is on-hook, the connection between the tip and ring wires is broken
and no electrical signal passes between them. Off-hook – When the phone is off-hook, the phone connects the tip and ring wires, completing
the circuit and allowing electrical signal to pass.
 Ringing – To cause an analogue phone to ring, the phone company sends an alternating current
(AC).
Informational Signalling
 Dial tone – Indicates the phone company is ready to receive digits
 Busy – Indicates the remote phone is already in use
 Ringback – Indicates the remote phone is currently ringing
 Congestion – Indicates the long-distance telephone network is not able to complete the call
 Reorder – Indicates the local telephone company is not able to complete the call
 Receiver off-hook – Indicates the local receiver has been off-hook for an extended period of
time
 No such number –Indicates the dialed number is invalid
 Confirmation – Indicates the telephone company is attempting to complete the call

Frame 1 1st DS0 2nd DS0 3rd DS0 ... 24th DS0
... ... ... ... ... ...
Frame 5 1st DS0 2nd DS0 3rd DS0 ... 24th DS0
Frame 6 1st DS0 S 2nd DS0 S 3rd DS0 S ... 24th DS0
S
Frame 6 1st DS0 A 2nd DS0 A 3rd DS0 A
Frame
12
1st DS0 B 2nd DS0 B 3rd DS0 B
Frame
18
1st DS0 C 2nd DS0 C 3rd DS0 C
Frame
24
1st DS0 D 2nd DS0 D 3rd DS0 D
 Dual-tone multi frequency (DTMF) – The buttons on a telephone keypad use a pair of high and
low electrical frequencies (thus “dual-tone”) to generate a signal each time a caller presses a digit.
 Pulse – The rotary-dial wheel of a phone connects and disconnects the local loop circuit as it
rotates to signal specific digits.
E1 / T1 Signalling
T1 CAS – Robbed Bit Signalling
Least significant bit in every 6th frame is signalling. Reduces quality very slightly.
T1 “Giganto” Frame – a set of 24 DS0 (T1). 193 bits at a time, 192 for data and 1 for framing.
T1 Super Frame (SF) – 12 Giganto frames at a time. For each SF there is two signalling bits per channel
(A & B)
T1 Extended Super Frame (ESF) – 24 Giganto frames at a time. For each ESF there are four signalling bits
(A, B, C & D). This is currently used for most if not all T1 providers
E1 CAS Signalling
Dedicated Framing and Signalling channels (DS0). Channel 0 (1st timeslot) is framing and channel 16 (17th
timeslot) is Signalling, channels 1-15 & 17-31 are voice.
Every signalling DS0 is broken down into two nibbles two provide signalling (A, B, C & D) for two DS0
voice channels. The first frame contains signalling for DSO 1 and DS0 31, the next contains signalling for
DS0 2 and DS0 30 etc.
Signalling is compatible with T1 CAS but very rarely used.

Like E1 CAS a dedicated DS0 channel (17th timeslot) is used for Signalling. Uses a signalling protocol
(Typically ISDN Q931, SS7) rather than four bit signalling. CCS leaves 23 channels available for voice on
T1 and 30 channels on E1.

Layer Purpose Examples
1 Endpoints IP Phone, Cell Phone, Video Phone,
IM Client
2 Applications Voice Mail, Conference Call apps,
Call Centre Apps, 911 Series
3 Call Processing Unified Communications Manager,
UCME, UC500
4 Infrastructure ASA Firewall Voice Router/Gateway,
Voice Switch
UC500 CME CCMBE CCM
Max users 48 250 500 30000+
Redundancy
support
no No No Yes
Host Router Router Server Server
Cisco Voice Infrastructure Model
Call Processing Layer
Cisco Unified Communications 500 (UC500) – Appliance providing firewall, NAT, Integrated Voicemail
& Auto Attendant, Built in FX0 & FXS Ports, VPN, Optional Wireless and Music on Hold. This is a part of
the Cisco Smart Business Communications System (SBCS) range.
Cisco Unified Communications Manager Express (CME) – Next step up from the UC500.
Cisco Unified Communications Manager Business Edition (CCMBE) – Provides CCM call processing,
Cisco Unity Connection and Cisco Unified Mobility applications.
Cisco Unified Communications Manager (CCM) – Call processing only. Supports redundancy and
clustering.
Applications Layer
Cisco Unity Express – Voicemail hardware (Network module or AIM) physically installed into a
supporting router. Supports up to 250 users. This unit provides limited IVR capabilities in order to
provide an Automated Attendant system.
Cisco Unity Connection – Cut down Cisco Unity supporting up to 500 users (7500 dedicated server). Also
provides Advanced Call Routing facilities to calls can be routed based on rules, time of day, caller ID etc.
Cisco Unity – Full unified solution integrating with Exchange, Lotus Notes & Novell GroupWise. Up to
7500 users per server. Supports redundancy.
Cisco Unified Contact Centre – Provides ACD functionality to support a call centre environment.
Cisco Unified Meeting Place - Provides a multimedia conference solution that gives you the capability to
conference voice, video, and data into a single conference call. For example, multiple offices could
participate in a conference call using IP phones, live video feeds, and instant messenger clients. The

conference call could include PowerPoint presentations, shared whiteboards, or live demonstrations.
The organization could also choose

to record the conference call for playback at a later time.

Cisco Unified Presence - Provides status and reach ability information for the users of the voice
network. For example, Joe might check the status for Samantha and find that she is available on an
instant messenger client but is currently engaged in a video call.
Cisco Unified Mobility - Allows users to have a single contact phone number that they can link to
multiple devices. For example, Mike could have the phone number +442920 454343 that links to his
desk phone, cell phone, and instant messenger client.
Cisco Emergency Responder - Because VoIP clients have the ability to “roam around” the network using
wireless phones, Soft Phones, or extension mobility functionality, emergency calls (911/999) could pose
a location problem. Cisco Emergency Responder (ER) dynamically updates location information for a
user based on the current position in the network and feeds that information to the emergency service
provider if an emergency call is placed. The Cisco ER product also helps manage emergency calls in a
centralized IP telephony deployment, ensuring that branch office.
Infrastructure Layer
The Infrastructure layer consists of the IP infrastructure to enable a VoIP telephone network (switched,
routers etc). The uptime of a traditional PBS system if 99.999 percent so as a result the main factors in
the IP infrastructure layer is redundancy and QoS to ensure good uptime and good quality speech.
Signalling
SIP - Developed by the IETF. This uses text strings similar to HTML for signalling. SIP itself is only
responsible for setting up and tearing down sessions between endpoints, the actual session is
transferred typically using RTP over UDP. Registrar, Redirect, Location and Proxy servers can be used.
H.323 - Created by the ITU-T to allow simultaneous voice, video and data transmission primarily across
ISDN links. The signalling is derived from Q.931 signalling and as a consequence is very difficult to
interpret. This is a peer to peer protocol so each gateway in the system is fully independent of any other
and needs full configuration for all other gateways. This administrative burden can be reduced by
incorporating a H.323 Gatekeeper, where the gatekeeper would have the full knowledge of the
infrastructure and all Gateways would ask the Gatekeeper how to find other non local extensions. The
Gatekeeper can also perform other tasks such as CAC (Call Admission Control) and bandwidth
management. H.232 is also responsible for the transport of the media stream. This is the only signalling
protocol that supports Fax connected to a Cisco ATA.
MGCP - Developed by Cisco and the IETF is a system which puts voice gateways under control of a
centralised call agent. The gateway is considered a dumb device, every action such as a phone going off
hook or a button pressed is relayed to the MGCP call agent to ask what to do next such as play a dial
tone. This is not supported by CME.

ocol used to control Cisco endpoints (IP Phones, ATA 186 etc). Works in a similar fashion to MGCP, the
end device communicates with CME for every action

H.323 MGCP SIP SCCP
Body ITU IETF IETF
Industry Support Excellent Fair Very Good Proprietary
Used on Gateways Yes Yes Yes Limited
Used on Cisco
phones
No Limited Yes Yes
Architecture Peer to Peer Client / Server Peer to Peer Client / Server
Version Header Length Type of Service Total Length
Identification Flags Fragment Offset
TTL Protocol Header Checksum
Source IP Address
Destination Source Address
Source Port (16bits) Destination Port (16bits)
Length (16bits) Checksum (16bits)
Ver P X
CC M PT
Sequence
Number
IP Transport
RTP - The media stream is carried using RTP on a negotiated UTP port between 16384 and 32767 (Even
numbers).
RTCP – A RTCP session is created at the same time as the RTP session, this is used to relay statistics
between the participating devices (and CME). Typically Packet count, Packet delay, Packet loss and Jitter
statistics is transmitted. Uses odd number UTP ports
IP Overhead
As raw voice data is sent across a network link, layer 2 and layer 3 frame headers are added to the
stream as below.
Layer 2
Ethernet – 18 bytes
Frame Relay – 4 to 6 bytes
Point to Point Protocol (PPP) – 6 bytes
Layer 3
Total of 40 Bytes
IP – 20 bytes
UDP – 8 bytes
Real-time Transport Protocol (RTP) – 12 bytes

Timestamp
SSRC Identifier
CSRC Identifiers
Compressed RTP
Compresses the network and transport layer headers from 40 bytes down to 2 bytes (without checksum)
or 4 bytes (with checksum). This is considered very processor intensive so is only used on low bandwidth
links (T1 or lower)
Problems with Digital Voice
Bandwidth – 21kbps to 320kbps per call depending on codec. QoS can help prioritise voice during
bandwidth use peaks.
Delay – A maximum one-way delay of 150ms, 200ms is considered the ultimate limit.
Jitter – Change of delay between packets, usually caused when there are multiple data paths available
between the endpoints. A maximum one-way jitter delay of 30ms is advisable. A “De-Jitter Buffer” can
be used to reduce the impact of jitter by buffering a small amount of speech in the device before
playing it. Cisco devices implement a variable sized de-jitter buffer to tune to the connection quality. As
a downside it introduces additional delay.
Packet Loss – As packets are lost there will be holes in the speech. Less than 1% is advisable.
Causes of Delay
Transmission delay – The physical time it takes for the packet to travel the wire (Fixed).
Serialization delay – The time it takes to place the bits on the wire (Fixed).
Codec delay – The time the codec takes to convert voice into a PCM stream.
Queuing delay – The time the packet remains in a queue waiting for transmission. QoS can influence
this by putting packets in to a high priority queue.
QoS
Data applications classes
Mission critical – Critical to the running of the business.
Transactional – Applications interact with the users and required rapid response times.

(config) Create a match all class map
(default)
Class-map classname
(config) Create a match any class map Class-map match-any classname
(config) Create a match all class map Class-map match-all classname
(config-cmap) Match on an ACL Match access-group
(config-cmap) Match on an input interface Match input-interface
(config-cmap) Match based on NBAR application
signature
Match protocol protocol
(config) Create a policy map Policy-map type policyname
(config-pmap) Set a class map for this policy Class classname
(config-cmap-c) Set a priority bandwidth of kbps Priority kbps
(config-cmap-c) Set a priority bandwidth of
percentage of interface bandwidth
Priority percent percent
(config-cmap-c) Set bandwidth of kbps Bandwidth kbps
Scavenger – Non productive and no business need. P2P apps etc.
Trust Boundary
All devices are capable if marking packets for priority. Upstream devices can either trust these markings
or generate new marking by inspecting the traffic. The most efficient way is to mark the traffic at the
closest point to the end device, this allows more efficient transport of the packet throughout the
network and avoids the Distribution and especially the Core switches classifying traffic. When
configuring AutoQoS it is possible to control whether the downstream devices marking are to be
trusted.
Queuing
Allows changing the default queuing method on Cisco devices (routers and switches). By default traffic
is sent on a FIFO basis.
Low Latency Queuing (LLQ) is the most popular. A single “priority queue” and many “custom queues”.
AutoQoS
Switch
(config-if) # auto qos voip
(config-if) # auto qos voip cisco-phone
(config-if) # auto qos voip cisco-softphone
(config-if) # auto qos voip trust
The first three options will only enable the trust boundary if a Cisco phone is detected using CDP. The
last command will trust any marking regardless, typically used where non Cisco phones are used.
Router
(config-if) # auto qos voip
(config-if) # auto qos voip trust
Notes-
Ensure serial links have a defined bandwidth using the ‘bandwidth XXX’ command under the interface as
routers cannot automatically detect it.
MQC – Modular QoS CLI
Class map
Used to identify and classify traffic. Matches on-
 ACL
 Input interface
 NBAR (Network based application recognition). This looks at the up layers to find the application
Match-any signifies an OR condition between statements
Match-all signifies an AND condition between statements
Policy-map
Controls what to do with traffic
Example-
(config) # Class-map match-any WEB_TRAFFIC - Class map to match on either HTTP or HTTPS
(config-cmap) # Match protocol http
(config-cmap) # Match protocol https
(config) # Class-map match-all VOIP - Class map to match on RTP traffic
(config-cmap) # Match protocol rtp
(config) # policy-map VOIP - Policy map to give priority bandwidth to VOIP
(config-pmap) # class VOIP
(config-pmap-c) # priority 4000
(config) # interface Ethernet 0 - Set the QoS on an interface
(config-if) # service-policy output VOIP

Analogue to Digital Conversion / Co

Codec Bandwidth MOS Codec
Delay
Complexity 20ms
Sample Size
(bytes)
Notes
iLBC 15.2kbps 4.1
G.711 64kbps 4.1 0.75ms Medium 160
G.729 8kbps 3.92 10ms High 20 Most
Supported
G.723.1 6.3kbps 3.9 30ms High
G.723.2 5.3kbps 3.8
G.726 32kbps 3.85 Medium
G.726 24kbps
G.729a 8kbps 3.7 10ms Medium
G.728 16kbps 3.61 High
Conversion
1. Sample the waveform – Pulse Amplitude Modulation
2. Calculate the number representing each sample (quantisation)
3. Convert to binary – Pulse Code Modulation (G711a etc)
4. Compress if required
Codec Summary
Standard PSTN is considered to have a MOS of 4
Comfort Noise - Digital based telephony in some cases introduces a small amount of noise on the call.
This avoids the scenario where the listener may believe that the transmission has been lost, and
therefore hangs up prematurely. Additionally reduces the effects of VAD introducing sudden change in
sound level
iLBC – Internet Low Bit rate Codec
MOS – Mean Opinion Score. Human based rating which scores the quality of speech between 1 (poor)
to 5 (excellent). http://en.wikipedia.org/wiki/Mean_opinion_score
PQSM – Perceptual Speech Quality Measurement. Machine based scoring from 6.5 (poor) to 0
(excellent)
G711
Two types-
 µ-law (North America & Japan)
 A-law (Europe and reset of World)

Similarities Between A-law and u-law

 Both are linear approximations of logarithmic input/output relationship.
 Both are implemented using eight-bit code words (256 levels, one for each quantization
interval). Eight-bit code words allow for a bit rate of 64 kilobits per second (kbps). This is
calculated by multiplying the sampling rate (twice the input frequency) by the size of the code
word (2 x 4 kHz x 8 bits = 64 kbps).
 Both break a dynamic range into a total of 16 segments:
o Eight positive and eight negative segments.
o Each segment is twice the length of the preceding one.
o Uniform quantization is used within each segment.
 Both use a similar approach to coding the eight-bit word:
o First (MSB) identifies polarity.
o Bits two, three, and four identify segment.
o Final four bits quantize the segment are the lower signal levels than A-law.
Differences Between A-law and u-law
 Different linear approximations lead to different lengths and slopes.
 The numerical assignment of the bit positions in the eight-bit code word to segments and the
quantization levels within segments are different.
 A-law provides a greater dynamic range than u-law.
 u-law provides better signal/distortion performance for low level signals than A-law.
 A-law requires 13-bits for a uniform PCM equivalent. u-law requires 14-bits for a uniform PCM
equivalent.
 An international connection needs to use A-law, u to A conversion is the responsibility of the u-
law country.

PSTN Numbering Plan
ITU-T E.164
 Country Code
 National Destination Code
 Subscriber Number
Example : North American Numbering Plan (NANP)
Country Code
Area Code
Central Office Code
Station Code
Example - 1 480 555 1212

Lines Switch XML Apps PoE Notes
Text Graphics Pre 802.3af
7906G 1 No Yes No Yes Yes
7911G 1 Yes Yes No Yes Yes
7914/791
5/7916
14 No No No No No Expansion
Module
7920 1 No Yes No No No 802.11b
Wifi Phone
7921 1 No Yes Yes Yes Yes A,B & G
Wifi, PTT
7931 24 Yes Yes No No Yes
7936 1 No No No No No Conference
Station
7937 1 No No No No Yes Conference
Station
7940G 2 Yes Yes Yes Yes No
7941G 2 Yes Yes Yes Yes Yes High res
screen
7941G-GE 2 Yes Yes Yes No Yes Gig
Ethernet
7942G 2 Yes Yes Yes Yes Yes High
Quality
Audio
7945G 2 Yes Yes Yes No Yes High res
screen
7960G 6 Yes Yes Yes Yes No
7961G 6 Yes Yes Yes Yes Yes High res
screen
7961G-GE 6 Yes Yes Yes No Yes Gig
Ethernet
7962G 6 Yes Yes Yes Yes Yes High
Quality
Audio
7965G 6 Yes Yes Yes No Yes High res
screen
7970G 8 Yes Yes Yes Yes Yes Colour
Touch
screen
7971G-GE 8 Yes Yes Yes No Yes Colour
Touch
screen
7975G 8 Yes Yes Yes No Yes Colour
Touch
screen
7985 1 Yes Yes Yes No Yes Video
Phone
ATA 186 2 No No No No No Dual FXS
ATA 188 2 Yes No No No No Dual FXS
VG224 24 No No No No No Analogue
Gateway
:FXS
VG248 48 No No No No No Analogue
Gateway
:FXS
IP
Communi
cator
8 - - - - - Soft Phone
Phone Range

Unified
Personal
Communi
cator
Expansion Module adds an additional 14 lines to a 796x and 797x phones. Up to two units can be added.
Phone Boot Process
1. Switch detects PoE capabilities and sends power if required.
2. Phone boots software image.
3. Switch sends the Voice VLAN info to the phone using CDP.
4. IP Phone uses DHCP to get its IP address including ‘option 150’ (TFTP IP Address).
5. Phone contacts TFTP server and gets configuration file.
6. Phone registers with the CME Server listed in the config file.

Class Allocated
Power
Actual Power
Used
0 15.4W 0.44 to 12.95
1 4.0W 0.44 – 3.84W
2 7.0W 3.84 – 6.49W
3 15.4W 6.49 – 12.95W
Inline Power
Cisco Pre-Standard PoE – A switch will send a tone (Fast Link Pulse – FLP) down the network cable, an
unpowered Cisco phone will loop the tone back to the switch. The switch then sends a maximum of 6.3
watts to the phone for it to begin powering up. The phone then sends it actual power requirements to
the switch using CDP. For non Cisco phones the switch will send the full 15.4 watts.
IEEE 802.3AF – The switch sends a constant DC current to the device (does not harm the device because
of DC filtering), a 802.3AF device has a specific value resistor allowing the switch to detect the power
requirements of the device. This standard is able to send power over Gigabit Ethernet.
Midspan Power
Power Patch Panel – Sits between the switch and patch panel to inject power. Avoids cost of replacing
switches for PoE switch.
Power Injector – Simple power injector, no intelligence.
Wall Power
CP-PWR-CUBE-3

# Show all defined vlans and assigned
ports
Show vlan
# Show total power available / used
and port power usage
Show power inline
# Show directly connected Cisco
Device information
Show cdp neighors
# Show VTP mode and status Show vtp status
Set Switch Port Trunking Mode
(config-if) Set the trunk encapsulation (ISL no
used much now)
Switchport trunk encapsulation
dot1q
(config-if) Enable the trunk mode Switchport mode trunk
(config-if) Auto mode. Will aggressively try to
raise a trunk. Default
Switchport mode dynamic desirable
(config-if) Auto. Will not raise trunk but will if
the other end does.
Switchport mode dynamic auto
(config-if) Set native (untagged) Vlan Switchport trunk native vlan vlan
Set Switch Port Access Mode
(config-if) Set access port Switchport mode access
(config-if) Set the data vlan Switchport access vlan vlan
(config-if) Set the voice/auxiliary vlan Switchport voice vlan vlan
(config-if) Set STP portfast Spanning-tree porftfast
Configure VLAN
(config) Create a vlan Vlan vlannumber
(config-vlan) Assign a name to the vlan Name name
Misc
(config-if) Set automatic power mode Power inline auto
(config-if) Turn off PoE Power inline never
(config-if) Leave power on for second after link
goes down
Power inline delay shutdown
seconds
# Display DHCP leases Show ip dhcp binding
(config) Create a DHCP pool Ip dhcp pool pool
(dhcp-config) Define network to enable & issues
addressed
Network x.x.x.x /24
(dhcp-config) Set default router Default-router x.x.x.x
(dhcp-config) Set DNS server Dns-server x.x.x.x
(dhcp-config) Set TFTP server address Option 150 ip x.x.x.x
(dhcp-config) Set TFTP server name (not
recommended)
Option 66 ascii tftpservername
(config) Set dhcp excluded addresses Ip dhcp excluded-address x.x.x.x
y.y.y.y
Switch configuration
Notes-
As a guideline make the voice VALNs lower in number than data. This allows spanning tree to get the
Voice vlan up quicker in the event of a network topology failure.
Typically a router will have an access list to stop data and voice traffic crossing the Vlans.
Configuring DHCP

(config-if) Set helper address for a DHCP
server on an interface
Ip helper-address x.x.x.x
# Show NTP sources and status Show ntp associations
(config) Set a time server Ntp server domainname
(config) Set a hour zone and hour difference
for the time
Clock timezone name x
Configure a router as an NTP Master
(config) Allow other devices to get the time
from device
Ntp master
(config) Assign an access list to restrict
access
Ntp access-group list
Notes-
 The ‘Network’ command allows the addition of a mask bit length or network mask. Otherwise is
will issue the default class full subnet mask. Common practice is to include the option 150 in data VLANs as well so phones will work if
plugged into the data VLAN.
 ‘Ip helper address’ is used to create a proxy to send a broadcast received on an interface to a
unicast address. When the unicast is sent it is sent to the address specified but with a source
address of the interface the broadcast was received from. This allows a DHCP server to identify
with DHCP pool to assign addresses accordingly. For this to work the DHCP server must have a
route to the network requiring DHCP services.
Configuring NTP
Stratum 0 – Atomic clock. Stratum 1 – NTP Server directly connection to a radio or atomic clock. Stratum 2 NTP
Server gets its time from a stratum 1 server......

CME Communications Manager Express

# Show all flash files and free space Show flash
# Think DOS... Dir flash:
# Install CME from TFTP Archive tar /xtract
tftp://x.x.x.x/cme..tar flash:
Licensing
IOS – License to run the required IOS (Voice / AdvancedEnterprise etc). Think Windows Server Licence.
Feature License – License CME for a specific number of users. Think Windows CAL.
Phone User License – License the IP phone to interact with CME / CCM. Think Windows XP License.
CME Files
While all the functionality for running voice is built into the routers IOS, Cisco provide TAR files to
provide additional resources for the phone system-
Basic Files – Phone loads / firmware.
GUI Files – HTML web front end.
XML Template Files – Allows the user to edit the GUI such as only allows certain user to perform certain
actions.
MoH Files – Music on hold.
Script Files - TCL scripts for advanced functions (auto attendant, ACD etc).
Miscellaneous Files – Other files such as Custom ring tones.
Installing
1. Get the files.
2. Place the files on a TFTP server
3. Copy the files to the routers flash memory, either-
1. Use the copy command for each file. Takes a long time.
or
2. Use the Archive command to unpack the archive on the router, quick.

# Show telephony-service
Basic Configuration
(config) Go to telephone service
configuration
Telephony-service
(config-telephony) Maximum directory numbers Max-dn x
(config-telephony) Maximum phones on the system
(up licenses purchased)
Max-ephones x
(config-telephony) Defines IP address the phones will
attempt to register
Ip source-address x.x.x.x
Auto Registration and DN assignment
(config-telephony) Disable automatically registering
ephones
No auto-reg-ephone
(config-telephony) Configure ephone-dn to ephone
auto assignment
Auto assign x to y
(config-telephony) Allow time admin from the GUI Time-webedit
(config-telephony) Allow DN admin from GUI. Required
for CUE
Dn-webedit
# Show the internal telephony service
tftp files
show telephony-service tftp-
bindings
# Display the contents of a text file More filename
(config) Define a file for the TFTP server Tftp-server filename alias name
(config-telephony) Define what firmware to load to a
phone
Load phonemodel filename
(config-telephony) Create the configuration files Create cnf-files
‘Ip source-address’ can be set to a loopback interface if supporting phones on more than one interface.
The network and phones must have routes to this address.
Phone Loads / files
As the phone only asks for the filename, not the full path the alias element of the ‘tftp-server’ command
provides the file alias.
Examples-
Tftp-server flash:/phone/7940-7960/P00308000500.bin alias P00308000500.bin
Tftp-server flash:/phone/7940-7960/P00308000500.loads alias P00308000500.loads
Tftp-server flash:/phone/7940-7960/P00308000500.sb2 alias P00308000500.sb2
Tftp-server flash:/phone/7940-7960/P00308000500.sbn alias P00308000500.sbn
To find the filename for the ‘Load’ command reference-
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/requirements/guide/cme43spc.htm

(config) Create a single line dn Ephone-dn tag
(config) Create a dual line dn Ephone-dn tag dual-line
(config-ephone-dn) Assign a number Number number
(config-ephone-dn) Assign a secondary/did number Number number secondary number
(config-ephone-dn) Assign a name for the telephone
directory
Name name
(config-ephone-dn) Preference to use when same
number assigned to many dn’s.
Default is 0.
Preference x
(config-ephone-dn) Consider this the last dn in the hunt
group. Don’t try to find another dn.
Huntstop
(config-ephone-dn) If any /line channel on a dual line dn
is used, don’t place a second call on
the same dn.
Huntstop channel
# Show ephone
# Show ephones trying to register.
Useful to find phone MAC when
setting up phones
Show ephone attempted-
registrations
(config) Create an ephone Ephone no
(config-ephone) Assign a MAC Mac-address xxxx.xxxx.xxxx
(config-ephone) Set phone type. Not required as
CME will find this out
Type phonemodel
(config-ephone) Assign a phone line with a dn Button x:y
(config-ephone) Cold reset phone Reset
(config-ephone) Warm reset phone Restart
XMLDefault.cnf.xml – Basic phone configuration file, contains what IP address is hosting CME and
firmware names to download. This can be viewed using the command ‘more
system:/its/vrf1/XMLDefault.cnf.xml’
Ephone-dn
Represents the phone numbers.
Single Line - Only able to handle on call
Dual Line - Handles two simultaneous calls – allows call waiting, conferencing, consultative transfers
EPhone
Represents the physical phone.

Auto Registration & Assignment

Mode From IOS Help Description
: Normal ring
S Silent ring
B Call waiting beep, no ring Silent ring but beep on call waiting
F Feature Mode Alternate ring tone for a incoming
call
M Monitor Mode Creates a button which shows the
status of the ephone-dn. Also acts
as a speed dial button. Ideal for
receptionist
W Watch Mode As monitor button but watches the
whole phone assigned to the dn
O Overlay Line (no call waiting) Allows multiple phones at the same
time to ring on incoming call
C Overlay Line (with call waiting) Allows multiple phones at the same
time to ring on incoming call
X Overlay Expansion / Overlay
Auto Registration - By default ephones will automatically register with CME, they won’t automatically
be created in the running config. Disabled with the ‘No auto-reg-ephone’ telephony service command.
Auto assignment – CME will automatically assign ephone-dn’s to ephones. Configured with ’Auto assign
x to y’ where x is the start dn and y is the end dn.
Button command options
Button assignments link a DN to a physical button on a telephone. A number of methods can be used on
the assignments-
Single telephone number multiple ephones
Some scenarios require a single extension number to be assigned to more than one telephone, such as
in a call centre environment, a number of approaches are available-
Single dn assigned multiple ephones
Using ‘button x:y’ – All ephones share the one DN/line. Not good for call centre type applications, if one
person receives a call all the ephones will be unable to use that DN / number for both incoming and
outgoing calls.
Multiple ‘ephone-dn’ using same incoming number
Multiple DNs are created with the same extension number, with each DN assigned to a single ephone.
As each phone has a unique DN, multiple phones can both receive and make calls using the number.
Incoming calls are randomly distributed among ephones (only a single phone will ring). If required the
‘Preference’ command allows control of the phone ring sequence, e.g. to always make the phone
assigned to DN 10 ring first followed by the phone assigned to DN 11 if the first phone is busy-
(config) # Ephone-dn 10 dual-line
(config-ephone-dn) # Number 1010
(config-ephone-dn) # Preference 0

There is a problem using this approach when using a dual line ephone-d

n, as each DN can handle two calls, a second call to shared number could go to the second line of the
DN resulting in a call waiting scenario.

The ‘Huntstop’ command stops a second call hitting a dn currently in use (huntstop channel) and places
it on the next dn (no huntstop) Note the last dn has doesn’t have a ‘no huntstop’ command .
(config-ephone-dn) # Huntstop channel
(config-ephone-dn) # No huntstop
(config-ephone-dn) # Huntstop channel
See the Cisco Website for more information.
Overlay Line buttons
Allows an incoming call to ring multiple phones simultaneously i.e. the incoming call will be overlayed to
multiple ephones.
(config) # Ephone-dn 10
(config-ephone-dn) # No huntstop
(config-ephone-dn) # Exit
(config) #Ephone-dn 11
(config-ephone-dn) # Exit
(config) #Ephone 8
(config-ephone) # Button 1o10,11
(config-ephone) # Exit
(config) #Ephone 8
(config-ephone) # Button 1o10,11
(config-ephone) # Exit
In this example multiple DNs are created allowing the shared number 1010 to be used multiple times for
incoming and outgoing calls. The DNs are then overlayed to the telephone buttons, in effect a phone
button will have multiple assigned DNs.
‘C’ Overlay Line (with call waiting). If the buttons are configured with ‘C’ instead of ‘O’, the first call will ring
ephone 8 & 9. A second call will ring the inactive phone but the active user will receive a call waiting beep.
Although the ephone-dn’s are single line and don’t support call waiting, the second call will come in on the
inactive dn, dn 11 which will generate the call waiting beep..
Recommendation is to not use dual lines with ‘O’ and ‘C’.

(config) Select Cisco ephone dn ephone-dn dn
(config-ephone-dn) Assign a name for the telephone
directory
Name name
(config) Select SIP register dn (for attached
sip phones)
voice register dn dn
(config-register-dn) Assign a name for the telephone
directory
Name name
(config) Select telephony service config
mode
Telephony-service
(config-telephony) Set directory sort order (default) Directory first-name-first
(config-telephony) Set directory sort order Directory last-name-first
(config-telephony) Create an entry (for non dn entries
– up to 100)
Directory entry id number name
name
(config-ephone-dn) Forward all calls Call-forward all number
(config-ephone-dn) Set forward when phone busy Call-forward busy number
(config-ephone-dn) Set forward when phone not
answered
Call-forward noan number timeout
seconds
(config-ephone-dn) Forward call on activated night
service
Call-forward night-service number
(config-ephone-dn) Restrict length of a forward number Call-forward max-length length
(config-register-dn) Set forward when phone busy call-forward b2bua busy number
(config-register-dn) Set forward when phone not
answered
call-forward b2bua noan number
timeout seconds
(config-telephony) Set valid call forward destinations Call-forward pattern pattern
Voice network Directory (Local Directory on phone)
Call forwarding
User call forward
‘CFwdAll’ phone soft key allows a user to enter an extension to forward all calls to.
System call forward
A DN can be configured with the command ‘Call-forward all XXX’ & ‘Call-forward busy XXX’ to define where to
forward calls.
Configuring-
‘Call-forward pattern pattern’ and ‘Call-forward max-length length’ are used to control what number calls can be
forwarded to, this helps avoid call toll fraud.
H.450.3 - Allows the original caller and the recipient of the forward to handle the transferred call directly
rather than via the intermediate party handling the media stream (call hair-pinning). This is enabled
when a ‘call-forward pattern pattern’ is specified.
Call transfer
Consulted transfer – User presses the ‘Transfer’ soft key and dials the number to be transferred to. The
user then consults the transfer recipient informing them of the call. The ‘Transfer’ soft key is then
pressed to connect the two parties. This is the default.

d as soon as the transfer number is entered.

(config-telephony) Sets blind transfer system using
H.450.2
Transfer-system full-blind
(config-telephony) Sets consult transfer system using
H.450.2
Transfer-system full-consult
(config-telephony) Sets consult transfer system using
proprietary method
Transfer-system local-consult
(config-telephony) Sets the pattern for valid transfers Transfer-pattern pattern
H.450.2 – Allows the original caller and the recipient of the transfer to handle the transferred call
directly rather than via the intermediate party handling the media stream (call hair-pinning).
By default call transfers can only take place between phones in the system. Setting a transfer pattern
allows calls to be transferred to external numbers. This is means to reduce the possibility of toll fraud.
Call Park
Example config to create a park slot-
(config) # ephone-dn 20
(config-ephone-dn) # number 399
(config-ephone-dn) # park-slot
park-slot timeout command
Basic form-
(config-ephone-dn) # park-slot timeout x limit y
The person who sent the call to the park slot is notified every x seconds for a maximum of y times
before taking action.
Notify a second extension of the parked call-
(config-ephone-dn) # park-slot timeout x limit y notify number
Recall the parked call back to the originator-
(config-ephone-dn) # park-slot timeout x limit y recall
Transfer the timed out parked call to an extension. If that extension is busy transfer to the alternate
number-
(config-ephone-dn) # park-slot timeout x limit y transfer number alternate number
Park Slot reservation
It is possible to assign a reservation group to a park slot. Only ephones configured with the same
reservation group can pick up the parked call.
(config-ephone-dn) # park-slot reservation-group 1 timeout 10 limit 3 transfer 700

(config-ephone-dn) # park reservation-group 1
Notes-
Once a park slot has been created the ‘Park’ button becomes available on the phones.
To pick the call up simply call the parked call number or press this ‘PickUp’ softkey then dial the call park
no. Additionally the person who parked the call can pick up the call by pressing ‘PickUp’ soft key then
press the * key.
Call Pickup
Directed Pickup – Pressing the Pickup button results in the phone sounding a dial tone waiting for the
user to enter the extension number of a ringing phone to pickup.
Local Group Pickup – Pressing the GPickup button picks up a ringing phone in the same pickup group.
Other Group Pickup - Pressing the GPickup button results in the phone sounding a dial tone waiting for
the user to enter the group number a ringing phone to pickup.
To assign a dn to a group use the command-
(config-ephone-dn) # pickup-group xxxx
Notes-
The GPickUp softkey functions differently depending on the call pickup configuration in CME. If there is
only one group configured in CME, pressing the GPickUp button automatically answers the call from
your own group number. You will not hear a second dial tone and you do not need to dial an asterisk to
signify your own group, because only one group is defined. Once you have configured multiple groups in
CME, you will hear a second dial tone after pressing the GPickUp softkey, at which point you can dial
either an asterisk for the local group or another group number.
Directed Pickup can be disabled by entering ‘no service directed-pickup’ from telephony service
configuration mode.
Intercom
A two way communication channel using speaker phone. When a user presses the button assigned to
the intercom the other phone will automatically answer using speaker phone but with the microphone
muted in case the other person is saying something secretive.
(config-ephone-dn) # number A100
(config-ephone-dn) # intercom A101 label “Manager”

(config-telephony) Define outside of hours on
particular day
After-hours date month dayno
(config-telephony) O of H on day between start &
endtime
After-hours day day starttime
endtime
(config-telephony) Define blocked number pattern (up
to 100)
After-hours block pattern no
pattern
(config-telephony) Permanent block (24-7) - no
exceptions
After-hours block pattern no
pattern 7-24
(config-ephone-dn) # number A101
(config-ephone-dn) # intercom A100 label “Assistant”
(config) # ephone 3
(config-ephone) # button 2:20
(config) # ephone 4
(config-ephone) # button 2:21
Further options for the Intercom command-
Barge-in – the intercom will force all other calls into the HOLD state and connect tyhe intercom call
No-auto-answer – Disable the intercom auto answer
No-mute – Disable the auto mute.
Paging
A one way speakerphone based announcement. There are two methods, unicast or multicast. As unicast
requires a single stream per page group member the group is limited to a maximum of 10 members. If
using multicast the network must be capable/configurable of supporting multicast streams. A phone can
only be a member of one paging group but a paging group can be a member of another parent paging
group.
Create a paging group-
(config-ephone-dn) # number 3000
(config-ephone-dn) # paging - Unicast paging or
(config-ephone-dn) # paging ip 239.4.3.4 port 200 - Multicast paging (cannot use 224.)
(config-ephone-dn) # paging group dnlist - Associate a child paging group
Assign a phone to the paging group-
(config) # ephone 3
(config-ephone) # paging-dn 25
After hours call blocking
Ability to block specified number outside of hours.

(config-ephone) Exempt phone from out of hours
block
After-hours exempt
(config-ephone) Set a pin for temporarily removing
blocks
Pin xxxx
(config-telephony) Enable login for pins. Parameters
not required
Login timeout mins clear time
Example-
After-hours day mon 17:00 8:00
After-hours day tue 17:00 8:00
After-hours day wed 17:00 8:00
After-hours day thu 17:00 8:00
After-hours day fri 17:00 8:00
After-hours date dec 25 00:00 00:00
After-hours block pattern 1 90.......... - Block all non local calls
Music on Hold
Stream a wav or au files in the routers flash memory using unicast (up-to 10 like paging) or multicast.
Example-
(config-telephony) # moh music.wav
(config-telephony) # multicast moh 239.4.3.2 port 2100 - Multicast if required
CME GUI
Provided the GIU Files have been installed on the router, the HTML front end can be enabled using the following
commands-
(config) # ip http server - Enable http server
(config) # ip http secure-server - Enable https server
(config) # ip http path flash:/gui - Set the location of the gui files
(config) # ip http authentication local - Set local authentication database
Additional commands to control the front end-
(config-telephony) # web admin system name mike secret password
(config-telephony) # dn-webedit - Enable changing dn through the gui
(config-telephony) # time-webedit - Enable changing time through the gui
To use simply browse to ‘http://x.x.x.x/ccme.html’

# Show the summary and status of all
voice ports
Show voice port summary
# Show the summary and status of all
dial peers
Show dial-peer voice summary
# Debug the dial peer process Debug voip dialpeer
# Show all voice calls Show voice call summary
# show call active voice
Create POTS FXS Dial Peer extension
(config) Create a dial peer Dial-peer voice tag pots
(config-dial-peer) Define the numbers to assign to the
port
Destination-pattern number
(config-dial-peer) Assign a physical port to the dial
peer
Port port
A Gateway is a link from the VoIP telephone system (CME) to a traditional PBS / PSTN or another VoIP
system. A number of gateway types can be employed-
Analogue gateways – Single call per port
FXO (Foreign Exchange Office) Acts as an analogue telephone socket, connecting to the PSTN
exchange / Telco central office. These facilitate Analogue trunks to the telco.
FXS (Foreign Exchange Station) Acts as an analogue PSTN exchange allowing analogue stations / devices
(phones, faxes etc) to be connected to the CME infrastructure. Typical devices for FXS ports - VIC2-2FXS /
ATA186 / ATA188 / VG224 / VG248
E&M (Ear & Mouth / Earth & Magneto) Specific analogue module purely for trunking. Typically used to
connect two PBX systems together
Digital gateways – Multiple calls per port
T1 & E1 CAS Example cards are VWIC-MFT-E1 / VWIC-1MFT-T1, typically used to connect to Telcos.
T1 & E1 CCS (Primary Rate Interface PRI) Example cards are VWIC-MFT-E1 / VWIC-1MFT-T1
Basic Rate Interface (BRI)
Dial Peers
A Dial peer defines how a call enters / leaves CME, there are two types–
POTS Dial Peer connects to a traditional voice system, the call is sent out a voice port where the voice
port is an FXO, PRI etc.
VoIP Dial Peer IP Based, calls are sent to an IP address, another CME system or SIP server can be used.

(config-dial-peer) Description description
Create VoIP Outbound Dial Peer
(config) Create a dial peer Dial-peer voice tag voip
(config-dial-peer) Set the destination pattern Destination-pattern pattern
(config-dial-peer) Send matching calls to the remove
voip server
Session target ipv4:x.x.x.x
Create a T1/E1 Outbound Dial Peer
(config) Create a dial peer Dial-peer voice tab pots
(config-dial-peer) Set the destination pattern Destination-pattern pattern
(config-dial-peer) Set the destination port Port x/x:z
Wildcard Meaning Example Matches
. A single digit 50. 500, 501 ... 509
+ One or more instances of 1+ 11, 111, 11111111
[] Range of digits [1-3]111 1111, 2111, 3111
[14-6]11 111, 411, 511, 611
[6789].. 6xx, 7xx, 8xx, 9xx
T Anything 9T Anything starting with a 9.
Wait for inter-digit time
out to dial
Destination-patterns
When sending a call out through a dial peer a destination pattern must be created to define what calls
should be sent through the dial peer. Various options are available to define the pattern as below-
Call Legs
When a call enters or leaves CME, a call leg is required, so for example if a call comes in on an FXO port
a call leg will be created for that call.
An extreme example could be where a call comes in to CME via an FXO port, CME then sends the call
out to another CME system via an IP trunk then finally the call is sent out an FXS port. The legs in this
example would be-
Leg 1 – Telco exchange to FXO port on voice switch (In to CME ‘A’)
Leg 2 – Voice switch to IP trunk over a Wan (Out of CME ‘A’)
Leg 3 – IP Wan trunk to voice switch (In to CME ‘B’)
Leg 4 – Voice switch FXS to analogue station (Out of CME ‘B’)
A call leg is basically a matching dial peer, as seen above to make an outbound call from CME a dial peer
is required to define the target/port and the destination pattern. Inbound calls ideally require a
matching dial peer as well, dial peers will be matched using the following criteria and order-
1. Matched the dialled 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.
Dial Peer 0
An implicit dial peer for all inbound calls with no matching dial peer. While this functions fine there are
benefits to have an explicitly defined matching dial peer for incoming calls as additional options can be
defined such as valid codecs, disabling vad etc.

POTS Auto stripping
Pots dial peers automatically strip any explicitly defined number from the destination pattern before
sending the call.
Examples
Destination-pattern 9[2-9]....... The 9 will be stripped
Destination-pattern 9[469]11 The 9 & 11 will be stripped
Destination-pattern 91[2-9]....... The 9 & 1 will be stripped
Destination-pattern 9011T The 9011 will be stripped
Prefix <digits> Add the prefix to the remaining dialled digits.
Forward-digits <number> forward number of right most digits, including any digits automatically
stripped.
Digit-strip Default action. Turn off auto stripping using the command no digit-strip.
Num-exp <match> <set> Effectively search and replace. Global config command.
Example PSTN Failover
Example - sending calls for 6... to a remote phone system using an IP trunk. If the trunk fails the calls will
be sent out a POTS voice port to numbers relating to the DID numbers of the extensions, eg 6001 will
get sent to the PSTN number 02920116001 which the receiving phone system will forward to the
extension 6001.
(config) # Dial-peer voice 6000 voip
(config-dial-peer) # Destination-pattern 6...
(config-dial-peer) # Session-target ipv4:10.1.1.2
(config-dial-peer) # Preference 0
(config) # Dial-peer voice 6001 pots
(config-dial-peer) # No digit-strip
(config-dial-peer) # Prefix 0292011
(config-dial-peer) # Port 1/0:15
Example 0 for operator
(config) # Num-exp 0 5000

Mode Description Commands
(config) Select interface Controller interface
CAS
(config-controller) Set framing (esf most common) Framing <sf / esf>
(config-controller) Set coding (b8zs used with esf) Linecoding <ami / b8zs>
(config-controller) Configure CAS Ds0-group groupnumber timeslots
x-y type signalling
CCS
(config) Set the ISDN switch type Isdn switch-type .....
(config-controller) Configure CCS Pri-group timeslots x-y
(config-voiceport) Set start method. Loopstart is
default. Used when trunking to a
pbx
Signal <groundstart / loopstart>
(config-voiceport) Set the dial tone. Also changes the
ring cadence accordingly
Cptone <countrycode>
(config-voiceport) Change the ringing AC frequency Ring frequency <25 / 50>
(config-voiceport) Set the ring pattern Ring cadence patternxx
(config-voiceport) Set custom ring cadence Ring cadence x y z . . . . .
Busyout
(config-voiceport) Set the caller ID Name Station-id name
(config-voiceport) Timeouts .....
Configuring VWIC T1 & E1 cards
Examples
Configure all 24 channels of a T1 line using loop start
(config) # controller t1 1/0
(config-controller) # Ds0-group 5 timeslots 1-24 type fxo-loop-start
(config) # Dial-peer voice 6001 pots
(config-dial-peer) # No digit-strip
(config-dial-peer) # Prefix 0292011
(config-dial-peer) # Port 1/0:5 - Same as tag number
Configure PRI CCS on an E1 line
(config) # controller E1 0/1/0
(config-controller) # pri-group timeslots 1-6
All calls are directed through 1/0:15 for E1 and 23 for T1
Configuring FXO/FXS ports
FXS

(config-voiceport) Set start method. Loopstart is
default. Used when trunking to a
pbx
Signal <groundstart / loopstart>
(config-voiceport) Set the dialling signalling method Dial-type <dtmf / pulse>
(config-voiceport) Length of time before the router
answers the call ?????
Ring number <1 – 10>
FXO

Unity Express Unity Connection Unity
Max Mailboxes 250 7500 7500 per server
Voice Mail Yes Yes Yes
Integrated
Messaging
Yes Yes Yes
Unified Messaging No No Yes
Auto Attendant Yes Yes Yes
Platform Linux router based Windows / Linux
Server
Windows Server
PBX / TDM Support No No Yes
Redundancy No No Yes
AIM-CUE NM-CUE N-CUE-EC NME-CUE
Max
Mailboxes
50 100 250 250
Voice Ports 6 8 16 24
Installation Internal NM Slot NM Slot NM Slot
Storage (hrs) 14 100 300 300
Concurrent
languages
2 5 5 5
Unity Range
Unity Express
CUE Features
Voicemail (User Mailbox). A user/subscriber has his/her own personal mailbox. A pin is required to
login.
Voicemail (General Delivery Mailbox) is a shared mailbox accessible by many subscribers. Subscribers
must be made a member of the GDM to access it and will be prompted to access it when checking their
own personal mailbox. A pin is not required.
IVR (Interactive Voice Prompt) is a system where the system the phone system plays a prompt then
waits for a user to respond. Typical uses are an auto attendant and bank automated balance enquiry.
Auto Attendant allows users to direct themselves to the correct person eg ‘Press 1 for Sales, 2 for
Accounts’. Two scripts are provided with the system ‘Auto Attendant Script’ & ‘Auto Attendant Simple
Script’. By default the following greetings are available ‘Welcome prompt’, ‘Business Open prompt’,
‘Business Closed prompt’ & ‘Holiday prompt’.
Administration via Telephone (AVT) allows an admin to record greetings and prompts.
Backup and restore functionality is provided making use of an FTP server. This requires administrator
access to the web gui.
Message Waiting indicator alerts the user there is a message waiting by flashing a red light and
displaying an envelope on the phone display.

are additional methods of alerting the user there is a message. The notification can be to ring a phone
or send an email.

Troubleshooting
From IOS-
Show interface service-engine 1/0
Service-module service-engine 1/0 status - Should be in a steady state
Show dial-peer voice <tag>
Debug ephone mwi
From CUE
Trace <all/ccn/dns/....>
Show trace buffer
Setup Process
1. Configure IOS service-engine and service-module for IP connectivity.
2. Create SIP dial peer for CUE.
3. Create MWI notification ephone dn’s.
4. Perform initial config – domain name, hostname, NTP servers & admin credentials.
5. Run Initialisation Wizard (import users, MWI methods, voicemail access number, administration by
telephone number etc).
Initial Engine Setup
Once installed a ‘service-engine x/y’ interface appears in the routers config, this is the routers interface
to the Unity Express module. There are two methods of assigning it an IP address-
Method 1
(config) # interface service-engine0/1
(config-if) # ip address 192.168.100.1 255.255.255.252
(config-if) # service-module ip address 192.168.100.2 255.255.255.252
(config-if) # service-module ip default-gateway 192.168.100.1
(config-if) # no shutdown
Method 2
(config ) # interface Loopback1
(config-if) # ip address 192.168.1.1 255.255.255.0
(config) # interface Service-engine0/1
(config-if) # ip unnumbered Loopback1
(config-if) # service-module ip address y.y.y.y y.y.y.y
(config-if) # no shutdown
(config) # ip route y.y.y.y Loopback1
(config) # Ip route 192.168.1.2 255.255.255.255 Service-engine0/1

Controlling / Connecting to the module

# service-module service –engine0/1 sessions - Connect to the module using the specific engine
# service-module service –engine0/1 reload - Reload the module
# service-module service –engine0/1 reset - Reset the module
# service-module service –engine0/1 shutdown - Shutdown the module (before powering off router)
# service-module service –engine0/1 status - Show the status of the CUE module
Initial Configuration of the Module
# service-module service –engine0/1 sessions - Initiates a telnet connection to the module
> enable - enter privileged mode
# offline - Take module offline
# restore factory default
Once restored the unit will reboot and show the prompt-
‘Do you wish to start configuration now (y,n)?’
Enter Host Name?
Enter Domain Name?
Would you like to use DNS for CUE (y,n)?
Enter IP Address of the Primary NTP Server?
Enter IP Address of the Secondary Server?
Please Identify a location so that time zone rules can be set correctly? 1) Africa, 2) Americas .......
Please select a country? 1) Anguilla, 2) Antigua & Barbados ......
Please select one of the following time zones regions. 1) Eastern Time, 2) Eastern Time – Michigan.... **
Is the above information OK? 1) Yes, 2) No
Waiting xxx .....
After booting it prompts for the admin user account details
Enter administrator user ID:
Enter password for :
** US Additional Option
Upgrading CUE
CUE # software install clean url ftp://x.x.x.x/cue-vm-k9.nm-aim.4.2.1.pkg *
Language Installation Menu :
1 ITA, 2 ESP ........ **
# enter the number for the language to sellec one
R # - remove the language for given #
I # - more information about the language for a given ‘
x- Done with language selection
Enter Command:

CUE # software install clean url ftp://x.x.x.x/license

*CUE uses a username and password of ‘anonymous’. Ensure the FTP server has this account setup.
** Corresponding language file must be downloaded as well.
NOTE an upgrade can be performed using the command software download upgrade only from version
2.3.4
Configure CME to access CUE
CME communicates with the CUE using a SIP dial-peer.-
(config) # dial-peer voice 700 voip
(config-dial-peer) # Destination-pattern 7..
(config-dial-peer) # session target ipv4:192.68.100.2
(config-dial-peer) # session protocol sipv2
(config-dial-peer) # dtmf-relay sip-notify - out of band DTMF
(config-dial-peer) # codec g711ulaw
(config-dial-peer) # no vad - Essential
Create the MWI dns’-
(config-ephone-dn) # number 1999...
(config-ephone-dn) # mwi on
(config-ephone-dn) # number 1998...
(config-ephone-dn) # mwi off
The CUE module will call 1999<ext> to turn the MWI on for this dn.
The CUE module will call 1998<ext> to turn the MWI on for this dn.
# Debug ephone mwi
Trace debugging

The Web username and password allows the CUE Module to get the current dn config from CME and
administer it.

Password – Web Interface (GUI)
PIN – Telephone interface(TUI)
Voice Mail Number – This configure the CUE voicemail number and configure the phones message
button to this number.
Auto Attendant Access Number- Configures the CUE AA number.

Administrator via Telephone Number (ATN)

Allows administering the CUE using a telephone number

(message prompt recording etc).

SIP MWI Notification Mechanism – Other options are ‘Subscribe – Notify’ .....

Smart Business Communication System

UC520 UC520 Model UC520
Users 8 or 16 8 or 16 24,32 or 48
1.5u desktop 2u rack
Music on Hold 3.5mm Jack 3.5mm Jack 3.5mm Jack
10/100 PoE 8 (Max 80 watt) 8 (Max 80 watt) 8 (Max 80 watt)
LAN Expansion
10/100
1 1 1
WAN 10/100 1 1 1
FXS 4 4 4
FXO 4 0 4
BRI 0 2 4
T1 / E1 0 0 1
VWIC 1 1 1
Integrated AP Yes Yes No
CE520-8PC CE520-24TT CE520-24LC CE520-24PC CE520G-24T
10/100 24 20
10/100 PoE 8 4 24
10/100/1000 2 2 2 2 24 + 2
UC520 – Central point of the IP based system. Provides routing, security, VPN (10 users), call processing
for 8-48 phones, voicemail & auto-attendant. This is based on CUCME 4.2 and CUE 3.1 for voice mail
CE520 – Catalyst Express Switch family
Cisco 521 – Wireless Express Access Point. This can operate in either standalone mode (mode one) or
Controller based mode (mode two). CCA can manage up to three independent access points.
Cisco 526 – Wireless Express Mobility Controller. Can control up to 6 Cisco 521 Access Points. CCA can
control two controllers allowing for up to 12 AP in a single SBCS deployment.
CCA – Cisco Configuration Assistant, the configuration tool for SBC devices. Default username /
password ‘cisco’ & ‘cisco’
Typical UC520 Models
Typical CE520 Models
The UC520 has the following default configuration
 Data Lan : 192.168.10.0 / 24 VLAN1
 Voice Lan : 10.1.1.0 / 24 VLAN100
 Telephone Ext length : 3
 Out of the box Extensions : 201 – xxx

CCA Communities
CCA can discover devices using three methods-
 FQDN
 IP Address
 Subnet search
Cisco Configuration Assistant Tabs
Device
Displays the platform and installed interfaces (VIC, Wireless, FXO etc)
Options to ‘Configure as a PBX’ or ‘Configure as a Key system’
System
Options for ‘Region’, ‘Phone Language’, ‘Voicemail Language’, Data & Time formats, ‘System Message’ &
System Speed Dials
Network
IP address, DHCP, Voice Vlan
AA & Voicemail
Configure the AA & Voicemail extension pilot number and PSTN numbers. Ability to choose the AA script
and number options
SIP Trunk
Settings to connect to an ITSP (Internet Telephony Service Provider). Registrar, Proxy & MWI Server.
Voice Features
Music on Hold, Paging, Group Pickup, Caller ID Block, Outgoing Call Block Number List, Intercom, Hunt
Group, Call Park, Multi-party Conference
Dial Plan
Number of digits per extension, Outgoing Call Handling (area code, local number etc size). Outgoing
access code (9). Incoming call Handling / DID
Users
User Phone assignment (names, numbers etc)

The Techexams Forums-
http://www.techexams.net/forums/ccna-voice/
Cisco Communications Manager Express Web site-
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/tsd_products_support_series_home.html
Various LABs for Cisco certifications-
http://configurethenetwork.com/

Cisco ccna voice

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