Cisco CCNA Voice
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CCNA Voice IIUC
(640-460)
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CCNA Voice IIUC (640-460)
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
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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
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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
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Legal notice and disclaimer
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,
host, distribute, or commercially exploit the content. If this publication is not obtained from
http://www.caerffili.co.uk/ or http://www.studyshorts.co.uk/ the publication held is considered a
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.
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CCNA Voice IIUC (640-460)
Introduction
Definitions
Term
Definition
FXO
FXS
CO.
Key Switch
Foreign Exchange Office – Connects to a Telco central office
Foreign Exchange Station – Connects to a local analogue phone or a fax
Telco Central Office
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
Private Branch Exchange - Typically uses digital PSTN trunks, provides unique telephone
extensions and have a large feature set
A call between to local ports
A call terminated outside of a local port (PSTN)
Dialed Number Identification Service. A service provider by the Telco to signal the number dialled
by the calling party (Direct Inward Dial)
Automatic Number Identification. Signals the telephone number of the calling party (Caller ID)
A subscriber can access both an email box and a voice mail box using a single client
PBX
Local call
Off net call
DNIS
ANI
Integrated
Messaging
Unified
Messaging
VAD
H.450
TDM
DS0
T1
E1
CAS
CCS
ITU-T
IETF
RTP
RTCP
ACD
CoS
QoS
ToS
TCL
T.37
T.38
A subscribers can access both email and voice mail from a single mail box
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%
Avoids hair-pinning forwarded and Transferred calls
Time Division Multiplexing
A single timeslot / channel. Carries 64kb/s
1.544mbps. 1.536mbps actual data, .008mbps framing. 24 x DS0 channels.
2.048mbps - 32 DS0 channels
Channel Associated Signalling. Signalling is placed in data carrying DS0 channels. Typically called
Robbed Bit Signalling
Common Channel Signalling. A dedicated DS0 timeslot is used for signalling. Commonly called
Primary Rate ISDN
International Telecommunication Union, Telecommunication Standardization Sector
Internet Engineering Task Force
Real-time Transport Protocol. Carries the media stream (even UDP port)
Real-time Transport Control Protocol. Carries statistic information (odd UDP port)
Automatic Call Distributution. Usually used in a call centre environment
Class of Service – Layer 2 process for prioritising traffic
Quality of Service
Type of Service – Layer 3 process for prioritising traffic
Scripting language allows advanced functionality for Auto attendant etc
Fax transmission by transporting the image file using SMTP (store and forward)
Fax Relay over an IP network
Well Known Ports
Protocol
Port
IP
FTP
SHH
20, 21
22
TCP
TCP
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Telnet
SMTP
DNS
DHCP / BOOTP
TFTP
NEWS
NTP
SNMP
23
25
53
67
69
119
123
161, 162
TCP
TCP
TCP, UDP
UDP
UDP
TCP
UDP
UDP
Miscellaneous commands
Mode
Description
Command
#
#
#
#
#
#
(config)
Show layer 1 & 2 info on all interfaces
As above but on specific interface
Show layer 3 info
As above but on specific interface
Show brief interface status
Clear all counters on one or all interfaces
Turn off domain lookups
Show interfaces
Show interfaces interface
Show ip interfaces
Show ip interfaces interface
Show ip interface brief
Clear counters
No ip domain-lookup
Telnet / Session Management
#
#
#
#
(config-line)
Show open sessions from this router
Show open sessions to this router
Kills one of the open sessions from this router
Kills one of the open sessions to this router
Timeout on the particular line connection
#
#
#
(config)
(config-line)
Redirect status messages to the current session
Turn off all debugging
Show log buffer memory stats and messages
Allocate buffer memory for log messages
Stop debug messages corrupting input field
#
#
#
(config-if)
(config)
Show basic info on connected neighbors
Show detailed info on connected neighbors
As above but with wildcards
Disable CDP broadcast on an interface
Disable CDP entirely
Show sessions
Show users
disconnect
Clear line <x>
Exec-timeout minutes seconds
Logging & Debugging
Terminal monitor
u all / undebug all / no debug all
Show logging
logging buffered 32000
Logging synchronous
CDP
Show cdp neighbors
Show cdp neighbours detail
Show cdp entry <name wildcard / *>
No cdp enable
No cdp run
Frequencies of Audio Signals
Human Ear
Human Speech
Telephone Channel
Lower Limit (Hz)
Upper Limit (Hz)
20
200
300
20000
9000
3400
Nyquist Theorem – Frequency sample must be twice the maximum frequency to accurately reconstruct
the original wave form.
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POTS Technologies
Analogue Connections
Two connectionsGround / 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
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Address Signalling
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.
Frame 1
...
Frame 5
Frame 6
1st DS0
...
1st DS0
1st DS0 S
2nd DS0
...
2nd DS0
2nd DS0 S
3rd DS0
...
3rd DS0
3rd DS0 S
...
...
...
...
24th DS0
...
24th DS0
24th DS0 S
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
Frame 6
Frame 12
Frame 18
Frame 24
1st DS0 A
1st DS0 B
1st DS0 C
1st DS0 D
2nd DS0 A
2nd DS0 B
2nd DS0 C
2nd DS0 D
3rd DS0 A
3rd DS0 B
3rd DS0 C
3rd DS0 D
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.
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T1 and E1 CCS Signalling
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.
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IP Voice Technologies
Cisco Voice Infrastructure Model
Layer
1
2
3
4
Purpose
Examples
Endpoints
Applications
Call Processing
Infrastructure
IP Phone, Cell Phone, Video Phone, IM Client
Voice Mail, Conference Call apps, Call Centre Apps, 911 Series
Unified Communications Manager, UCME, UC500
ASA Firewall Voice Router/Gateway, Voice Switch
Call Processing Layer
Max users
Redundancy support
Host
UC500
CME
CCMBE
CCM
48
no
Router
250
No
Router
500
No
Server
30000+
Yes
Server
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
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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.
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SCCP - Cisco proprietary protocol 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
Body
Industry Support
Used on Gateways
Used on Cisco phones
Architecture
H.323
MGCP
SIP
SCCP
ITU
Excellent
Yes
No
Peer to Peer
IETF
Fair
Yes
Limited
Client / Server
IETF
Very Good
Yes
Yes
Peer to Peer
Proprietary
Limited
Yes
Client / Server
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
Version
Header Length
Identification
TTL
Type of Service
Flags
Protocol
Total Length
Fragment Offset
Header Checksum
Source IP Address
Destination Source Address
UDP – 8 bytes
Source Port (16bits)
Length (16bits)
Destination Port (16bits)
Checksum (16bits)
Real-time Transport Protocol (RTP) – 12 bytes
Ver
P X
CC
M
PT
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Sequence Number
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CCNA Voice IIUC (640-460)
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.
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Best Effort – Noncritical – web browsing, email, ftp etc.
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
NotesEnsure 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 onhttp://www.studyshorts.co.uk
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ACL
Input interface
NBAR (Network based application recognition). This looks at the up layers to find the application
Mode
Description
Command
(config)
(config)
(config)
(config-cmap)
(config-cmap)
(config-cmap)
Create a match all class map (default)
Create a match any class map
Create a match all class map
Match on an ACL
Match on an input interface
Match based on NBAR application signature
Class-map classname
Class-map match-any classname
Class-map match-all classname
Match access-group
Match input-interface
Match protocol protocol
Match-any signifies an OR condition between statements
Match-all signifies an AND condition between statements
Policy-map
Controls what to do with traffic
Mode
Description
Command
(config)
(config-pmap)
(config-cmap-c)
(config-cmap-c)
(config-cmap-c)
Create a policy map
Set a class map for this policy
Set a priority bandwidth of kbps
Set a priority bandwidth of percentage of interface bandwidth
Set bandwidth of kbps
Policy-map type policyname
Class classname
Priority kbps
Priority percent percent
Bandwidth kbps
Example(config) # Class-map match-any WEB_TRAFFIC
(config-cmap) # Match protocol http
(config-cmap) # Match protocol https
- Class map to match on either HTTP or HTTPS
(config) # Class-map match-all VOIP
(config-cmap) # Match protocol rtp
- Class map to match on RTP traffic
(config) # policy-map VOIP
(config-pmap) # class VOIP
(config-pmap-c) # priority 4000
- Policy map to give priority bandwidth to VOIP
(config) # interface Ethernet 0
(config-if) # service-policy output VOIP
- Set the QoS on an interface
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Analogue to Digital Conversion / Codecs
Conversion
1.
2.
3.
4.
Sample the waveform – Pulse Amplitude Modulation
Calculate the number representing each sample (quantisation)
Convert to binary – Pulse Code Modulation (G711a etc)
Compress if required
Codec Summary
Codec
Bandwidth
MOS
iLBC
G.711
G.729
G.723.1
G.723.2
G.726
G.726
G.729a
G.728
15.2kbps
64kbps
8kbps
6.3kbps
5.3kbps
32kbps
24kbps
8kbps
16kbps
4.1
4.1
3.92
3.9
3.8
3.85
3.7
3.61
Codec
Delay
Complexity
20ms Sample Size
(bytes)
0.75ms
10ms
30ms
Medium
High
High
160
20
Notes
Most Supported
Medium
10ms
Medium
High
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)
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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 ulaw country.
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CCNA Voice IIUC (640-460)
Numbering Plans
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
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CCNA Voice IIUC (640-460)
Phones
Phone Range
Lines
7906G
7911G
7914/7915/7916
7920
7921
7931
7936
7937
7940G
7941G
7941G-GE
7942G
7945G
7960G
7961G
7961G-GE
7962G
7965G
7970G
7971G-GE
7975G
7985
ATA 186
ATA 188
VG224
VG248
IP Communicator
Unified Personal
Communicator
1
1
14
1
1
24
1
1
2
2
2
2
2
6
6
6
6
6
8
8
8
1
2
2
24
48
8
Switch
XML Apps
PoE
Text Graphics Pre 802.3af
No
Yes
No
No
No
Yes
No
No
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
No
Yes
No
No
-
Yes
Yes
No
Yes
Yes
Yes
No
No
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
No
No
No
No
-
No
No
No
No
Yes
No
No
No
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
No
No
No
No
-
Yes
Yes
No
No
Yes
No
No
No
Yes
Yes
No
Yes
No
Yes
Yes
No
Yes
No
Yes
No
No
No
No
No
No
No
-
Yes
Yes
No
No
Yes
Yes
No
Yes
No
Yes
Yes
Yes
Yes
No
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
No
No
No
No
-
Notes
Expansion Module
802.11b Wifi Phone
A,B & G Wifi, PTT
Conference Station
Conference Station
High res screen
Gig Ethernet
High Quality Audio
High res screen
High res screen
Gig Ethernet
High Quality Audio
High res screen
Colour Touch screen
Colour Touch screen
Colour Touch screen
Video Phone
Dual FXS
Dual FXS
Analogue Gateway :FXS
Analogue Gateway :FXS
Soft Phone
Expansion Module adds an additional 14 lines to a 796x and 797x phones. Up to two units can be added.
Phone Boot Process
1.
2.
3.
4.
5.
6.
Switch detects PoE capabilities and sends power if required.
Phone boots software image.
Switch sends the Voice VLAN info to the phone using CDP.
IP Phone uses DHCP to get its IP address including ‘option 150’ (TFTP IP Address).
Phone contacts TFTP server and gets configuration file.
Phone registers with the CME Server listed in the config file.
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CCNA Voice IIUC (640-460)
Powering
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.
Class
Allocated Power
Actual Power Used
0
1
2
3
15.4W
4.0W
7.0W
15.4W
0.44 to 12.95
0.44 – 3.84W
3.84 – 6.49W
6.49 – 12.95W
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
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CCNA Voice IIUC (640-460)
Basic Configuration
Switch configuration
Mode
Description
Command
#
#
#
#
Show all defined vlans and assigned ports
Show total power available / used and port power usage
Show directly connected Cisco Device information
Show VTP mode and status
Show vlan
Show power inline
Show cdp neighors
Show vtp status
Set Switch Port Trunking Mode
(config-if)
(config-if)
(config-if)
(config-if)
(config-if)
Set the trunk encapsulation (ISL no used much now)
Enable the trunk mode
Auto mode. Will aggressively try to raise a trunk. Default
Auto. Will not raise trunk but will if the other end does.
Set native (untagged) Vlan
(config-if)
(config-if)
(config-if)
(config-if)
Set access port
Set the data vlan
Set the voice/auxiliary vlan
Set STP portfast
Switchport trunk encapsulation dot1q
Switchport mode trunk
Switchport mode dynamic desirable
Switchport mode dynamic auto
Switchport trunk native vlan vlan
Set Switch Port Access Mode
Switchport mode access
Switchport access vlan vlan
Switchport voice vlan vlan
Spanning-tree porftfast
Configure VLAN
(config)
(config-vlan)
Create a vlan
Assign a name to the vlan
Vlan vlannumber
Name name
(config-if)
(config-if)
(config-if)
Set automatic power mode
Turn off PoE
Leave power on for second after link goes down
Misc
Power inline auto
Power inline never
Power inline delay shutdown seconds
NotesAs 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
Mode
Description
Command
#
(config)
(dhcp-config)
(dhcp-config)
(dhcp-config)
(dhcp-config)
(dhcp-config)
(config)
Display DHCP leases
Create a DHCP pool
Define network to enable & issues addressed
Set default router
Set DNS server
Set TFTP server address
Set TFTP server name (not recommended)
Set dhcp excluded addresses
Show ip dhcp binding
Ip dhcp pool pool
Network x.x.x.x /24
Default-router x.x.x.x
Dns-server x.x.x.x
Option 150 ip x.x.x.x
Option 66 ascii tftpservername
Ip dhcp excluded-address x.x.x.x y.y.y.y
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(config-if)
Set helper address for a DHCP server on an interface
Ip helper-address x.x.x.x
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
Mode
Description
Command
#
(config)
(config)
Show NTP sources and status
Set a time server
Set a hour zone and hour difference for the time
Show ntp associations
Ntp server domainname
Clock timezone name x
Configure a router as an NTP Master
(config)
(config)
Allow other devices to get the time from device
Assign an access list to restrict access
Ntp master
Ntp access-group list
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......
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CCNA Voice IIUC (640-460)
CME Communications Manager Express
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 systemBasic 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, either1.
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.
Mode
Description
Command
#
#
#
Show all flash files and free space
Think DOS...
Install CME from TFTP
Show flash
Dir flash:
Archive tar /xtract tftp://x.x.x.x/cme..tar flash:
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Basic CME Configuration
Mode
Description
Command
#
Show telephony-service
Basic Configuration
(config)
(config-telephony)
(config-telephony)
(config-telephony)
Go to telephone service configuration
Maximum directory numbers
Maximum phones on the system (up licenses purchased)
Defines IP address the phones will attempt to register
Telephony-service
Max-dn x
Max-ephones x
Ip source-address x.x.x.x
(config-telephony)
(config-telephony)
Disable automatically registering ephones
Configure ephone-dn to ephone auto assignment
No auto-reg-ephone
Auto assign x to y
(config-telephony)
(config-telephony)
Allow time admin from the GUI
Allow DN admin from GUI. Required for CUE
Time-webedit
Dn-webedit
Auto Registration and DN assignment
‘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
Mode
Description
Command
#
#
(config)
(config-telephony)
(config-telephony)
Show the internal telephony service tftp files
Display the contents of a text file
Define a file for the TFTP server
Define what firmware to load to a phone
Create the configuration files
show telephony-service tftp-bindings
More filename
Tftp-server filename alias name
Load phonemodel filename
Create cnf-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.
ExamplesTftp-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 referencehttp://www.cisco.com/en/US/docs/voice_ip_comm/cucme/requirements/guide/cme43spc.htm
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Phone configuration files
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
Mode
Description
Command
(config)
(config)
(config-ephone-dn)
(config-ephone-dn)
(config-ephone-dn)
Create a single line dn
Create a dual line dn
Assign a number
Assign a secondary/did number
Assign a name for the telephone directory
Ephone-dn tag
Ephone-dn tag dual-line
Number number
Number number secondary number
Name name
(config-ephone-dn)
Preference to use when same number assigned to
many dn’s. Default is 0.
Consider this the last dn in the hunt group. Don’t try
to find another dn.
If any /line channel on a dual line dn is used, don’t
place a second call on the same dn.
Preference x
(config-ephone-dn)
(config-ephone-dn)
Huntstop
Huntstop channel
EPhone
Represents the physical phone.
Mode
#
#
(config)
(config-ephone)
(config-ephone)
(config-ephone)
(config-ephone)
(config-ephone)
Description
Command
Show ephones trying to register. Useful to find phone
MAC when setting up phones
Create an ephone
Assign a MAC
Set phone type. Not required as CME will find this out
Assign a phone line with a dn
Cold reset phone
Warm reset phone
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Show ephone
Show ephone attempted-registrations
Ephone no
Mac-address xxxx.xxxx.xxxx
Type phonemodel
Button x:y
Reset
Restart
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CCNA Voice IIUC (640-460)
Auto Registration & Assignment
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-
Mode From IOS Help
:
S
B
F
M
Normal ring
Silent ring
Call waiting beep, no ring
Feature Mode
Monitor Mode
W
O
C
X
Watch Mode
Overlay Line (no call waiting)
Overlay Line (with call waiting)
Overlay Expansion / Overlay
Description
Silent ring but beep on call waiting
Alternate ring tone for a incoming call
Creates a button which shows the status of the ephone-dn. Also acts
as a speed dial button. Ideal for receptionist
As monitor button but watches the whole phone assigned to the dn
Allows multiple phones at the same time to ring on incoming call
Allows multiple phones at the same time to ring on incoming call
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 availableSingle 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
(config) # Ephone-dn 11 dual-line
(config-ephone-dn) # Number 1010
(config-ephone-dn) # Preference 1
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There is a problem using this approach when using a dual line ephone-dn, 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 10 dual-line
(config-ephone-dn) # Number 1010
(config-ephone-dn) # Preference 0
(config-ephone-dn) # Huntstop channel
(config-ephone-dn) # No huntstop
(config) # Ephone-dn 11 dual-line
(config-ephone-dn) # Number 1010
(config-ephone-dn) # Preference 1
(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) # Number 1010
(config-ephone-dn) # Preference 0
(config-ephone-dn) # No huntstop
(config-ephone-dn) # Exit
(config) #Ephone-dn 11
(config-ephone-dn) # Number 1010
(config-ephone-dn) # Preference 1
(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’.
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Page 28 of 50
CCNA Voice IIUC (640-460)
Additional functions
Voice network Directory (Local Directory on phone)
Mode
Description
Command
(config)
(config-ephone-dn)
(config)
(config-register-dn)
(config)
(config-telephony)
(config-telephony)
(config-telephony)
Select Cisco ephone dn
Assign a name for the telephone directory
Select SIP register dn (for attached sip phones)
Assign a name for the telephone directory
Select telephony service config mode
Set directory sort order (default)
Set directory sort order
Create an entry (for non dn entries – up to 100)
ephone-dn dn
Name name
voice register dn dn
Name name
Telephony-service
Directory first-name-first
Directory last-name-first
Directory entry id number name name
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-
Mode
Description
Command
(config-ephone-dn)
(config-ephone-dn)
(config-ephone-dn)
(config-ephone-dn)
(config-ephone-dn)
(config-register-dn)
(config-register-dn)
(config-telephony)
Forward all calls
Set forward when phone busy
Set forward when phone not answered
Forward call on activated night service
Restrict length of a forward number
Set forward when phone busy
Set forward when phone not answered
Set valid call forward destinations
Call-forward all number
Call-forward busy number
Call-forward noan number timeout seconds
Call-forward night-service number
Call-forward max-length length
call-forward b2bua busy number
call-forward b2bua noan number timeout seconds
Call-forward pattern pattern
‘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.
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Blind transfer – The call is transferred as soon as the transfer number is entered.
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).
Mode
Description
Command
(config-telephony)
(config-telephony)
(config-telephony)
(config-telephony)
Sets blind transfer system using H.450.2
Sets consult transfer system using H.450.2
Sets consult transfer system using proprietary method
Sets the pattern for valid transfers
Transfer-system full-blind
Transfer-system full-consult
Transfer-system local-consult
Transfer-pattern pattern
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 30
(config-ephone-dn) # park-slot reservation-group 1 timeout 10 limit 3 transfer 700
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CCNA Voice IIUC (640-460)
(config) # ephone 1
(config-ephone-dn) # park reservation-group 1
NotesOnce 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
NotesThe 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 20
(config-ephone-dn) # number A100
(config-ephone-dn) # intercom A101 label “Manager”
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(config) # ephone-dn 21
(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 commandBarge-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 25
(config-ephone-dn) # number 3000
(config-ephone-dn) # paging
(config-ephone-dn) # paging ip 239.4.3.4 port 200
(config-ephone-dn) # paging group dnlist
- Unicast paging or
- Multicast paging (cannot use 224.)
- 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.
Mode
Description
Command
(config-telephony)
(config-telephony)
(config-telephony)
(config-telephony)
Define outside of hours on particular day
O of H on day between start & endtime
Define blocked number pattern (up to 100)
Permanent block (24-7) - no exceptions
After-hours date month dayno
After-hours day day starttime endtime
After-hours block pattern no pattern
After-hours block pattern no pattern 7-24
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CCNA Voice IIUC (640-460)
(config-ephone)
(config-ephone)
(config-telephony)
Exempt phone from out of hours block
Set a pin for temporarily removing blocks
Enable login for pins. Parameters not required
After-hours exempt
Pin xxxx
Login timeout mins clear time
ExampleAfter-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
(config-telephony) # time-webedit
- Enable changing dn through the gui
- Enable changing time through the gui
To use simply browse to ‘http://x.x.x.x/ccme.html’
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CCNA Voice IIUC (640-460)
Gateways
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.
Mode
Description
Command
#
#
#
#
#
Show the summary and status of all voice ports
Show the summary and status of all dial peers
Debug the dial peer process
Show all voice calls
Show voice port summary
Show dial-peer voice summary
Debug voip dialpeer
Show voice call summary
show call active voice
Create POTS FXS Dial Peer extension
(config)
(config-dial-peer)
(config-dial-peer)
Create a dial peer
Define the numbers to assign to the port
Assign a physical port to the dial peer
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Dial-peer voice tag pots
Destination-pattern number
Port port
Page 34 of 50
CCNA Voice IIUC (640-460)
(config-dial-peer)
Description description
Create VoIP Outbound Dial Peer
(config)
(config-dial-peer)
(config-dial-peer)
(config-dial-peer)
Create a dial peer
Set the destination pattern
Send matching calls to the remove voip server
Dial-peer voice tag voip
Destination-pattern pattern
Session target ipv4:x.x.x.x
Description description
Create a T1/E1 Outbound Dial Peer
(config)
(config-dial-peer)
(config-dial-peer)
(config-dial-peer)
Create a dial peer
Set the destination pattern
Dial-peer voice tab pots
Destination-pattern pattern
Description description
Port x/x:z
Set the destination port
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-
Wildcard Meaning
.
+
[]
A single digit
One or more instances of
Range of digits
T
Anything
Example Matches
50.
1+
[1-3]111
[14-6]11
[6789]..
9T
500, 501 ... 509
11, 111, 11111111
1111, 2111, 3111
111, 411, 511, 611
6xx, 7xx, 8xx, 9xx
Anything starting with a 9. Wait for inter-digit time out to
dial
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 beLeg 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 order1.
Matched the dialled number (DNIS) using the ‘incoming called-number’ dial peer
configuration command.
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CCNA Voice IIUC (640-460)
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.
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CCNA Voice IIUC (640-460)
Digit Manipulation
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) # Destination-pattern 6...
(config-dial-peer) # No digit-strip
(config-dial-peer) # Prefix 0292011
(config-dial-peer) # Port 1/0:15
(config-dial-peer) # Preference 1
Example 0 for operator
(config) # Num-exp 0 5000
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CCNA Voice IIUC (640-460)
Configuring Voice Ports
Configuring VWIC T1 & E1 cards
Mode
Description
Commands
(config)
Select interface
Controller interface
(config-controller)
(config-controller)
(config-controller)
Set framing (esf most common)
Set coding (b8zs used with esf)
Configure CAS
(config)
(config-controller)
Set the ISDN switch type
Configure CCS
CAS
Framing <sf / esf>
Linecoding <ami / b8zs>
Ds0-group groupnumber timeslots x-y type signalling
CCS
Isdn switch-type .....
Pri-group timeslots x-y
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) # Destination-pattern 6...
(config-dial-peer) # No digit-strip
(config-dial-peer) # Prefix 0292011
(config-dial-peer) # Port 1/0:5
(config-dial-peer) # Preference 1
- 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
Mode
Description
Command
(config-voiceport)
Set start method. Loopstart is default. Used when trunking
to a pbx
Set the dial tone. Also changes the ring cadence accordingly
Change the ringing AC frequency
Set the ring pattern
Set custom ring cadence
Signal <groundstart /
loopstart>
Cptone <countrycode>
Ring frequency <25 / 50>
Ring cadence patternxx
Ring cadence x y z . . . . .
Busyout
Station-id name
Timeouts .....
(config-voiceport)
(config-voiceport)
(config-voiceport)
(config-voiceport)
(config-voiceport)
(config-voiceport)
Set the caller ID Name
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CCNA Voice IIUC (640-460)
FXO
Mode
Description
Command
(config-voiceport)
Set start method. Loopstart is default. Used when
trunking to a pbx
Set the dialling signalling method
Length of time before the router answers the call ?????
Signal <groundstart / loopstart>
(config-voiceport)
(config-voiceport)
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Dial-type <dtmf / pulse>
Ring number <1 – 10>
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CCNA Voice IIUC (640-460)
Unity
Unity Range
Max Mailboxes
Voice Mail
Integrated Messaging
Unified Messaging
Auto Attendant
Platform
PBX / TDM Support
Redundancy
Unity Express
Unity Connection
Unity
250
Yes
Yes
No
Yes
Linux router based
No
No
7500
Yes
Yes
No
Yes
Windows / Linux Server
No
No
7500 per server
Yes
Yes
Yes
Yes
Windows Server
Yes
Yes
Unity Express
AIM-CUE
Max Mailboxes
Voice Ports
Installation
Storage (hrs)
Concurrent languages
50
6
Internal
14
2
NM-CUE
N-CUE-EC
NME-CUE
100
8
NM Slot
100
5
250
16
NM Slot
300
5
250
24
NM Slot
300
5
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.
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CCNA Voice IIUC (640-460)
Message Notifications 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 IOSShow interface service-engine 1/0
Service-module service-engine 1/0 status
Show dial-peer voice <tag>
Debug ephone mwi
- Should be in a steady state
From CUE
Trace <all/ccn/dns/....>
Show trace buffer
Setup Process
1.
2.
3.
4.
5.
Configure IOS service-engine and service-module for IP connectivity.
Create SIP dial peer for CUE.
Create MWI notification ephone dn’s.
Perform initial config – domain name, hostname, NTP servers & admin credentials.
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 addressMethod 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
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CCNA Voice IIUC (640-460)
Controlling / Connecting to the module
# service-module service –engine0/1 sessions
# service-module service –engine0/1 reload
# service-module service –engine0/1 reset
# service-module service –engine0/1 shutdown
# service-module service –engine0/1 status
- Connect to the module using the specific engine
- Reload the module
- Reset the module
- Shutdown the module (before powering off router)
- Show the status of the CUE module
Initial Configuration of the Module
# service-module service –engine0/1 sessions
> enable
# offline
# restore factory default
- Initiates a telnet connection to the module
- enter privileged mode
- Take module offline
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:
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CCNA Voice IIUC (640-460)
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
(config-dial-peer) # codec g711ulaw
(config-dial-peer) # no vad
- out of band DTMF
- Essential
Create the MWI dns’(config) # ephone-dn 120
(config-ephone-dn) # number 1999...
(config-ephone-dn) # mwi on
(config) # ephone-dn 120
(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
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CCNA Voice IIUC (640-460)
CUE Web Interface
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CCNA Voice IIUC (640-460)
Initialisation Wizard
The Web username and password allows the CUE Module to get the current dn config from CME and
administer it.
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CCNA Voice IIUC (640-460)
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.
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CCNA Voice IIUC (640-460)
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’ .....
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CCNA Voice IIUC (640-460)
Smart Business Communication System
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
Users
Music on Hold
10/100 PoE
LAN Expansion
10/100
WAN 10/100
FXS
FXO
BRI
T1 / E1
VWIC
Integrated AP
UC520
UC520 Model
UC520
8 or 16
1.5u desktop
3.5mm Jack
8 (Max 80 watt)
1
8 or 16
3.5mm Jack
8 (Max 80 watt)
1
24,32 or 48
2u rack
3.5mm Jack
8 (Max 80 watt)
1
1
4
4
0
0
1
Yes
1
4
0
2
0
1
Yes
1
4
4
4
1
1
No
Typical CE520 Models
CE520-8PC
10/100
10/100 PoE
10/100/1000
8
2
CE520-24TT
CE520-24LC
CE520-24PC CE520G-24T
24
20
4
2
24
2
2
24 + 2
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
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CCNA Voice IIUC (640-460)
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)
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CCNA Voice IIUC (640-460)
Additional Resources
The Techexams Forumshttp://www.techexams.net/forums/ccna-voice/
Cisco Communications Manager Express Web sitehttp://www.cisco.com/en/US/products/sw/voicesw/ps4625/tsd_products_support_series_home.html
Various LABs for Cisco certificationshttp://configurethenetwork.com/
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Page 50 of 50
(640-460)
Studyshorts
Help get yourself certified with Studyshorts
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CCNA Voice IIUC (640-460)
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
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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
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CCNA Voice IIUC (640-460)
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
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CCNA Voice IIUC (640-460)
Legal notice and disclaimer
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,
host, distribute, or commercially exploit the content. If this publication is not obtained from
http://www.caerffili.co.uk/ or http://www.studyshorts.co.uk/ the publication held is considered a
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.
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CCNA Voice IIUC (640-460)
Introduction
Definitions
Term
Definition
FXO
FXS
CO.
Key Switch
Foreign Exchange Office – Connects to a Telco central office
Foreign Exchange Station – Connects to a local analogue phone or a fax
Telco Central Office
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
Private Branch Exchange - Typically uses digital PSTN trunks, provides unique telephone
extensions and have a large feature set
A call between to local ports
A call terminated outside of a local port (PSTN)
Dialed Number Identification Service. A service provider by the Telco to signal the number dialled
by the calling party (Direct Inward Dial)
Automatic Number Identification. Signals the telephone number of the calling party (Caller ID)
A subscriber can access both an email box and a voice mail box using a single client
PBX
Local call
Off net call
DNIS
ANI
Integrated
Messaging
Unified
Messaging
VAD
H.450
TDM
DS0
T1
E1
CAS
CCS
ITU-T
IETF
RTP
RTCP
ACD
CoS
QoS
ToS
TCL
T.37
T.38
A subscribers can access both email and voice mail from a single mail box
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%
Avoids hair-pinning forwarded and Transferred calls
Time Division Multiplexing
A single timeslot / channel. Carries 64kb/s
1.544mbps. 1.536mbps actual data, .008mbps framing. 24 x DS0 channels.
2.048mbps - 32 DS0 channels
Channel Associated Signalling. Signalling is placed in data carrying DS0 channels. Typically called
Robbed Bit Signalling
Common Channel Signalling. A dedicated DS0 timeslot is used for signalling. Commonly called
Primary Rate ISDN
International Telecommunication Union, Telecommunication Standardization Sector
Internet Engineering Task Force
Real-time Transport Protocol. Carries the media stream (even UDP port)
Real-time Transport Control Protocol. Carries statistic information (odd UDP port)
Automatic Call Distributution. Usually used in a call centre environment
Class of Service – Layer 2 process for prioritising traffic
Quality of Service
Type of Service – Layer 3 process for prioritising traffic
Scripting language allows advanced functionality for Auto attendant etc
Fax transmission by transporting the image file using SMTP (store and forward)
Fax Relay over an IP network
Well Known Ports
Protocol
Port
IP
FTP
SHH
20, 21
22
TCP
TCP
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CCNA Voice IIUC (640-460)
Telnet
SMTP
DNS
DHCP / BOOTP
TFTP
NEWS
NTP
SNMP
23
25
53
67
69
119
123
161, 162
TCP
TCP
TCP, UDP
UDP
UDP
TCP
UDP
UDP
Miscellaneous commands
Mode
Description
Command
#
#
#
#
#
#
(config)
Show layer 1 & 2 info on all interfaces
As above but on specific interface
Show layer 3 info
As above but on specific interface
Show brief interface status
Clear all counters on one or all interfaces
Turn off domain lookups
Show interfaces
Show interfaces interface
Show ip interfaces
Show ip interfaces interface
Show ip interface brief
Clear counters
No ip domain-lookup
Telnet / Session Management
#
#
#
#
(config-line)
Show open sessions from this router
Show open sessions to this router
Kills one of the open sessions from this router
Kills one of the open sessions to this router
Timeout on the particular line connection
#
#
#
(config)
(config-line)
Redirect status messages to the current session
Turn off all debugging
Show log buffer memory stats and messages
Allocate buffer memory for log messages
Stop debug messages corrupting input field
#
#
#
(config-if)
(config)
Show basic info on connected neighbors
Show detailed info on connected neighbors
As above but with wildcards
Disable CDP broadcast on an interface
Disable CDP entirely
Show sessions
Show users
disconnect
Clear line <x>
Exec-timeout minutes seconds
Logging & Debugging
Terminal monitor
u all / undebug all / no debug all
Show logging
logging buffered 32000
Logging synchronous
CDP
Show cdp neighbors
Show cdp neighbours detail
Show cdp entry <name wildcard / *>
No cdp enable
No cdp run
Frequencies of Audio Signals
Human Ear
Human Speech
Telephone Channel
Lower Limit (Hz)
Upper Limit (Hz)
20
200
300
20000
9000
3400
Nyquist Theorem – Frequency sample must be twice the maximum frequency to accurately reconstruct
the original wave form.
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Page 7 of 50
CCNA Voice IIUC (640-460)
POTS Technologies
Analogue Connections
Two connectionsGround / 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
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Address Signalling
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.
Frame 1
...
Frame 5
Frame 6
1st DS0
...
1st DS0
1st DS0 S
2nd DS0
...
2nd DS0
2nd DS0 S
3rd DS0
...
3rd DS0
3rd DS0 S
...
...
...
...
24th DS0
...
24th DS0
24th DS0 S
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
Frame 6
Frame 12
Frame 18
Frame 24
1st DS0 A
1st DS0 B
1st DS0 C
1st DS0 D
2nd DS0 A
2nd DS0 B
2nd DS0 C
2nd DS0 D
3rd DS0 A
3rd DS0 B
3rd DS0 C
3rd DS0 D
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.
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CCNA Voice IIUC (640-460)
T1 and E1 CCS Signalling
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.
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CCNA Voice IIUC (640-460)
IP Voice Technologies
Cisco Voice Infrastructure Model
Layer
1
2
3
4
Purpose
Examples
Endpoints
Applications
Call Processing
Infrastructure
IP Phone, Cell Phone, Video Phone, IM Client
Voice Mail, Conference Call apps, Call Centre Apps, 911 Series
Unified Communications Manager, UCME, UC500
ASA Firewall Voice Router/Gateway, Voice Switch
Call Processing Layer
Max users
Redundancy support
Host
UC500
CME
CCMBE
CCM
48
no
Router
250
No
Router
500
No
Server
30000+
Yes
Server
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
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CCNA Voice IIUC (640-460)
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.
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CCNA Voice IIUC (640-460)
SCCP - Cisco proprietary protocol 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
Body
Industry Support
Used on Gateways
Used on Cisco phones
Architecture
H.323
MGCP
SIP
SCCP
ITU
Excellent
Yes
No
Peer to Peer
IETF
Fair
Yes
Limited
Client / Server
IETF
Very Good
Yes
Yes
Peer to Peer
Proprietary
Limited
Yes
Client / Server
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
Version
Header Length
Identification
TTL
Type of Service
Flags
Protocol
Total Length
Fragment Offset
Header Checksum
Source IP Address
Destination Source Address
UDP – 8 bytes
Source Port (16bits)
Length (16bits)
Destination Port (16bits)
Checksum (16bits)
Real-time Transport Protocol (RTP) – 12 bytes
Ver
P X
CC
M
PT
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Sequence Number
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CCNA Voice IIUC (640-460)
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.
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Best Effort – Noncritical – web browsing, email, ftp etc.
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
NotesEnsure 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 onhttp://www.studyshorts.co.uk
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ACL
Input interface
NBAR (Network based application recognition). This looks at the up layers to find the application
Mode
Description
Command
(config)
(config)
(config)
(config-cmap)
(config-cmap)
(config-cmap)
Create a match all class map (default)
Create a match any class map
Create a match all class map
Match on an ACL
Match on an input interface
Match based on NBAR application signature
Class-map classname
Class-map match-any classname
Class-map match-all classname
Match access-group
Match input-interface
Match protocol protocol
Match-any signifies an OR condition between statements
Match-all signifies an AND condition between statements
Policy-map
Controls what to do with traffic
Mode
Description
Command
(config)
(config-pmap)
(config-cmap-c)
(config-cmap-c)
(config-cmap-c)
Create a policy map
Set a class map for this policy
Set a priority bandwidth of kbps
Set a priority bandwidth of percentage of interface bandwidth
Set bandwidth of kbps
Policy-map type policyname
Class classname
Priority kbps
Priority percent percent
Bandwidth kbps
Example(config) # Class-map match-any WEB_TRAFFIC
(config-cmap) # Match protocol http
(config-cmap) # Match protocol https
- Class map to match on either HTTP or HTTPS
(config) # Class-map match-all VOIP
(config-cmap) # Match protocol rtp
- Class map to match on RTP traffic
(config) # policy-map VOIP
(config-pmap) # class VOIP
(config-pmap-c) # priority 4000
- Policy map to give priority bandwidth to VOIP
(config) # interface Ethernet 0
(config-if) # service-policy output VOIP
- Set the QoS on an interface
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CCNA Voice IIUC (640-460)
Analogue to Digital Conversion / Codecs
Conversion
1.
2.
3.
4.
Sample the waveform – Pulse Amplitude Modulation
Calculate the number representing each sample (quantisation)
Convert to binary – Pulse Code Modulation (G711a etc)
Compress if required
Codec Summary
Codec
Bandwidth
MOS
iLBC
G.711
G.729
G.723.1
G.723.2
G.726
G.726
G.729a
G.728
15.2kbps
64kbps
8kbps
6.3kbps
5.3kbps
32kbps
24kbps
8kbps
16kbps
4.1
4.1
3.92
3.9
3.8
3.85
3.7
3.61
Codec
Delay
Complexity
20ms Sample Size
(bytes)
0.75ms
10ms
30ms
Medium
High
High
160
20
Notes
Most Supported
Medium
10ms
Medium
High
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)
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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 ulaw country.
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CCNA Voice IIUC (640-460)
Numbering Plans
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
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CCNA Voice IIUC (640-460)
Phones
Phone Range
Lines
7906G
7911G
7914/7915/7916
7920
7921
7931
7936
7937
7940G
7941G
7941G-GE
7942G
7945G
7960G
7961G
7961G-GE
7962G
7965G
7970G
7971G-GE
7975G
7985
ATA 186
ATA 188
VG224
VG248
IP Communicator
Unified Personal
Communicator
1
1
14
1
1
24
1
1
2
2
2
2
2
6
6
6
6
6
8
8
8
1
2
2
24
48
8
Switch
XML Apps
PoE
Text Graphics Pre 802.3af
No
Yes
No
No
No
Yes
No
No
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
No
Yes
No
No
-
Yes
Yes
No
Yes
Yes
Yes
No
No
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
No
No
No
No
-
No
No
No
No
Yes
No
No
No
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
No
No
No
No
-
Yes
Yes
No
No
Yes
No
No
No
Yes
Yes
No
Yes
No
Yes
Yes
No
Yes
No
Yes
No
No
No
No
No
No
No
-
Yes
Yes
No
No
Yes
Yes
No
Yes
No
Yes
Yes
Yes
Yes
No
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
No
No
No
No
-
Notes
Expansion Module
802.11b Wifi Phone
A,B & G Wifi, PTT
Conference Station
Conference Station
High res screen
Gig Ethernet
High Quality Audio
High res screen
High res screen
Gig Ethernet
High Quality Audio
High res screen
Colour Touch screen
Colour Touch screen
Colour Touch screen
Video Phone
Dual FXS
Dual FXS
Analogue Gateway :FXS
Analogue Gateway :FXS
Soft Phone
Expansion Module adds an additional 14 lines to a 796x and 797x phones. Up to two units can be added.
Phone Boot Process
1.
2.
3.
4.
5.
6.
Switch detects PoE capabilities and sends power if required.
Phone boots software image.
Switch sends the Voice VLAN info to the phone using CDP.
IP Phone uses DHCP to get its IP address including ‘option 150’ (TFTP IP Address).
Phone contacts TFTP server and gets configuration file.
Phone registers with the CME Server listed in the config file.
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Powering
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.
Class
Allocated Power
Actual Power Used
0
1
2
3
15.4W
4.0W
7.0W
15.4W
0.44 to 12.95
0.44 – 3.84W
3.84 – 6.49W
6.49 – 12.95W
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
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Basic Configuration
Switch configuration
Mode
Description
Command
#
#
#
#
Show all defined vlans and assigned ports
Show total power available / used and port power usage
Show directly connected Cisco Device information
Show VTP mode and status
Show vlan
Show power inline
Show cdp neighors
Show vtp status
Set Switch Port Trunking Mode
(config-if)
(config-if)
(config-if)
(config-if)
(config-if)
Set the trunk encapsulation (ISL no used much now)
Enable the trunk mode
Auto mode. Will aggressively try to raise a trunk. Default
Auto. Will not raise trunk but will if the other end does.
Set native (untagged) Vlan
(config-if)
(config-if)
(config-if)
(config-if)
Set access port
Set the data vlan
Set the voice/auxiliary vlan
Set STP portfast
Switchport trunk encapsulation dot1q
Switchport mode trunk
Switchport mode dynamic desirable
Switchport mode dynamic auto
Switchport trunk native vlan vlan
Set Switch Port Access Mode
Switchport mode access
Switchport access vlan vlan
Switchport voice vlan vlan
Spanning-tree porftfast
Configure VLAN
(config)
(config-vlan)
Create a vlan
Assign a name to the vlan
Vlan vlannumber
Name name
(config-if)
(config-if)
(config-if)
Set automatic power mode
Turn off PoE
Leave power on for second after link goes down
Misc
Power inline auto
Power inline never
Power inline delay shutdown seconds
NotesAs 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
Mode
Description
Command
#
(config)
(dhcp-config)
(dhcp-config)
(dhcp-config)
(dhcp-config)
(dhcp-config)
(config)
Display DHCP leases
Create a DHCP pool
Define network to enable & issues addressed
Set default router
Set DNS server
Set TFTP server address
Set TFTP server name (not recommended)
Set dhcp excluded addresses
Show ip dhcp binding
Ip dhcp pool pool
Network x.x.x.x /24
Default-router x.x.x.x
Dns-server x.x.x.x
Option 150 ip x.x.x.x
Option 66 ascii tftpservername
Ip dhcp excluded-address x.x.x.x y.y.y.y
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(config-if)
Set helper address for a DHCP server on an interface
Ip helper-address x.x.x.x
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
Mode
Description
Command
#
(config)
(config)
Show NTP sources and status
Set a time server
Set a hour zone and hour difference for the time
Show ntp associations
Ntp server domainname
Clock timezone name x
Configure a router as an NTP Master
(config)
(config)
Allow other devices to get the time from device
Assign an access list to restrict access
Ntp master
Ntp access-group list
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......
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CME Communications Manager Express
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 systemBasic 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, either1.
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.
Mode
Description
Command
#
#
#
Show all flash files and free space
Think DOS...
Install CME from TFTP
Show flash
Dir flash:
Archive tar /xtract tftp://x.x.x.x/cme..tar flash:
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Basic CME Configuration
Mode
Description
Command
#
Show telephony-service
Basic Configuration
(config)
(config-telephony)
(config-telephony)
(config-telephony)
Go to telephone service configuration
Maximum directory numbers
Maximum phones on the system (up licenses purchased)
Defines IP address the phones will attempt to register
Telephony-service
Max-dn x
Max-ephones x
Ip source-address x.x.x.x
(config-telephony)
(config-telephony)
Disable automatically registering ephones
Configure ephone-dn to ephone auto assignment
No auto-reg-ephone
Auto assign x to y
(config-telephony)
(config-telephony)
Allow time admin from the GUI
Allow DN admin from GUI. Required for CUE
Time-webedit
Dn-webedit
Auto Registration and DN assignment
‘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
Mode
Description
Command
#
#
(config)
(config-telephony)
(config-telephony)
Show the internal telephony service tftp files
Display the contents of a text file
Define a file for the TFTP server
Define what firmware to load to a phone
Create the configuration files
show telephony-service tftp-bindings
More filename
Tftp-server filename alias name
Load phonemodel filename
Create cnf-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.
ExamplesTftp-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 referencehttp://www.cisco.com/en/US/docs/voice_ip_comm/cucme/requirements/guide/cme43spc.htm
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Phone configuration files
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
Mode
Description
Command
(config)
(config)
(config-ephone-dn)
(config-ephone-dn)
(config-ephone-dn)
Create a single line dn
Create a dual line dn
Assign a number
Assign a secondary/did number
Assign a name for the telephone directory
Ephone-dn tag
Ephone-dn tag dual-line
Number number
Number number secondary number
Name name
(config-ephone-dn)
Preference to use when same number assigned to
many dn’s. Default is 0.
Consider this the last dn in the hunt group. Don’t try
to find another dn.
If any /line channel on a dual line dn is used, don’t
place a second call on the same dn.
Preference x
(config-ephone-dn)
(config-ephone-dn)
Huntstop
Huntstop channel
EPhone
Represents the physical phone.
Mode
#
#
(config)
(config-ephone)
(config-ephone)
(config-ephone)
(config-ephone)
(config-ephone)
Description
Command
Show ephones trying to register. Useful to find phone
MAC when setting up phones
Create an ephone
Assign a MAC
Set phone type. Not required as CME will find this out
Assign a phone line with a dn
Cold reset phone
Warm reset phone
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Show ephone
Show ephone attempted-registrations
Ephone no
Mac-address xxxx.xxxx.xxxx
Type phonemodel
Button x:y
Reset
Restart
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Auto Registration & Assignment
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-
Mode From IOS Help
:
S
B
F
M
Normal ring
Silent ring
Call waiting beep, no ring
Feature Mode
Monitor Mode
W
O
C
X
Watch Mode
Overlay Line (no call waiting)
Overlay Line (with call waiting)
Overlay Expansion / Overlay
Description
Silent ring but beep on call waiting
Alternate ring tone for a incoming call
Creates a button which shows the status of the ephone-dn. Also acts
as a speed dial button. Ideal for receptionist
As monitor button but watches the whole phone assigned to the dn
Allows multiple phones at the same time to ring on incoming call
Allows multiple phones at the same time to ring on incoming call
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 availableSingle 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
(config) # Ephone-dn 11 dual-line
(config-ephone-dn) # Number 1010
(config-ephone-dn) # Preference 1
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There is a problem using this approach when using a dual line ephone-dn, 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 10 dual-line
(config-ephone-dn) # Number 1010
(config-ephone-dn) # Preference 0
(config-ephone-dn) # Huntstop channel
(config-ephone-dn) # No huntstop
(config) # Ephone-dn 11 dual-line
(config-ephone-dn) # Number 1010
(config-ephone-dn) # Preference 1
(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) # Number 1010
(config-ephone-dn) # Preference 0
(config-ephone-dn) # No huntstop
(config-ephone-dn) # Exit
(config) #Ephone-dn 11
(config-ephone-dn) # Number 1010
(config-ephone-dn) # Preference 1
(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’.
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Additional functions
Voice network Directory (Local Directory on phone)
Mode
Description
Command
(config)
(config-ephone-dn)
(config)
(config-register-dn)
(config)
(config-telephony)
(config-telephony)
(config-telephony)
Select Cisco ephone dn
Assign a name for the telephone directory
Select SIP register dn (for attached sip phones)
Assign a name for the telephone directory
Select telephony service config mode
Set directory sort order (default)
Set directory sort order
Create an entry (for non dn entries – up to 100)
ephone-dn dn
Name name
voice register dn dn
Name name
Telephony-service
Directory first-name-first
Directory last-name-first
Directory entry id number name name
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-
Mode
Description
Command
(config-ephone-dn)
(config-ephone-dn)
(config-ephone-dn)
(config-ephone-dn)
(config-ephone-dn)
(config-register-dn)
(config-register-dn)
(config-telephony)
Forward all calls
Set forward when phone busy
Set forward when phone not answered
Forward call on activated night service
Restrict length of a forward number
Set forward when phone busy
Set forward when phone not answered
Set valid call forward destinations
Call-forward all number
Call-forward busy number
Call-forward noan number timeout seconds
Call-forward night-service number
Call-forward max-length length
call-forward b2bua busy number
call-forward b2bua noan number timeout seconds
Call-forward pattern pattern
‘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.
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Blind transfer – The call is transferred as soon as the transfer number is entered.
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).
Mode
Description
Command
(config-telephony)
(config-telephony)
(config-telephony)
(config-telephony)
Sets blind transfer system using H.450.2
Sets consult transfer system using H.450.2
Sets consult transfer system using proprietary method
Sets the pattern for valid transfers
Transfer-system full-blind
Transfer-system full-consult
Transfer-system local-consult
Transfer-pattern pattern
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 30
(config-ephone-dn) # park-slot reservation-group 1 timeout 10 limit 3 transfer 700
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(config) # ephone 1
(config-ephone-dn) # park reservation-group 1
NotesOnce 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
NotesThe 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 20
(config-ephone-dn) # number A100
(config-ephone-dn) # intercom A101 label “Manager”
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(config) # ephone-dn 21
(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 commandBarge-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 25
(config-ephone-dn) # number 3000
(config-ephone-dn) # paging
(config-ephone-dn) # paging ip 239.4.3.4 port 200
(config-ephone-dn) # paging group dnlist
- Unicast paging or
- Multicast paging (cannot use 224.)
- 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.
Mode
Description
Command
(config-telephony)
(config-telephony)
(config-telephony)
(config-telephony)
Define outside of hours on particular day
O of H on day between start & endtime
Define blocked number pattern (up to 100)
Permanent block (24-7) - no exceptions
After-hours date month dayno
After-hours day day starttime endtime
After-hours block pattern no pattern
After-hours block pattern no pattern 7-24
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(config-ephone)
(config-ephone)
(config-telephony)
Exempt phone from out of hours block
Set a pin for temporarily removing blocks
Enable login for pins. Parameters not required
After-hours exempt
Pin xxxx
Login timeout mins clear time
ExampleAfter-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
(config-telephony) # time-webedit
- Enable changing dn through the gui
- Enable changing time through the gui
To use simply browse to ‘http://x.x.x.x/ccme.html’
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Gateways
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.
Mode
Description
Command
#
#
#
#
#
Show the summary and status of all voice ports
Show the summary and status of all dial peers
Debug the dial peer process
Show all voice calls
Show voice port summary
Show dial-peer voice summary
Debug voip dialpeer
Show voice call summary
show call active voice
Create POTS FXS Dial Peer extension
(config)
(config-dial-peer)
(config-dial-peer)
Create a dial peer
Define the numbers to assign to the port
Assign a physical port to the dial peer
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Dial-peer voice tag pots
Destination-pattern number
Port port
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(config-dial-peer)
Description description
Create VoIP Outbound Dial Peer
(config)
(config-dial-peer)
(config-dial-peer)
(config-dial-peer)
Create a dial peer
Set the destination pattern
Send matching calls to the remove voip server
Dial-peer voice tag voip
Destination-pattern pattern
Session target ipv4:x.x.x.x
Description description
Create a T1/E1 Outbound Dial Peer
(config)
(config-dial-peer)
(config-dial-peer)
(config-dial-peer)
Create a dial peer
Set the destination pattern
Dial-peer voice tab pots
Destination-pattern pattern
Description description
Port x/x:z
Set the destination port
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-
Wildcard Meaning
.
+
[]
A single digit
One or more instances of
Range of digits
T
Anything
Example Matches
50.
1+
[1-3]111
[14-6]11
[6789]..
9T
500, 501 ... 509
11, 111, 11111111
1111, 2111, 3111
111, 411, 511, 611
6xx, 7xx, 8xx, 9xx
Anything starting with a 9. Wait for inter-digit time out to
dial
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 beLeg 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 order1.
Matched the dialled number (DNIS) using the ‘incoming called-number’ dial peer
configuration command.
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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.
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CCNA Voice IIUC (640-460)
Digit Manipulation
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) # Destination-pattern 6...
(config-dial-peer) # No digit-strip
(config-dial-peer) # Prefix 0292011
(config-dial-peer) # Port 1/0:15
(config-dial-peer) # Preference 1
Example 0 for operator
(config) # Num-exp 0 5000
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CCNA Voice IIUC (640-460)
Configuring Voice Ports
Configuring VWIC T1 & E1 cards
Mode
Description
Commands
(config)
Select interface
Controller interface
(config-controller)
(config-controller)
(config-controller)
Set framing (esf most common)
Set coding (b8zs used with esf)
Configure CAS
(config)
(config-controller)
Set the ISDN switch type
Configure CCS
CAS
Framing <sf / esf>
Linecoding <ami / b8zs>
Ds0-group groupnumber timeslots x-y type signalling
CCS
Isdn switch-type .....
Pri-group timeslots x-y
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) # Destination-pattern 6...
(config-dial-peer) # No digit-strip
(config-dial-peer) # Prefix 0292011
(config-dial-peer) # Port 1/0:5
(config-dial-peer) # Preference 1
- 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
Mode
Description
Command
(config-voiceport)
Set start method. Loopstart is default. Used when trunking
to a pbx
Set the dial tone. Also changes the ring cadence accordingly
Change the ringing AC frequency
Set the ring pattern
Set custom ring cadence
Signal <groundstart /
loopstart>
Cptone <countrycode>
Ring frequency <25 / 50>
Ring cadence patternxx
Ring cadence x y z . . . . .
Busyout
Station-id name
Timeouts .....
(config-voiceport)
(config-voiceport)
(config-voiceport)
(config-voiceport)
(config-voiceport)
(config-voiceport)
Set the caller ID Name
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CCNA Voice IIUC (640-460)
FXO
Mode
Description
Command
(config-voiceport)
Set start method. Loopstart is default. Used when
trunking to a pbx
Set the dialling signalling method
Length of time before the router answers the call ?????
Signal <groundstart / loopstart>
(config-voiceport)
(config-voiceport)
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Dial-type <dtmf / pulse>
Ring number <1 – 10>
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CCNA Voice IIUC (640-460)
Unity
Unity Range
Max Mailboxes
Voice Mail
Integrated Messaging
Unified Messaging
Auto Attendant
Platform
PBX / TDM Support
Redundancy
Unity Express
Unity Connection
Unity
250
Yes
Yes
No
Yes
Linux router based
No
No
7500
Yes
Yes
No
Yes
Windows / Linux Server
No
No
7500 per server
Yes
Yes
Yes
Yes
Windows Server
Yes
Yes
Unity Express
AIM-CUE
Max Mailboxes
Voice Ports
Installation
Storage (hrs)
Concurrent languages
50
6
Internal
14
2
NM-CUE
N-CUE-EC
NME-CUE
100
8
NM Slot
100
5
250
16
NM Slot
300
5
250
24
NM Slot
300
5
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.
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CCNA Voice IIUC (640-460)
Message Notifications 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 IOSShow interface service-engine 1/0
Service-module service-engine 1/0 status
Show dial-peer voice <tag>
Debug ephone mwi
- Should be in a steady state
From CUE
Trace <all/ccn/dns/....>
Show trace buffer
Setup Process
1.
2.
3.
4.
5.
Configure IOS service-engine and service-module for IP connectivity.
Create SIP dial peer for CUE.
Create MWI notification ephone dn’s.
Perform initial config – domain name, hostname, NTP servers & admin credentials.
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 addressMethod 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
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Controlling / Connecting to the module
# service-module service –engine0/1 sessions
# service-module service –engine0/1 reload
# service-module service –engine0/1 reset
# service-module service –engine0/1 shutdown
# service-module service –engine0/1 status
- Connect to the module using the specific engine
- Reload the module
- Reset the module
- Shutdown the module (before powering off router)
- Show the status of the CUE module
Initial Configuration of the Module
# service-module service –engine0/1 sessions
> enable
# offline
# restore factory default
- Initiates a telnet connection to the module
- enter privileged mode
- Take module offline
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:
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CCNA Voice IIUC (640-460)
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
(config-dial-peer) # codec g711ulaw
(config-dial-peer) # no vad
- out of band DTMF
- Essential
Create the MWI dns’(config) # ephone-dn 120
(config-ephone-dn) # number 1999...
(config-ephone-dn) # mwi on
(config) # ephone-dn 120
(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
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CCNA Voice IIUC (640-460)
CUE Web Interface
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CCNA Voice IIUC (640-460)
Initialisation Wizard
The Web username and password allows the CUE Module to get the current dn config from CME and
administer it.
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CCNA Voice IIUC (640-460)
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.
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CCNA Voice IIUC (640-460)
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’ .....
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CCNA Voice IIUC (640-460)
Smart Business Communication System
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
Users
Music on Hold
10/100 PoE
LAN Expansion
10/100
WAN 10/100
FXS
FXO
BRI
T1 / E1
VWIC
Integrated AP
UC520
UC520 Model
UC520
8 or 16
1.5u desktop
3.5mm Jack
8 (Max 80 watt)
1
8 or 16
3.5mm Jack
8 (Max 80 watt)
1
24,32 or 48
2u rack
3.5mm Jack
8 (Max 80 watt)
1
1
4
4
0
0
1
Yes
1
4
0
2
0
1
Yes
1
4
4
4
1
1
No
Typical CE520 Models
CE520-8PC
10/100
10/100 PoE
10/100/1000
8
2
CE520-24TT
CE520-24LC
CE520-24PC CE520G-24T
24
20
4
2
24
2
2
24 + 2
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
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CCNA Voice IIUC (640-460)
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)
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CCNA Voice IIUC (640-460)
Additional Resources
The Techexams Forumshttp://www.techexams.net/forums/ccna-voice/
Cisco Communications Manager Express Web sitehttp://www.cisco.com/en/US/products/sw/voicesw/ps4625/tsd_products_support_series_home.html
Various LABs for Cisco certificationshttp://configurethenetwork.com/
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