Professional Documents
Culture Documents
Cisco CCNA Voice PDF
Cisco CCNA Voice PDF
(640-460)
Studyshorts
Help get yourself certified with Studyshorts
www.studyshorts.co.uk
M Morgan 2010
Page 2 of 50
M Morgan 2010
Page 3 of 50
http://www.studyshorts.co.uk
M Morgan 2010
Page 4 of 50
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.
http://www.studyshorts.co.uk
M Morgan 2010
Page 5 of 50
Introduction
Definitions
Term
Definition
FXO
FXS
CO.
Key Switch
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
Port
IP
FTP
SHH
20, 21
22
TCP
TCP
http://www.studyshorts.co.uk
M Morgan 2010
Page 6 of 50
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 interfaces
Show interfaces interface
Show ip interfaces
Show ip interfaces interface
Show ip interface brief
Clear counters
No ip domain-lookup
#
#
#
(config)
(config-line)
#
#
#
(config-if)
(config)
Show sessions
Show users
disconnect
Clear line <x>
Exec-timeout minutes seconds
CDP
Show cdp neighbors
Show cdp neighbours detail
Show cdp entry <name wildcard / *>
No cdp enable
No cdp run
Human Ear
Human Speech
Telephone Channel
20
200
300
20000
9000
3400
Nyquist Theorem Frequency sample must be twice the maximum frequency to accurately reconstruct
the original wave form.
http://www.studyshorts.co.uk
M Morgan 2010
Page 7 of 50
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
http://www.studyshorts.co.uk
M Morgan 2010
Page 8 of 50
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.
http://www.studyshorts.co.uk
M Morgan 2010
Page 9 of 50
http://www.studyshorts.co.uk
M Morgan 2010
Page 10 of 50
IP Voice Technologies
Cisco Voice Infrastructure Model
Layer
1
2
3
4
Purpose
Examples
Endpoints
Applications
Call Processing
Infrastructure
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
http://www.studyshorts.co.uk
M Morgan 2010
Page 11 of 50
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.
http://www.studyshorts.co.uk
M Morgan 2010
Page 12 of 50
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)
Sequence Number
Page 13 of 50
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)
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.
http://www.studyshorts.co.uk
M Morgan 2010
Page 14 of 50
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.
M Morgan 2010
Page 15 of 50
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)
Class-map classname
Class-map match-any classname
Class-map match-all classname
Match access-group
Match input-interface
Match protocol protocol
Mode
Description
Command
(config)
(config-pmap)
(config-cmap-c)
(config-cmap-c)
(config-cmap-c)
http://www.studyshorts.co.uk
M Morgan 2010
Page 16 of 50
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
0.75ms
10ms
30ms
Medium
High
High
160
20
Notes
Most Supported
Medium
10ms
Medium
High
G711
Two types -law (North America & Japan)
A-law (Europe and reset of World)
http://www.studyshorts.co.uk
M Morgan 2010
Page 17 of 50
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).
http://www.studyshorts.co.uk
M Morgan 2010
Page 18 of 50
Numbering Plans
PSTN Numbering Plan
ITU-T E.164
Country Code
National Destination Code
Subscriber Number
http://www.studyshorts.co.uk
M Morgan 2010
Page 19 of 50
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.
http://www.studyshorts.co.uk
M Morgan 2010
Page 20 of 50
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
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
http://www.studyshorts.co.uk
M Morgan 2010
Page 21 of 50
Basic Configuration
Switch configuration
Mode
Description
Command
#
#
#
#
Show vlan
Show power inline
Show cdp neighors
Show vtp status
(config-if)
(config-if)
(config-if)
(config-if)
Configure VLAN
(config)
(config-vlan)
Create a vlan
Assign a name to the vlan
Vlan vlannumber
Name name
(config-if)
(config-if)
(config-if)
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)
http://www.studyshorts.co.uk
M Morgan 2010
Page 22 of 50
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)
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......
http://www.studyshorts.co.uk
M Morgan 2010
Page 23 of 50
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.
2.
3.
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 flash
Dir flash:
Archive tar /xtract tftp://x.x.x.x/cme..tar flash:
http://www.studyshorts.co.uk
M Morgan 2010
Page 24 of 50
Description
Command
Show telephony-service
Basic Configuration
(config)
(config-telephony)
(config-telephony)
(config-telephony)
Telephony-service
Max-dn x
Max-ephones x
Ip source-address x.x.x.x
(config-telephony)
(config-telephony)
No auto-reg-ephone
Auto assign x to y
(config-telephony)
(config-telephony)
Time-webedit
Dn-webedit
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.
Description
Command
#
#
(config)
(config-telephony)
(config-telephony)
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
http://www.studyshorts.co.uk
M Morgan 2010
Page 25 of 50
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)
Ephone-dn tag
Ephone-dn tag dual-line
Number number
Number number secondary number
Name name
(config-ephone-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
http://www.studyshorts.co.uk
M Morgan 2010
Show ephone
Show ephone attempted-registrations
Ephone no
Mac-address xxxx.xxxx.xxxx
Type phonemodel
Button x:y
Reset
Restart
Page 26 of 50
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
http://www.studyshorts.co.uk
M Morgan 2010
Page 27 of 50
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-dns are single line and dont 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.
http://www.studyshorts.co.uk
M Morgan 2010
Page 28 of 50
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)
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)
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.
http://www.studyshorts.co.uk
M Morgan 2010
Page 29 of 50
Mode
Description
Command
(config-telephony)
(config-telephony)
(config-telephony)
(config-telephony)
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
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
http://www.studyshorts.co.uk
M Morgan 2010
Page 30 of 50
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
http://www.studyshorts.co.uk
M Morgan 2010
Page 31 of 50
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
Mode
Description
Command
(config-telephony)
(config-telephony)
(config-telephony)
(config-telephony)
http://www.studyshorts.co.uk
M Morgan 2010
Page 32 of 50
After-hours exempt
Pin xxxx
Login timeout mins clear time
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
http://www.studyshorts.co.uk
M Morgan 2010
Page 33 of 50
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-
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
#
#
#
#
#
http://www.studyshorts.co.uk
M Morgan 2010
Page 34 of 50
Description description
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
Anything
Example Matches
50.
1+
[1-3]111
[14-6]11
[6789]..
9T
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.
http://www.studyshorts.co.uk
M Morgan 2010
Page 35 of 50
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.
http://www.studyshorts.co.uk
M Morgan 2010
Page 36 of 50
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
http://www.studyshorts.co.uk
M Morgan 2010
Page 37 of 50
Description
Commands
(config)
Select interface
Controller interface
(config-controller)
(config-controller)
(config-controller)
(config)
(config-controller)
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
Mode
Description
Command
(config-voiceport)
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)
http://www.studyshorts.co.uk
M Morgan 2010
Page 38 of 50
FXO
Mode
Description
Command
(config-voiceport)
(config-voiceport)
(config-voiceport)
http://www.studyshorts.co.uk
M Morgan 2010
Page 39 of 50
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
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.
http://www.studyshorts.co.uk
M Morgan 2010
Page 40 of 50
Troubleshooting
From IOSShow interface service-engine 1/0
Service-module service-engine 1/0 status
Show dial-peer voice <tag>
Debug ephone mwi
From CUE
Trace <all/ccn/dns/....>
Show trace buffer
Setup Process
1.
2.
3.
4.
5.
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
http://www.studyshorts.co.uk
M Morgan 2010
Page 41 of 50
Once restored the unit will reboot and show the promptDo 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 .....
** 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:
http://www.studyshorts.co.uk
M Morgan 2010
Page 42 of 50
*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
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
http://www.studyshorts.co.uk
M Morgan 2010
Page 43 of 50
http://www.studyshorts.co.uk
M Morgan 2010
Page 44 of 50
Initialisation Wizard
The Web username and password allows the CUE Module to get the current dn config from CME and
administer it.
http://www.studyshorts.co.uk
M Morgan 2010
Page 45 of 50
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.
http://www.studyshorts.co.uk
M Morgan 2010
Page 46 of 50
http://www.studyshorts.co.uk
M Morgan 2010
Page 47 of 50
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
8
2
CE520-24TT
CE520-24LC
CE520-24PC CE520G-24T
24
20
4
2
24
2
24 + 2
M Morgan 2010
Page 48 of 50
CCA Communities
CCA can discover devices using three methods FQDN
IP Address
Subnet search
http://www.studyshorts.co.uk
M Morgan 2010
Page 49 of 50
Additional Resources
The Techexams Forumshttp://www.techexams.net/forums/ccna-voice/
http://www.studyshorts.co.uk
M Morgan 2010
Page 50 of 50