Documente Academic
Documente Profesional
Documente Cultură
Implementing
Cisco
IOS
Unified
Communications
(640-‐460
IIUC)
CiscoVoiceGuru.com
-‐
Guru
Guide
Prepared
by
Matthew
Berry,
CCIE
#26721
For
more
Cisco
Voice
study
resources
visit
www.ciscovoiceguru.com
Prepared
by
Matthew
Berry,
CCIE
#26721
1
For
more
Cisco
Voice
study
resources
visit
www.ciscovoiceguru.com
Signaling
System
7
(SS7)
–
Voice
signaling
protocol
used
around
the
world
is
which
is
an
out-‐of-‐band
Exam
Topics
Covered
in
Part
I:
(CCS-‐style)
signaling
method
that
communicates
call
setup,
routing,
billing,
and
informational
messages
between
telephone
company
COs
around
the
world.
Describe
the
components
of
the
Cisco
Unified
Communications
Architecture
1. When
user
places
a
call,
first
CO
to
receive
call
performs
an
SS7
lookup
to
locate
number
Describe
the
function
of
the
infrastructure
in
a
UC
environment
2. When
destination
found,
SS7
is
responsible
for
routing
the
call
through
the
voice
Describe
the
function
of
endpoints
in
a
UC
environment
network
to
the
destination
and
providing
all
informational
signaling
to
the
calling
device
Describe
the
function
of
the
call
processing
agent
in
a
UC
environment
Describe
the
function
of
messaging
in
a
UC
environment
PSTN
Numbering
Plan
Describe
the
function
of
auto
attendants
and
IVRs
in
a
UC
environment
E.164
is
an
intl.
numbering
plan
created
by
the
Intl.
Telecommunication
Union
(ITU)
Describe
the
function
of
contact
center
in
a
UC
environment
• Each
number
contains
(1)
country
code,
(2)
national
dest.
code,
(3)
subscriber
code
Describe
the
applications
available
in
the
UC
environment,
including
Mobility,
Presence,
and
Telepresence
Perspectives
on
Voice
After
Convergence
(c.
2)
Describe
how
the
Unified
Communications
components
work
together
to
create
the
Cisco
Unified
Benefits
of
VoIP
Communications
Architecture
• Reduced
communication
costs
–
calls
can
traverse
WAN
Describe
PSTN
components
and
technologies
• Reduced
cabling
costs
Describe
the
services
provided
by
the
PSTN
• Seamless
voice
networks
Describe
time
division
and
statistical
multiplexing
• Take
your
phone
with
you
Describe
supervisory,
informational,
and
address
signalling
• IP
soft
phones
Describe
numbering
plans
• Unified
email,
voice
mail,
and
fax
Describe
analog
circuits
• Increased
productivity
–
eliminate
phone
tag
Describe
digital
voice
circuits
• Feature-‐rich
communications
Describe
PBX,
trunk
lines,
key-‐systems,
and
tie
lines
• Open,
compatible
standards
Perspectives
on
Voice
Before
Convergence
(c.
1)
Cisco
VoIP
Infrastructure
[Endpoints,
Applications,
Call
Processing,
Infrastructure]
• Endpoints
–
IP
phone,
wireless
phone,
video
phone,
IM
client
o Entry-‐level
IP
phones
(3911,
7906G,
7911,
7931G)
LOOP
START,
GROUND
START,
SUPERVISORY,
INFORMATIONAL,
ADDRESS
o Business-‐class
IP
phones
(7940G,
7941G,
7941G-‐GE,
7942G,
7945G)
o Touch
screen
IP
phones
(7970G,
7971G-‐GE,
7975G)
Analog
Connections
o Specialty
devices
(7985G
video,
7921G
wireless,
7939G
conference,
ATA)
Loop-‐start
signaling
(Dial
Tone)
–
analog
phone
off-‐hook,
48V
DC
voltage
flows
from
CO
into
phone
• Applications
–
voice
mail,
conference
call,
call
center,
911
services
• Susceptible
to
glare
–
call
comes
in
when
you
go
off-‐hook
to
make
an
outgoing
call
• Avoid
using
ground
start
signaling
–
ground
signal
first
sent
to
CO
to
send
a
dial
tone
Supervisory
signaling
(Ringing)
• On-‐hook
–
connection
between
tip/ring
broken,
no
electrical
signal
passes
• Off-‐hook
–
phone
connects
tip/ring
wires,
completes
circuit
• Ringing
–
CO
sends
alternating
current
at
40
AC
voltage
down
one
of
the
wires,
phone
detects
and
generates
ringing
signal
Informational
signaling
(Info)
• Dial
tone,
busy,
ringback,
congestion,
reorder,
receiver
off-‐hook,
no
such
number,
confirmation
Address
signaling
(Dialing)
• Dual-‐tone
multifrequency
(DTMF)
–
buttons
use
a
pair
of
high/low
electrical
frequencies
• Pulse
–
rotary-‐dial
wheel
of
phone
connects/disconnects
the
local
loop
circuit
as
it
rotates
to
signal
specific
digits
Problem:
Analog
signals
experience
degradation
(signal
fading)
over
long
distances
>>
installed
repeaters
to
regenerate
the
signal
• Call
Processing
–
UCM,
CME,
UC500
• Repeaters
would
amplify
the
analog
signal
plus
the
line
noise
>>
difficult
to
understand
the
signal
Problem:
Too
many
wires
were
needed
to
support
the
number
of
calls
made/received
Digital
Connections
PAM
>>
Quantization
>>
PCM
Converting
Analog
to
Digital
Signals
1. Sample
the
signal
thousands
of
times
per
second
–
pulse-‐amplitude
modulation
(PAM)
2. Quantize
the
signal
–
assigns
a
value
from
the
voltage
range
based
on
the
amplitude
of
each
audio
sample
a. Divides
voltage
range
into
16
total
segments
(0-‐7
positive,
0-‐7
negative)
b. If
samples
move
outside
the
range
of
normal
human
voice,
quantization
• Infrastructure
–
ASA
firewall,
voice
router/gateway,
voice
switch
process
doesn’t
measure
them
as
accurately
>>
quantization
error
3. Encode
quantized
values
into
8-‐bit,
binary
numbers
called
pulse-‐code
modulation
(PCM)
BOSON
AND
CISCO
PRESS
a. 8-‐bit
samples
*
8000
samples
per
second
=
64000
bits
=
64
kbps
• Which
of
the
following
platforms
would
you
use
to
run
CU
CME?
Cisco
1861
Router
4. Optionally
compress
the
sample
to
save
bandwidth
• On
what
technology
are
T1
and
E1
circuits
based?
TDM.
T1
and
E1
are
digital
circuits,
Time-‐division
multiplexing
(TDM)
allows
CO
to
carry
multiple
conversations
simultaneously
over
a
which
rules
out
analog.
PPP
and
ISFN
are
higher-‐level
protocols,
not
electrical
circuit
single,
four-‐wire
path.
Numeric
values
transmitted
in
specific
time
slots
that
separate
them.
technologies.
Ground
start
is
associated
with
analog
FX
connections.
T1
–
United
States,
Canada,
Japan
–
24,
64-‐kbps
channels
each
known
as
digital
signal
0
(DS0)
• What
are
two
examples
of
informational
signaling?
Dial
tone
and
Ringback.
Ringing
is
E1
–
Other
countries
–
30,
64-‐kbps
channels
supervisory
signal
and
hence
the
wrong
answer.
Loop
and
ground
start
are
electrical
Channel
associated
signaling
(CAS)
–
signaling
info
t(x)
using
same
bandwidth
as
the
voice
signal
transmission
stands
on
FX
connections.
DTMF
is
a
type
of
address
signaling.
• Steal
binary
bits
used
to
send
voice
traffic,
voice
quality
drops
slightly
This
leaves
ringback
and
dial
tone
as
the
right
answer.
• Often
called
robbed
bit
signaling
(RBS)
• Which
of
the
following
is
an
example
of
CCS?
SS7.
CAS
is
an
alternative
to
CCS.
Loop
th th
• Steals
the
8
bit
(least
significant
bit)
on
every
6
frame
sent
start,
ground
start,
and
E&M
are
related
to
analog
connections.
E.164
is
the
• T1
will
send
all
24
of
the
smaller
DS0
frames
in
one
big
T1
frame
worldwide
numbering
plan
and
is
unrelated
to
CCS,
which
leaves
SS&
as
the
only
• T1
frame
is
193
bits
=
(8-‐bit
DS0
frame
*
24
DS0s
per
T1)
+
1
framing
bit
example
of
CCS.
Other
examples
would
include
Q.931,
QSIG,
and
T-‐CCS.
• T1
lines
run
at
1.544
Mbps
=
193-‐bit
T1
frame
*
8000
frames/sec
=
1,544,000
bps
• Which
E&M
type
is
not
supported
by
Cisco?
Type
4.
Cisco
does
support
1,
2,
3,
and
5.
• Extended
Super
Frame
(ESF)
sends
groups
of
24
T1
frames
at
a
time
Loop
start
is
a
signaling
standard
for
FX
connections
and
has
nothing
to
do
with
E&M.
• Three
types
of
signaling
used
on
T1/E1
CAS
connections:
Loop
start,
Ground
start,
E&M
Common
channel
signaling
(CCS)
–
signaling
info
t(x)
using
separate,
dedicated
signaling
channel
Two
main
formats
available
on
T1
circuits:
• Often
called
out
of
band
signaling,
leaving
only
23
DS0s
for
voice
t(x)
• D4
or
Super
Frame
(SF):
D4
or
SF
is
the
original
format
and
designates
a
frame
of
8
bits
• Full
signaling
protocol
used
is
Q.931,
which
is
used
for
ISDN
connections
from
each
of
the
24
DS-‐0s
(192
bits)
and
adds
1
bit
for
framing,
for
a
total
of
193
bits.
• Allows
flexibility
with
signaling
messages,
more
bandwidth,
higher
security
Twelve
193-‐bit
time
slices
or
frames
combine
to
create
what
is
called
the
Super
Frame
th th
• T1
lines
use
24
time
slot,
E1
lines
use
17
time
slot
(SF).
AMI
line
coding
is
configured
on
D4-‐type
circuits
• Extended
Super
Frame
(ESF):
ESF
is
the
most
commonly
used
format
today.
This
Understanding
the
PSTN
expands
the
concept
of
the
Super
Frame
from
12
to
24
time
slices
or
frames.
B8ZS
line
Pieces
of
the
PSTN
coding
is
configured
on
ESF
circuits.
• Local
loop
–
link
between
the
customer
and
the
telco
service
provider
• CO
switch
–
provides
services
to
the
devices
on
the
local
loop
(eg.
signaling,
digit
D4/Super
Frame
12,
193-‐bit
frames
Encoding:
AMI
collection,
call
routing,
setup,
and
teardown
ESF
24,
193-‐bit
frames
Encoding:
B8Z6
• Trunk
–
provides
connection
between
switches,
either
CO
or
private
• Private
switch
–
allows
business
to
operate
“mini
PSTN”
where
all
phones
don’t
need
to
connect
to
CO
• Digital
telephone
–
connects
to
a
PBX
system,
converts
audio
into
binary
• Tie
line:
connection
between
CO
switches
or
PBXs
Prepared
by
Matthew
Berry,
CCIE
#26721
2
For
more
Cisco
Voice
study
resources
visit
www.ciscovoiceguru.com
Prepared
by
Matthew
Berry,
CCIE
#26721
3
For
more
Cisco
Voice
study
resources
visit
www.ciscovoiceguru.com
Prepared
by
Matthew
Berry,
CCIE
#26721
4
For
more
Cisco
Voice
study
resources
visit
www.ciscovoiceguru.com
o Best
Practice:
Either
use
ephone-‐dns
not
configured
with
the
dual-‐line
command
or
just
add
the
huntstop
channel
and
no
huntstop
commands
under
each
line
in
Forwarding
Calls
from
the
IP
Phone
the
overlay
group
to
avoid
confusion.
• “CFwdAll”
button
on
the
IP
phone
• (Config-‐telephony)#
restart
all
–
Restarts
all
phones
in
this
CUCME
environment
• Note:
You
can’t
change
ephone-‐dn
mode
(single/dual-‐line)
without
deleting
a
recreating
it.
Forwarding
Calls
from
the
CLI
• You
can
use
the
x
separator
to
configure
an
overflow
line
of
another
button
on
an
ephone
• (Config)#
ephone-‐dn
21
• (Config-‐ephone-‐dn)#
call-‐forward
{all
|
busy
|
max-‐length
|
night-‐service
|
noan
}
o Max-‐length:
max
number
of
digits
allowed
for
CFwdAll
from
IP
phone
§ If
=
0,
disable
button
on
IP
phone
o Night-‐service:
forward
calls
on
activated
night-‐service
This
does
not
configured
button
2
for
ephone-‐dn
1,
it
makes
button
2
an
overflow
for
button
1.
o In
the
US,
phone
rings
for
2
seconds,
followed
by
4
seconds
of
silence
This
gives
user
the
ease
of
use
to
use
a
second
line
from
overlay
set
for
incoming/outgoing
calls.
• (Example)
(Config-‐ephone-‐dn)#
call-‐forward
noan
1599
timeout
25
Troubleshooting
IP
Phone
Registration
Call-‐Forward
Pattern
to
Support
H.450.3
• Phone
displays
“Configuring
IP”
while
it
attempts
to
get
an
IP
address
using
DHCP.
If
this
lasts
• Call-‐forward-‐pattern
{pattern}
1-‐2
minutes,
there
might
be
issues
with
DHCP
services
(or
voice
VLAN
configuration).
o Settings
>>
Network
Configuration
–
See
if
phone
has
received
an
IP
address
from
DHCP
server.
Fast
way
to
diagnose
problem.
o If
there
is
no
IP
address,
check:
§ Voice
VLAN
settings
on
the
interface
§ DHCP
pool
configuration
§ Properly
configured
ip
helper-‐address
(if
necessary)
§ **#
to
unlock
phone
>>
select
Erase
–
Clears
out
old
settings
• Phone
displays
“Configuring
CM
List”
as
it
attempts
to
download
config
file
from
TFTP
server.
If
this
lasts
for
more
than
30
seconds,
there
might
be
issues
with
the
TFTP
server.
o If
there
is
an
IP
address,
check:
§ Ping
IP
address
of
the
phone
from
the
TFTP
server
§ Make
sure
TFTP
server
is
serving
up
the
right
files
• debug
tftp
events
• show
telephony-‐service
tftp-‐bindings
–
To
verify
what
files
the
TFTP
server
is
serving
up
• show
run
|
include
tftp-‐server
§ Make
sure
CUCME-‐to-‐IP
phone
communication
is
up
and
running
• Skinny
Client
Control
Protocol
(SCCP)
• debug
ephone
register
Supporting
Auto-‐Registration
and
Auto-‐Assignment
of
IP
Phones
Configuring
Call-‐Transfer
• Auto-‐registration
is
enabled
by
default
in
CME
4.0
• Use
“Trnsfer”
softkey
on
a
Cisco
IP
phone
nd
• (Config-‐telephony)#
no
auto-‐reg-‐ephone
–
To
disable
autoregistration
• Consult
transfer:
allows
you
to
speak
to
other
party
before
transferring,
req.
2
line
• show
ephone
attempted-‐registrations
–
To
see
devices
failing
registration
• Blind
transfer:
immediately
transfers
the
call
after
you
dial
the
number,
only
single
line
st
• (Config-‐telephony)#
auto
assign
{1
DN}
{Last
DN}
type
{Phone
Model}
SYSTEM-‐WIDE
Additional
IP
Phone
Configuration
Parameters
• (Config)#
telephony-‐service
• Rebooting
IP
Phones
• (Config-‐telephony)#
transfer-‐system
{full-‐blind
|
full-‐consult
|
local-‐consult
}
o Restart
–
Performs
a
warm-‐boot
of
the
phone,
no
DHCP/TFTP
requests,
FASTER
o Full-‐blind/Full-‐consult:
industry-‐standard
H.450.2
method
of
§ For
phone
line
changes,
speed
dial
changes
transferring,
CME
router
completely
drops
call
from
t(x)
phone
and
starts
o Reset
–
Complete
power
cycle
a
new
call
at
the
phone
to
which
the
call
was
t(x)
§ Changes:
DHCP
scope,
date/time,
firmware,
locale,
o Local-‐consult:
Cisco-‐proprietary
t(x)
method,
if
multiple
lines
or
dual-‐line
button
URLs
(services,
messages,
directories),
voicemail
number
configurations
are
available,
otherwise
uses
blind
t(x)
• Phone
Language
Settings
PER-‐LINE
BASIS
o User
locale
–
Affects
language
settings
for
softkeys,
help
messages,
and
buttons
• (Config-‐ephone-‐dn)#
transfer-‐mode
{blind
|
consult}
§ (Config-‐telephony)#
user-‐locale
{US
|
FR
|
JP
|
…
}
o Network
locale
–
Affects
the
locale
of
the
tones
played
by
the
Cisco
IP
phones
TO
ALLOW
OUTSIDE
TRANSFERS
§ (Config-‐telephony)#
network-‐locale
{US
|
FR
|
JP
|
…
}
Allows
transfers
to
5000
numbers
and
local
ten-‐digit
PSTN
numbers
o (Config)#
ephone-‐template
–
Support
different
locales
in
a
single
location
• (Config-‐telephony)#
transfer-‐pattern
5…
§ CUCME
currently
supports
up
to
five
different
locale
settings
for
• (Config-‐telephony)#
transfer-‐pattern
9..........
the
devices.
Configuring
Call-‐Park
Adding
an
ephone-‐dn
designated
for
call-‐park
purposes
SIMPLE
MODE
• (Config)#
ephone-‐dn
50
• (Config-‐ephone-‐dn)#
number
3001
• (Config-‐ephone-‐dn)#
name
Maintenance
• (Config-‐ephone-‐dn)#
park-‐slot
• (Config-‐ephone-‐dn)#
exit
DETAILED
MODE
• (Config)#
ephone-‐dn
50
• (Config-‐ephone-‐dn)#
number
3002
• (Config-‐ephone-‐dn)#
name
Sales
• (Config-‐ephone-‐dn)#
park-‐slot
{reserved-‐for
<dn>
|
timeout
<sec>
}
• (Config-‐ephone-‐dn)#
park-‐slot
timeout
60
limit
10
{recall
|
notify
|
transfer
|
<cr>}
o 60
second
call-‐park
timeout
limit
o 10
cycles
before
the
call
is
disconnected
• Date
and
Time
Format
o Notify
<dn>:
notifies
different
DN
in
addition
to
the
phone
that
originally
o (Config-‐telephony)#
date-‐format
{mm
|
dd
|
yy}
parked
the
call.
o (Config-‐telephony)#
time-‐format
{12
|
24}
o Notify
only
<db>:
notifies
different
DN
only,
and
not
the
phone
that
• System
Message
originally
parked
the
call.
o All
phones
display
“Cisco
Unified
CME”
on
LCD
screen
by
default
o Recall:
after
timeout,
call
goes
back
to
the
phone
that
originally
parked
o (Config-‐telephony)#
system
message
{Text}
the
call.
o No
restart
required
o Transfer
<dn>:
after
the
timeout,
transfer
the
call
to
specified
DN
o Alternative
<dn>:
specify
alternate
transfer
destination
should
the
destination
DN
specified
in
transfer
<dn>
be
on
the
phone.
Configuring
CUCME
Voice
Productivity
Features
(c.
6)
o Retry
<sec>:
sets
the
amount
of
time
before
CME
attempts
to
transfer
a
Configuring
a
Voice
Network
Directory
parked
call
again.
When
you
call
the
number,
your
name
will
show
up
on
the
display
o Need
to
specify
at
least
one
ephone-‐dn
with
the
park-‐slot
command
• (Config)#
ephone-‐dn
1
before
the
Call
Park
button
will
be
available
on
the
IP
phone.
• (Config-‐ephone-‐dn)#
name
John
Smith
• (Config-‐ephone-‐dn)#
exit
RETRIEVING
PARKED
CALL
(Options)
1. Dial
directly
into
call-‐park
slot
Configuring
Manual
Local
Directory
Entries
2. Press
the
PickUp
softkey
and
dial
the
call-‐park
number
• (Config-‐telephony)#
directory
{entry
|
first-‐name-‐first
|
last-‐name-‐last}
3. From
phone
where
call
was
parked,
press
the
PickUp
softkey
followed
by
*
to
pickup
most
recently
parked
call.
• (Config-‐telephony)#
directory
entry
{1-‐100}
{digits}
name
{directory
name}
You
can
add
up
to
100
manual
entries
this
way.
Prepared
by
Matthew
Berry,
CCIE
#26721
5
For
more
Cisco
Voice
study
resources
visit
www.ciscovoiceguru.com
Configuring
Call
Pickup
Configure
timeout,
when
pin
code
must
be
re-‐entered
Allows
you
to
answer
another
ringing
phone
from
your
own
phone
by
pushing
the
PickUp
softkey.
• (Config)#
telephony-‐service
You
can
also
configure
groups
of
lines
so
you
don’t
have
to
know
individual
DNs.
• (Config-‐telephony)#
login
timeout
120
clear
23:00
• (Config)#
ephone-‐dn
1
o Default
timeout:
60
seconds
• (Config-‐ephone-‐dn)#
pickup-‐group
5509
o clear:
absolute
time
when
pin
timeout,
users
can
then
log
back
in
• (Config-‐ephone-‐dn)#
ephone-‐dn
2
o Requires
a
phone
reset
• (Config-‐ephone-‐dn)#
pickup-‐group
5509
• (Config-‐ephone-‐dn)#
ephone-‐dn
3
Configuring
CDRs
and
Call
Accounting
• (Config-‐ephone-‐dn)#
pickup-‐group
5510
Option
One:
Storing
CDR
data
in
buffered
memory
(RAM)
• …
512,000
bytes
of
memory
dedicated
to
logging
functions.
CDRs
retained
for
10,080
minutes
(7
days).
CME
router
keeps
a
maximum
of
700
CDRs
in
memory.
There
are
three
methods
to
answer
other
ringing
phones:
• (Config)#
logging
buffered
512000
• Directed
pickup:
press
the
PickUp
softkey
and
dial
the
ringing
DN
• (Config)#
dial-‐control-‐mib
{max-‐side
|
retain-‐timer}
• Local
group
pickup:
press
the
GPickUp
softkey
and
enter
an
asterisk
(*)
o (Config)#
dial-‐control-‐mib
max-‐size
700
• Other
group
pickup:
press
the
GPickUp
softkey
and
enter
other
group
number
o (Config)#
dial-‐control-‐mib
retain-‐timer
10080
• (Verify)
show
logging
Additional
Notes:
• If
there
are
multiple
phones
ringing,
it
will
present
you
with
the
oldest
call
Option
Two:
Storing
CDR
data
to
a
Syslog
Server
(e.g.
Kiwi
Syslog
Server)
• If
there
is
only
one
group
configured
in
CME,
you
don’t
need
to
dial
a
group
number
• (Config)#
gw-‐accounting
syslog
• no
service
directed-‐pickup
disables
directed
pickup,
PickUp
f(x)
as
local
group
pickup
• (Config)#
logging
172.30.100.101
Configuring
Intercom
Call
Accounting:
Users
can
use
the
“Acct”
softkey
on
their
IP
phones
to
enter
an
account
code
during
call-‐out
or
during
the
call.
Must
be
followed
by
#.
Allows
for
easy
filtering
and
accurate
billing.
Works
through
speed-‐dial
and
auto-‐answer.
Configure
DNs
that
cannot
be
dialed
by
other
phones.
intercom
{ephone-‐dn
number}
(barge-‐in
|
no-‐auto-‐answer
|
no-‐mute}
label
{description}
Configuring
Music
on
Hold
(MoH)
• Barge-‐In:
places
any
existing
call
on-‐hold
Can
support
G.729
or
G.711,
but
prefers
G.711
to
avoid
using
transcoding
resources
to
convert
codec
• Auto-‐answer:
causes
phone
to
ring
rather
than
auto-‐answer
• (Config)#
telephony-‐service
• No-‐mute:
two-‐way
conversation
immediately
• (Config-‐telephony)#
moh
{filename.wav
|
filename.au}
•
(Config-‐telephony)#
multicast
moh
239.1.1.55
port
2123
Configuration
Example:
• (Config)#
ephone-‐dn
60
Enabling
the
CME
GUI
• (Config-‐ephone-‐dn)#
number
A100
1. Configure
CME
Router
as
a
Web
Server
• (Config-‐ephone-‐dn)#
intercom
A101
label
“Manager”
• (Config)#
ip
http
server
• (Config-‐ephone-‐dn)#
exit
• (Config)#
ip
http
secure-‐server
• (Config)#
ephone-‐dn
61
•
(Config)#
ip
http
path
flash:/gui
• (Config-‐ephone-‐dn)#
number
A101
• (Config)#
ip
http
authentication
local
• (Config-‐ephone-‐dn)#
intercom
A100
label
“Assistant”
2. Create
CME
Web
Administrator
with
Permissions
• (Config-‐ephone-‐dn)#
exit
• (Config)#
telephony-‐service
• (Config)#
ephone
1
• (Config-‐telephony)#
web
admin
system
name
NinjaAdmin
secret
0
cisco
• (Config-‐ephone)#
button
2:60
• (Config-‐telephony)#
web
admin
customer
name
CustomerAdmin
secret
0
cisco
• (Config-‐ephone)#
restart
• (Config-‐telephony)#
dn-‐webedit
• (Config-‐ephone)#
exit
• (Config-‐telephony)#
time-‐webedit
• (Config)#
ephone
2
o If
using
NTP,
do
not
allow
time-‐webedit.
This
will
give
the
user
the
• (Config-‐ephone)#
button
2:61
ability
to
change
the
clock
on
the
CME
router.
• (Config-‐ephone)#
restart
3. Access
CME
GUI
at
http://CME_IP_Address/ccme.html
Configuring
Paging
BOSON
AND
CISCO
PRESS
Similar
to
intercom,
but
is
only
one-‐way
communication.
Only
one
paging
group
per
IP
phone,
but
you
• Licenses
required
to
operate
CUCME
router:
IOS,
Phone
user,
and
Feature
licenses
can
define
paging
numbers
that
page
multiple
paging
groups,
thus
allowing
company-‐wide
functions
• CUCME
creates
configuration
files
for
Cisco
IP
phones
(1)
automatically
as
configuration
• Unicast:
CME
streams
individual
audio
streams
for
up
to
ten
IP
phones
information
is
entered,
(2)
or
by
entering
the
create
cnf-‐files
command
o (Config)#
ephone-‐dn
81
• Benefit
of
CUCM
Biz
Edition
over
UCM
is
integrated
voicemail
support
o (Config-‐ephone-‐dn)#
number
5555
• Phones
that
can
support
three
lines
and
custom
XML
apps
are
Cisco
7960G,
7971G
o (Config-‐ephone-‐dn)#
paging
• When
a
phone
is
plugged
in,
by
default
CME
will
register
the
phone
but
not
create
an
o (Config-‐ephone-‐dn)#
exit
ephone
entry
in
the
running
config
o (Config)#
ephone
1
• How
does
CME
support
shared-‐line
environments
that
allow
multiple
phones
to
receive
o (Config-‐ephone)#
paging-‐dn
80
calls
on
the
same
DN
simultaneous?
Using
overlay
configurations
o
(Config-‐ephone)#
exit
• Feature
license
grants
CME
router
to
support
a
set
number
of
IP
phones
o (Config)#
ephone
2
• Maximum
number
of
paging
groups
to
which
a
Cisco
IP
phone
can
belong:
One
o (Config-‐ephone)#
paging-‐dn
80
• CME
GUI,
by
default,
doesn’t
allow
you
to
modify
(1)
time
on
router,
(2)
ephone-‐dn
• Multicast:
CME
sends
one
audio
stream,
limitless
number
of
phones,
network
must
be
configured
to
support
multicast,
which
can
be
involved
process.
o (Config)#
ephone-‐dn
82
o (Config-‐ephone-‐dn)#
number
6666
o (Config-‐ephone-‐dn)#
paging
ip
239.1.1.100
port
2000
o Continue
configuring
ephones
with
paging-‐dn
• Using
Paging
Groups
o (Config-‐ephone-‐dn)#
paging
group
{PG1,
PG2,
…,
PG10}
Configuring
After-‐Hours
Call-‐Blocking
AFTER-‐HOURS
>>
BLOCKING
PATTERNS
>>
EXEMPTIONS
>>
PIN
CODES
>>
PIN
TIMEOUTS
1. DEFINE
AFTER-‐HOURS
• (Config)#
telephone
service
• (Config-‐telephony)#
after-‐hours
{block
|
date
|
day}
o
(Config-‐telephony)#
after-‐hours
day
mon
17:00
08:00
o (Config-‐telephony)#
after-‐hours
date
jan
1
00:00
00:00
2. SPECIFY
BLOCKING
PATTERNS
• (Config)#
telephone
service
• (Config-‐telephony)#
after-‐hours
block
pattern
{1-‐32}
91..........
• (Config-‐telephony)#
after-‐hours
block
pattern
{1-‐32}
9011T
• (Config-‐telephony)#
after-‐hours
block
pattern
{1-‐32}
91900.......
7-‐24
o 7-‐24
blocks
this
pattern
at
all
times,
24
hours,
7
days
a
week
o No
exceptions
when
using
this
keyword
3. CREATE
EXEMPTIONS
Configure
exemptions
without
pin
code
feature
• (Config)#
ephone
1
• (Config-‐ephone)#
after-‐hour
exempt
• (Config-‐ephone)#
exit
Configure
exemptions
with
the
pin
code
feature
• (Config)#
ephone
2
• (Config-‐ephone)#
ping
{4-‐8
digit
pin}
• (Config-‐ephone)#
exit
Prepared
by
Matthew
Berry,
CCIE
#26721
6
For
more
Cisco
Voice
study
resources
visit
www.ciscovoiceguru.com
Prepared
by
Matthew
Berry,
CCIE
#26721
7
For
more
Cisco
Voice
study
resources
visit
www.ciscovoiceguru.com
Prepared
by
Matthew
Berry,
CCIE
#26721
8
For
more
Cisco
Voice
study
resources
visit
www.ciscovoiceguru.com
Trunking
CME
to
other
VoIP
Systems
Configuring
Gateways
and
Trunks
(c.
8)
Common
VoIP
Signaling
Protocols:
• H.323:
first
of
four
voice
signaling
protocols,
created
by
ITU-‐T
to
allow
simultaneous
voice,
video,
and
data
to
transmit
across
ISDN
connections
Configuring
Analog
Voice
Ports
o Peer-‐to-‐peer
protocol
• FXS
–
Connect
to
end
stations
o Every
device
is
completely
independent
• show
voice
port
summary
o Requires
a
lot
of
configuration
• If
using
router
for
CME,
each
ephone-‐dn
configured
shows
up
under
output
as
EXFS
port
o Each
device
needs
full
knowledge
of
the
network.
For
2XXX,
go
to
H.323
Gateway
2.
For
3XXX,
go
to
H.3232
Gateway
3.
Three
common
areas
of
configuration:
§ H.323
Gatekeeper
-‐
Avoid
massive
configuration
1. Signaling
• “Centralized
phonebook”
to
H.323
network
a. (Config)#
voice-‐port
0/0/0
• Provide
call
admission
control
(CAC)
b. (Config-‐voiceport)#
signal
{groundStart
|
loopStart}
• Bandwidth
management.
o Cisco
IP
phones
not
support
H.323
due
to
severe
resource
consumption
2. Call
progress
tones
a. (Config)#
voice-‐port
0/0/0
b. (Config-‐voiceport)#
cptone
{US
|
UK
|
…}
3. Caller
ID
information
a. (Config)#
voice-‐port
0/0/0
• Session
Initiation
Protocol
(SIP):
next
generation
of
H.323,
created
IETF,
more
b. (Config-‐voiceport)#
station-‐id
name
3rd
Floor
Fax
lightweight
and
scalable
than
H.323,
poised
to
become
primary
VoIP
signaling
standard
c. (Config-‐voiceport)#
station-‐id
number
5551000
o All
SIP
does
is
start,
manage,
and
terminate
the
session;
it
passes
off
responsibility
to
other
protocols
FXO
–
Act
as
trunks
to
connect
to
the
PSTN
CO
or
PBX
systems
o Designed
by
the
IETF
as
an
alternative
to
H.323
Uses
many
of
the
same
commands
as
the
FXS
ports
do
1. Dual-‐tone
multifrequency
(DTMF)
or
pulse
a. (Config)#
voice-‐port
0/0/0
b. (Config-‐voiceport)#
dial-‐type
{dtmf
|
pulse}
2. Rings
before
router
answers
cal
a. (Config)#
voice-‐port
0/0/0
b. (Config-‐voiceport)#
ring
number
<number>
(def.
1
ring)
Configuring
Digital
Voice
Ports
Configuring
T1
CAS
PSTN
Interface
• show
controllers
t1
• (Config)#
controller
t1
1/0
• (Config-‐controller)#
framing
{esf
|
sf}
• (Config-‐controller)#
linecode
{ami
|
b8zs}
• Media
Gateway
Control
Protocol
(MGCP):
first
true
client/server
VoIP
signaling
• (Config-‐controller)#
clock
source
{free-‐running
|
internal
|
line}
protocol,
vast
majority
of
gateway
configuration
is
from
a
centralized
system
called
a
• (Config-‐controller)#
ds0-‐group
{1-‐23}
timeslots
{1-‐24}
type
{<cr>
|
…
|
fxo-‐loop-‐start}
call
agent.
Not
as
widely
supported
as
H.323
or
SIP.
o ds0-‐group
command
is
useful
for
fractional
T1s
o Allows
you
to
put
gateways
under
the
control
of
a
centralized
call
agent
• show
voice
port
summary
o Turns
gateway
into
a
dumb
terminal
Many
US
providers
use
ESF
and
B8ZS
o Commands
sent
from
call
agent
to
gateway
using
UDP
port
2427
Configuring
T1
CCS
PSTN
Interface
• (Config)#
isdn
switch-‐type
{primary-‐5ess
|
…}
o Must
match
switch
of
service
provider
• (Config)#
controller
t1
1/0
• (Config-‐controller)#
pri-‐group
timeslots
1-‐24
{nfas_d
|
service
|
<cr>}
o (Config-‐controller)#
pri-‐group
{nfas_d
|
service
|
timeslots
|
<cr>}
• show
voice
port
summary
• Skinny
Client
Control
Protocol
(SCCP):
only
Cisco-‐proprietary
VoIP
protocol
currently
in
use,
supported
by
limited
number
of
Cisco
gateways,
provides
signaling
protocol
between
CUCM
and
Cisco
IP
phones.
Reports
every
action
to
CM
server,
which
then
Understanding
and
Configuring
Dial
Peers
responds
with
the
action
the
device
should
take.
• Think
of
dial
peers
as
static
routes
for
your
voice
network
o Cisco-‐proprietary
protocol
• Dial
peers
define
voice
reachability
information
Understanding
Internet
Telephony
Service
Providers
(ITSP)
• ITSP
providers
allow
you
to
connect
to
the
PSTN
using
VoIP
communications
;
Prepared
by
Matthew
Berry,
CCIE
#26721
9
For
more
Cisco
Voice
study
resources
visit
www.ciscovoiceguru.com
Prepared
by
Matthew
Berry,
CCIE
#26721
10
For
more
Cisco
Voice
study
resources
visit
www.ciscovoiceguru.com
Practical
Scenario
1:
PSTN
Failover
Using
the
“prefix”
Command
Quality
of
Service
(config)#
dial-‐peer
voice
10
voip
Three
enemies
of
VoIP
traffic:
(config-‐dial-‐peer)#
destination-‐pattern
6…
• Lack
of
bandwidth:
multiple
streams
of
voice
and
data
traffic
are
competing
for
a
(config-‐dial-‐peer)#
session
target
ipv4:10.1.1.2
limited
amount
of
bandwidth.
(config-‐dial-‐peer)#
preference
0
• Delay:
time
it
takes
a
packet
to
move
from
the
original
starting
point
to
the
final
(config-‐dial-‐peer)#
exit
destination;
delay
comes
in
three
forms:
(config)#
dial-‐peer
voice
11
pots
o Fixed
delay:
cannot
be
changed,
such
as
geographical
distances
(config-‐dial-‐peer)#
destination-‐pattern
6…
o Variable
delay:
can
change,
such
as
queuing
delay
(config-‐dial-‐peer)#
port
1/0:1
o Jitter
(delay
variations):
packets
with
different
amounts
of
delay
(config-‐dial-‐peer)#
preference
1
• Packet
loss:
due
to
congested/unreliable
network
connections
(config-‐dial-‐peer)#
no
digit-‐strip
(config-‐dial-‐peer)#
prefix
1512555
QoS
is
designed
to
keep
voice
traffic
running
smoothly
during
temporary
moments
of
congestion.
(config-‐dial-‐peer)#
exit
You
need
to
have
QoS
in
some
form
at
any
point
of
the
network
where
congestion
exists.
If
multiple
dial
peers
have
exactly
equal
patterns
and
preferences,
router
chooses
one
randomly
Network
Requirements
for
Voice
and
Video
• End-‐to-‐end
delay:
150
ms
or
less
Practical
Scenario
2:
Directing
Operator
Calls
to
the
Receptionist
• Jitter:
30
ms
or
less
(config)#
voice-‐port
1/0/1
• Packet
loss:
1%
or
less
(config-‐voiceport)#
connection
plar
0
(config-‐voiceport)#
exit
Network
Requirements
for
Data
(config)#
num-‐exp
0
5000
When
designing
QoS,
divide
applications
into
no
more
than
4-‐5
categories
• Mission-‐critical
applications:
require
dedicated
bandwidth
Because
this
is
a
“universal”
transformation,
you
can
• Transactional
applications:
database
applications
accomplish
this
objective
using
the
num-‐exp
global
• Best-‐effort
applications:
web
browser,
e-‐mail,
FTP
file
transfers
configuration
command.
• Scavenger
applications:
peer-‐to-‐peer
file-‐sharing
The
router
applies
the
num-‐exp
command
the
instant
it
Cisco
AutoQoS
receives
a
dialed
number,
even
before
it
attempts
to
Offers
multiple
advantages
to
manual
QoS
configuration
match
an
inbound
dial
peer.
• Reduces
the
time
of
deployment
• Provides
configuration
consistency
Practical
Scenario
3:
Specific
POTS
Lines
for
Emergency
Calls
• Reduces
deployment
cost
(config)#
dial-‐peer
voice
10
pots
• Allows
manual
tuning
(config-‐dial-‐peer)#
destination-‐pattern
911
(config-‐dial-‐peer)#
port
1/0/0
Cisco
AutoQoS
Implementation
(config-‐dial-‐peer)#
no
digit-‐strip
• It
is
ideal
to
have
Cisco
IP
phones
mark
traffic.
Always
best
to
apply
QoS
in
as
many
(config-‐dial-‐peer)#
exit
places
as
possible
where
there
is
a
potential
bottle-‐neck.
(config)#
dial-‐peer
voice
11
pots
• AutoQoS
uses
CDP
to
detect
Cisco
IP
phones
on
Cisco
switches
and
properly
configure
(config-‐dial-‐peer)#
destination-‐pattern
9911
the
QoS
settings.
This
ensures
that
a
user
cannot
disconnect
their
IP
phone
and
attach
(config-‐dial-‐peer)#
port
1/0/0
another
device
to
receive
high-‐priority
network
treatment.
Don’t
disable
CDP
on
(config-‐dial-‐peer)#
forward-‐digits
3
switches
supporting
Cisco
IP
phones.
(config-‐dial-‐peer)#
exit
• AutoQoS
uses
sophisticated
queuing
method
known
as
Low
Latency
Queuing
(LLQ).
• Using
AutoQoS
features
with
incorrectly
configured
bandwidth
commands
may
cause
Practical
Scenario
4:
Using
Translation
Profiles
substandard
network
service.
Requires
a
three
step
process:
auto
qos
voip
{<cr>
|
trust
|
cisco-‐phone
|
cisco-‐softphone}
CREATE
RULE
(Config)#
voice
translation-‐rule
[rule
number]
cisco-‐phone
and
cisco-‐softphone
are
conditional
trust
boundaries
that
are
only
formed
if
that
device
is
(Cfg-‐translation-‐rule)#
rule
1
/match/
/set/
detected
using
CDP,
otherwise
any
QOS
markings
are
discarded
and
remarked.
(Verify)
test
voice
translation-‐rule
[rule
number]
[dialed
number]
Cisco
AutoQoS
Configuration
on
Access
Layer
Switchports
ASSIGN
TO
PROFILE
Will
only
enable
a
trust
boundary
if
CDP
detects
a
Cisco
IP
phone
or
Cisco
IP
Communicator
(Config)#
voice
translation-‐profile
[profile
name]
• (Config)#
interface
fa0/3
(cfg-‐translation-‐profile)#
translate
{called
|
calling
|
redirect-‐called
|
redirect-‐target}
{rule
#}
• (Config-‐if)#
auto
qos
voip
{cisco-‐phone
|
cisco-‐softphone
|
trust}
• (Verify)
show
run
interface
FastEthernet
0/3
ASSIGN
TO
DIAL-‐PEER
(Config)#
dial-‐peer
voice
100
pots
Cisco
AutoQoS
Configuration
on
Switch-‐Router
Uplink
(config-‐dial-‐peer)#
translation-‐profile
{incoming
|
outgoing}
[profile
name]
•
(Config)#
interface
fa0/1
• (Config-‐if)#
auto
qos
voip
trust
• (Verify)
show
run
interface
FastEthernet
0/1
NEVER
DO
VERY
POOR
FRAMING
Cisco
AutoQoS
Configuration
on
Router
Interfaces
•
(Config)#
interface
fa0/1
• (Config-‐if)#
auto
qos
voip
trust
• (Verify)
show
run
interface
FastEthernet
0/1
Prepared
by
Matthew
Berry,
CCIE
#26721
11
For
more
Cisco
Voice
study
resources
visit
www.ciscovoiceguru.com
• Tutorial
o Recorded/spoken
name
o Record
standard
greeting
o Change
password
o Note:
initial
access
and
first-‐time
setup
only
done
from
primary
extension
• Greetings
o Standard
greeting
o Alternate
greeting
–
used
for
emergencies/holidays,
disabled
by
default
• Message
Management
and
Playback
o Play
and
restart
o Fast
forward
and
remind
–
3-‐second
intervals
o Skip
o Save
o Reply
to
original
sender
o Forward
to
another
subscriber
o Delete
o Undelete
• Message
Types
• Message
Waiting
Indicator
o In
order
for
the
MWI
light
to
display
for
GDMs,
each
group
member
must
have
that
extension
assigned
to
their
phone.
• Message
Notifications
o Can
place
phone
call
or
send
email
as
notification
• Message
Notifications:
System
Level
o By
default,
message
notification
is
disabled
both
globally
and
per-‐user
o Must
first
enable
and
configure
system-‐wide
and
then
per-‐user
§ Enable
message
notifications
for
urgent
or
all
messages
§ If
subscriber
notified
via
phone
call,
can
retrieve
message
during
the
notification
call
§ Attach
messages
to
outgoing
e-‐mail
notifications
§ Enable
cascading
notifications
§ Set
the
ring
no
answer
(RNA)
timeout,
in
seconds
§ User
a
restriction
table
• Message
Notifications:
User
Level
o Can
add
extra
digits
after
dialed
number,
asterisk
(*)
inserts
1
sec
pause
• Live
Reply
–
reply
to
caller
while
listening
to
voicemail
(using
ANI)
• Live
Record
–
record
a
phone
conversation,
save
to
mailbox
• Distribution
Lists
Cisco
Unity
Express
o Public
Distribution
Lists
–
anyone
can
leave
a
message
Determined
by
three
factors:
o Private
Distribution
Lists
–
only
creator
can
leave
a
message
• Mailbox
Storage
• IP
PBX
type
that
connects
to
Cisco
Unity
Express
o If
user’s
mailbox
is
90%+
full
>>
prompted
to
delete
messages
• Cisco
Unity
Express
module
type
the
license
file
will
be
loaded
on
o If
user’s
mailbox
is
100%
>>
caller
cannot
leave
message
• Number
of
mailboxes
required
for
the
site
that
the
CUE
will
be
serving
Mailbox
Caller
Features
Features
and
Functions
of
CUE
Voice
Mail
• Record
Message
Options
Users
and
Groups
Voice
mail
message
must
be
at
least
2
seconds
long
to
be
valid
and
retained
• User/Subscriber
Accounts
–
dictates
extension
number
of
mailbox,
message
storage
o Review
the
recorded
message
options,
message
retrieval/retention,
permissions
o Re-‐record
the
message
• Groups
–
collection
of
subscribers
(same
function
as
user/subscriber)
o Set
normal
or
urgent
priority
on
the
message
o Administrators
–
permission
to
configure
all
aspects
of
Unity
o Cancel
message
o Broadcast
–
permission
to
send
a
broadcast
message
to
all
subscribers
• Operator
Assistance
o Default
system
operator
–
site-‐wide,
typically
used
for
operator,
used
if
specific
operator
not
configured
o Specific
operator
–
defined
per
mailbox
• Mailbox
Login
–
subscriber
can
listen
to
voicemail
from
outside
the
network
Prepared
by
Matthew
Berry,
CCIE
#26721
12
For
more
Cisco
Voice
study
resources
visit
www.ciscovoiceguru.com
Subscriber
Management
VoiceView
Express
• Accessed
via
GUI
using
CUE
password
(not
PIN)
-‐
http://CUE-‐Server
• Operates
as
an
IP
phone
service,
which
is
XML-‐based
• Configuration
options:
• With
CUE
3.X+,
VoiceView
Express
is
available
for
3
simultaneous
sessions
o Change
password
and
PIN
o Configure
zero-‐out
extension
Integrated
Messaging
o Select
language
• Provides
access
to
voicemail
messages
via
an
email
client
and
allows
a
subscriber
to
o Enable
alternate
greeting
treat
voicemail
messages
similarly
to
email
messages.
o View
public
distribution
lists
• Integrated
-‐
Adds
Unity
mailbox
as
an
additional
IMAP
account
o Configure
private
distribution
lists
• Unified
–
Combined
into
one
email
account
o Configure
message
notification
o Search
local
directory
o Obtain
help
Voice
Profile
for
Internet
Mail
(VPIM)
• Allows
voicemail
systems
to
exchange
messages
with
other
voicemail
systems
• Forwards
voicemails
as
email
messages
using
SMTP,
w/o
need
for
real-‐time
connection
Telephony
User
Interface
Subscriber
Management
Features
and
Functions
of
CUE
Auto
Attendant
CUE
Automated
Attendant
(AA)
Enables
business
to
answer
and
direct
incoming
phone
calls
without
human
intervention
• Default
Auto
Attendant
Scripts
o Auto
Attendant
script
(default,
used
by
CUE)
o Auto
Attendant
simple
script
• AA
Greetings/Prompts
o Welcome
Prompt,
Business
Open
Prompt,
Business
Close
Prompt,
Holiday
Prompt
• Business
Hours
Settings
–
Broken-‐down
into
30
minute
time
slots
• Holiday
Settings
–
Define
full-‐day
increments
that
holiday
message
plays
• AA
Operator
–
Dialing
0
directs
to
a
live
operator,
not
available
in
AA
Simple
script
• Dial
by
Name
and
Dial
by
Extension
–
Last
name,
First
name
only
in
AA
Simple
script
• Administration
via
Telephone
System
(AVT)
–
record
greetings
using
the
phone
CUE
Custom
Scripting
• Looks/Works
similar
to
the
CRS
Script
Editor
CUE
Management
Administrator
Management
• Configuring
using
CLI
•
Configuring
using
GUI
-‐
http://CUE-‐Server
User
and
Group
Administration
• Reset
password
and
PINs
• Configure
name
information
• Associate
a
phone
• Assign
a
primary
extension
• Assign
an
E.164
address
• Assign
a
language
• Enable/configure
message
notification
• Configure
mailbox
settings
• Configure
group
membership
Voice
Mail
Administration
• Mailbox
size
limitations
• Message
size
limitations
• MWI
configuration
• Integration
messaging
configuration
• VoiceView
Express
configuration
Auto
Attendant
Administration
• Call-‐in
number
• Script
choice
• Language
• Prompt
choices
• Dial
by
first
or
last
name
• Operator
extension
• Schedule
choice
Backup
and
Restore
• Options
for
backup
o Configuration
o Data
o Historical
reporting
• During
backup/restore,
CUE
is
taken
offline
and
all
user
sessions
are
terminated
• CUE
must
be
started
after
the
process
is
finished
Reports
• Voice
mail
• Mailboxes
• Backup
history
• Restore
history
• Network
Time
Protocol
• Call
history
• Real-‐time
reports
CME
Configuration
–
most
CME
CLI
features
can
be
configured
from
the
CUE
GUI
Synchronizing
and
Saving
Information
-‐
option
to
sync
CME
and
CUE
settings
Prepared
by
Matthew
Berry,
CCIE
#26721
13
For
more
Cisco
Voice
study
resources
visit
www.ciscovoiceguru.com
Record
Message
Options
for
the
Caller
Cisco
Unity
Express
Configuration
(c.
10)
CUE
Installation
and
Upgrade
Adding
Cisco
Unity
Express
Licenses
Installing
the
CUE
Module
• After
CUE
Initialization
Wizard
has
been
completed,
the
license
cannot
be
changed
between
CUCM
and
CUCME
support
without
a
fresh
install
of
CUE
software.
If
you
install
a
CUE
module,
it
shows
up
as
interface
Service-‐Engine1/0
• Licensing
provides:
• (Config)#
service-‐module
ip
address
10.100.1.10
255.255.255.0
o Support
for
CUCM,
CUCME
• (Config)#
service-‐module
ip
default-‐gateway
10.100.1.1
o Number
of
general
delivery
mailboxes
(GDM)
supported
o Number
of
subscriber
mailboxes
supported
Since,
the
service-‐module
is
the
internal
interface
of
Cisco
Unity
Express,
endpoint
traffic
destined
for
o Number
of
IVR
ports
supported
CUE
cannot
access
this
interface
directly
without
passing
through
the
external
interface,
the
service
engine.
We
need
a
static
IP
route
to
redirect
all
traffic
destined
for
the
service-‐module
to
the
service-‐ Verifying
Cisco
Unity
Express
Licenses
engine.
• (Config)#
ip
route
10.100.1.10
255.255.255.255
Service-‐Engine1/0
To
connect
to
the
CLIE
of
CUE
module:
• service-‐module
service-‐Engine
1/0
session
Installing
and
Upgrading
the
CUE
CUE
Configuration
CUE
Post-‐Installation
Configuration
System
will
run
the
service-‐engine
post-‐installation
configuration
tool
• Hostname
• Domain
name
• Primary/Secondary
DNS
servers
• Primary/Secondary
NTP
servers
• Time
zone
• Administrative
credentials
Configuring
CUCME
to
Support
CUE
• HTTP
Server
Options
o (Config)#
ip
http
server
o (Config)#
ip
http
path
flash:
o (Config)#
ip
http
authentication
aaa
• Dial
Peers
for
CUE
o (Config)#
dial-‐peer
voice
7000
voip
o (Config-‐dial-‐peer)#
description
VoiceMail
o (Config-‐dial-‐peer)#
destination-‐pattern
7000
o (Config-‐dial-‐peer)#
session
protocol
sipv2
§ CUE
uses
SIP
for
call
control
o (Config-‐dial-‐peer)#
session
target
ipv4:10.100.1.10
§ Needs
location
to
route
calls
o (Config-‐dial-‐peer)#
dtmf-‐relay
sip-‐notify
§ Since
VoIP
is
digital,
need
to
tell
CUCME
to
relay
the
DTMF
tones
in
a
format
CUE
can
interpret
o (Config-‐dial-‐peer)#
codec
g711ulaw
§ CUE
only
supports
G.711
codec
Prepared
by
Matthew
Berry,
CCIE
#26721
14
For
more
Cisco
Voice
study
resources
visit
www.ciscovoiceguru.com
o debug
ccsip
calls
Prepared
by
Matthew
Berry,
CCIE
#26721
15
For
more
Cisco
Voice
study
resources
visit
www.ciscovoiceguru.com
• UC520
does
not
support
dynamic
routing
protocols,
SRST,
Communications
Manager
integration,
or
WAN
interface
cards
for
remote
office
data
connectivity.
• Not
field
upgradable
• All
520
product
SKUs
have
the
following:
o Console
port
o 3.5mm
MoH
audio
jack
o Integrated
eight-‐port
10/100BASE-‐TX
PoE
switch
o Four
FXS
ports
o 10/100
BASE-‐TX
WAN
port
o 10/100
BASE-‐TX
LAN
port
Cisco
Mobility
Express
System
Cisco
521
Wireless
Express
802.11b/g
AP
operates
in
one
of
two
modes:
• Standalone
mode
(mode
one)
–
CCA
used
to
managed
up
to
three
independent
Cisco
521
APs
• Controller-‐based
mode
(mode
two)
–
Used
in
cases
where
four
or
more
wireless
APs
are
required
or
where
this
mode
is
a
better
fit
for
a
particular
implementation.
For
this,
CCA
requires
the
introduction
of
the
Cisco
526
Wireless
Express
Mobility
Controller.
o Each
Cisco
526
Mobility
Controller
can
manage
up
to
six
APs,
and
a
maximum
of
two
Controllers
can
be
added
to
a
single
SBCS
system.
Prepared
by
Matthew
Berry,
CCIE
#26721
16
For
more
Cisco
Voice
study
resources
visit
www.ciscovoiceguru.com
1. Fill
out
the
AA
&
Voicemail
tab
that
comes
up
automatically
a. Auto
Attendant
Extension,
enter
6001
b. Auto
Attendant
PSTN
Number,
enter
AA’s
full
E.164
number
–
in
this
case
4445556001
c. Select
drop-‐down
list
box
to
select
AA
script
d. Voicemail
Access
Extension
field,
enter
6000.
This
becomes
the
voicemail
pilot
number
for
the
phone
system.
e. Voicemail
Access
PSTN
Number
field,
enter
the
full
E.164
number
of
the
voicemail
pilot
–
in
this
case
4445556000
2. Click
the
Device
tab
a. Voice
System
Type
must
be
either
PBX
or
Key
System
3. Click
Configure
as
a
PBX
radio
button
4. Click
Dial
Plan
tab
5. On
Dial
Plan
tab,
enter
4
in
the
Number
of
Digits
Per
Extension
field
6. In
the
Incoming
Call
Handling
Section,
for
the
FXO
Trunks
drop-‐down
list
box,
choose
Auto
Attendant.
This
tells
the
system
what
to
do
with
calls
inbound
from
the
PSTN
a. Operator:
If
chosen,
must
enter
extension
of
the
operator
b. Custom
Configuration:
manually
map
each
FXO
line
to
separate
number
c. Auto
Attendant:
sends
the
call
to
the
number
designated
as
auto
attendant
7. Select
Auto
Attendant
8. Click
the
Voice
Features
tab
a. Configures
MoH,
hunt
groups,
paging,
intercom,
call
park,
conferencing,
etc.
9. Click
on
the
User
tab
10. Change
extensions
from
default
range
(3
digits,
beginning
with
201)
to
four-‐digit
range
11. Complete
the
LastName,
FirstName,
UserID,
and
Password
fields
for
each
phone
12. Click
More
in
the
row
for
the
first
phone.
This
is
where
most
phone
settings
configured.
13. To
add
an
additional
line,
click
the
button
you
want
to
configure
a. In
Type
column,
choose
Normal.
b. In
Extension
column,
enter
a
four-‐digit
DN
–
in
this
case
2005
c. Set
Class
of
Restrictions
(COR)
i. Internal:
access
to
internal
DNs
only
ii. Local:
access
to
internal
DNs
and
local
PSTN
iii. Domestic:
access
to
internal
DNs,
local/LD
PSTN
iv. International:
access
to
all
route
patterns
v. Unrestricted:
no
COR
lists
associated
with
them,
functionally
the
same
as
International
14. Click
OK
in
the
More
Options
dialog
box
>>
Takes
you
back
to
Users
tab
a. All
tabs
should
now
turn
GREEN
15. Before
clicking
Apply,
scroll
down
and
click
the
Network
tab
a. Modify
default
voice
VLAN,
DHCP
scope
16. Click
Apply
Configuring
and
Maintaining
UC500
Series
for
Voice
(c.
12)
SIP
Trunking
on
the
UC520
Preconfigured
Data,
Security,
and
VPN
Templates
• Verify
UC520
system
has
a
valid
DNS
server
and
domain
name
configured
• Voice
and
data
VLAN
o Configure
>>
Device
Properties
>>
IP
Address
>>
Device
Configuration
Tab
• WAN
interface
configured
to
receive
IP
via
DHCP
from
ISP
• Click
the
SIP
Trunk
tab
• Private
addressing
via
DHCP
scope
(192.168.10.0/24
DATA
and
10.1.1.0/24
VOICE)
o Service
Provider
drop-‐down
list,
select
SIP
provider
• IOS
firewall
o Fill
out
the
fields
on
the
SIP
Trunk
tab
as
required
by
provider
• NAT,
best-‐practice
ACL
filtering,
IPsec-‐based
VPN
network
§ Proxy
Server
only
mandatory
field
on
this
tab
• EZ-‐VPNs
between
disparate
UC520
systems
• Click
the
Dial
Plan
tab
o Click
the
Configure
button
next
to
Direct
Inward
Dial
Preconfigured
Basic
Voice
Platform
o Click
add
in
the
one-‐to-‐one
DID
translation
section
to
input
the
DID
• Configured
FXO
ports
mappings
appropriate
for
your
installation.
• Power
failover
(PFO)
feature
delivers
emergency
dial
tone
to
analog
handset
by
o Enter
your
DID
mappings
by
entering
a
description,
DID
number
range,
electrically
connection
FXO
port
to
FXS
port
when
power
is
removed
from
chassis.
and
internal
number
that
should
be
translated.
o Click
in
the
trunk
portion
of
the
row
and
select
SIP
Trunk
Cisco
Configuration
Assistance
o Click
OK
at
the
bottom
of
the
DID
configuration
box
• CCA
is
free
tool
from
Cisco.com
and
used
to
configure/manage/maintain
SBCS
o Click
Apply
at
the
bottom
of
the
Voice
dialog
box
• CCA
Community
is
a
“family”
of
SBCS
suite
devices
Using
CCA
for
System
Maintenance
and
Troubleshooting
Creating
Communities
Management
features
of
CCA
are
found
under
Maintenance
menu
of
the
feature
bar.
• Create
Community
Administrator
can
perform
following
tasks
from
CCA:
• Enter
name
for
device,
click
Discover
>>
Devices
Using
a
Seed
IP
Address
>>
Enter
1. Upgrade
software
within
the
suite
default
IP
address
of
the
UC520,
which
is
192.168.10.1
2. Manage
file
systems
• Default
password
is
cisco/cisco
3. Backup
and
restore
suite
component
configurations
4. Restart
and
reset
devices
Understanding
CCA
Interface
Network
Troubleshooting
with
the
CCA
Tool
• Event
Notification
icon
or
Monitor
>>
Event
Notificiation
• Health
icon
or
Monitor
>>
Health
• Troubleshoot
menu
of
the
left-‐paned
feature
bar
o Links
and
Connectivity
Tool
§ ICMP
pings
§ Port-‐by-‐port
diagnostics
§ Messages
offering
advice
§ e
1. Voice:
primary
button
used
to
configure
UC520
for
voice.
Shortcut
for
Configure
>>
Telephony
>>
Voice
2. SmartPorts:
configure
ports
and
devices
by
assigning
roles
3. Health:
used
for
troubleshooting
4. Event
Notification:
displays
important
network
conditions
5. Front
Panel
View:
shows
devices
managed
in
community
6. Topology
View:
shows
network
map
of
community
Configuring
the
UC520
for
Voice
Clicking
Telephony
icon
in
CCA
toolbar
or
drilling
down
to
Configure
>>
Telephony
>
Voice
Prepared
by
Matthew
Berry,
CCIE
#26721
17
For
more
Cisco
Voice
study
resources
visit
www.ciscovoiceguru.com