640 460 Iiuc

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


Assign ports to VLANs

Exam Topics Covered in Part II: o
§ (Config-‐if)# switchport mode access
§ (Config-‐if)# switchport access vlan 50
Implement Cisco Unified Communications Manager Express to support endpoints using CLI § (Config-‐if)# switchport voice vlan 10
Describe the appropriate software components needed to support endpoints o (Optional) Configure routing between VLANs
Describe the requirements and correct settings for DHCP, NTP, and TFTP 1. Layer 3/multilayer switching – router is internal to the switch, preferred
Configure DHCP, NTP and TFTP method because there is virtually no bandwidth constraints for hosts
Describe the differences between key system and PBX mode assigned to the VLAN to reach the router. Only speed constraint is the
Describe the differences between the different types of ephones and ephone-‐dns backplane speed of the switch.
Configure Cisco Unified Communications Manager Express endpoints 2. Separate router ports per VLAN
Configure call-‐transfer per design specifications 3. Router connected to a trunk port (ROAS)
Configure voice productivity features, including hunt groups, call park, call pickup, paging groups, and • (Config)# interface fa0/0.10
paging/intercom • (config-‐subif)# description ROUTER INT – VOICE VLAN
Configure Music on Hold • (config-‐subif)# encapsulation dot1q 10
• (config-‐subif)# ip address 172.16.1.1 255.255.255.0
Connecting IP Phones to the LAN Infrastructure (c. 3) Forward specific broadcast types to a configured IP address
In this case, forwards DHCP request to VLAN 50 to its DHCP server
Commonly called DHCP Relay
Cisco Inline Power • (config-‐subif)# ip helper-‐address 172.16.2.5
1. Cisco prestandard PoE device physically connects to the switch • (config-‐subif)# exit
2. Switch sends a Fast Link Pulse (FLP) tone signal to the device. Only an unpowered, Cisco • (Config)# interface fa0/0.50
prestandard PoE device will loop this tone back • (config-‐subif)# description ROUTER INT – DATA VLAN
3. When switch receives this tone back, it realizes this device is an unpowered Cisco • (config-‐subif)# encapsulation dot1q
prestandard PoE device and applies a minimal amount of power (6.3W) to the line. • (config-‐subif)# ip address 172.16.2.1 255.255.255.0
4. Unpowered device boots and communicates its actual power requirements to the
switch using CDP, which Cisco devices traditionally use to detect directly attached
Understanding Cisco IP Phone Boot Process
neighbors.
1. Cisco phone connected to switch port and receives either Cisco-‐proprietary PoE or
If CDP is disabled on the switch, the switch allocates the maximum amount of power to each port. The
802.af PoE.
device will only use the power it needs and the rest will be wasted.
2. As phone powers on, switch sends VLAN info to phone using CDP as primary
• Both PoE standards can deliver power from 0 to 15.4W.
mechanism. It now knows what VLAN it should use.
• Oversubscription can take place if the power consumption is more than the switch can
3. Cisco phone sends DHCP request. Router uses ip helper-‐address command to forward
handle. In this scenario, the higher ports will be the first to lose their power.
request directly to the DHCP server
4. DHCP server responds with a DHCP offer. Once accepted, phone receives all the DHCP
802.3af Power over Ethernet options that go along with the DHCP request. Receives information on DHCP Option
• Instead of applying an FLP tone to detect an unpowered PoE device, 802.3af uses a 150, which is for a TFTP server.
small DC current that is constantly applied to the line. 5. Cisco phone contacts TFTP server and downloads configuration file.
• An 802.3af-‐capable device will be equipped with a resistor that returns a specific level of 6. Cisco phone attempts to contact the first call processing server listed in its configuration
resistance to the line. When the switch detects this resistance level, it will know how file to register.
much power to send to the attached device.
Configuring a Router-‐Based DHCP Server
(Config)# ip dhcp excluded-‐address 172.16.1.1 172.16.1.9
(Config)# ip dhcp excluded-‐address 172.16.2.1 172.16.2.9
(Config)# ip dhcp pool DATA_SCOPE
(dhcp-‐config)# network 172.16.2.0 255.255.255.0
(dhcp-‐config)# dns-‐server 4.2.2.2
(dhcp-‐config)# exit
o Class 0 group is a catch-‐all simply asking for power (Config)# ip dhcp pool VOICE_SCOPE
o Too many class 0 devices can prematurely exhaust a switch power supply (dhcp-‐config)# network 172.16.1.0 255.255.255.0
o Difference in power allocated versus power used is a result of cabling and (dhcp-‐config)# default-‐router 172.16.1.1
device power supplies (10-‐20% loss of power) (dhcp-‐config)# option 150 ip 172.16.1.1
• 802.3af sends power over all four pairs of wires in an Ethernet cable, sharing the same Enable option 150 for voice phones, followed by the IP address of the TFTP server
wires as the data. This allows 802.3af to function even over 1000BASE-‐T, which uses all (dhcp-‐config)# dns-‐server 4.2.2.2
four pairs of wires for transmission.
Power patch panels can inject PoE onto lines >> Provides a low-‐cost PoE solution DHCP server will offer an IP address from the subnet of the router that initially receive the request. If it
was received on fa0/0.10, the router (acting as a DHCP relay agent) will add to the packet the IP
VLAN Concepts and Configuration address of the interface that received the DHCP request and then send it out to the DHCP server.
A VLAN = A Broadcast Domain = An IP Subnet
Benefits of using VLANs: If you wanted to map specific MAC address to IP via DHCP, you would need to use the hardware
• Increased performance address {mac address} syntax to reserve the scope for a specific device.
• Improved manageability – division into logical groups of users
• Physical topology independence – group users regardless of physical location Setting the Clock of a Cisco Device with NTP
• Increased security – VLAN boundary marks the end of a logical subnet Stratum 1 – Time server has a radio/atomic clock directly attached
Trunk ports are also called tagged ports because switches send frames between each other with a Stratum 2 – Device that receives its time from this server via NTP
VLAN “tag” in place. The tag is removed at the switch level before being sent to the host. The
endpoint workstations never know anything about the VLAN. (Config)# ntp server 64.209.210.20
(Config)# clock timezone ARIZONA -‐7
(Config)# clock summer-‐time CA1_DST recurring 2 Sunday March 02:00 1 Sunday November 02:00
Verify: show ntp associations | show clock

VLAN Trunking Protocol (VLAN Replication Protocol)
• Each VTP server maintain a VLAN database, which holds all the VLANs that exist
Installing CUCME (c. 4)

• VTP modes: server, client, transparent

Licensing and Models for Cisco Unified CME
Voice VLANs • IOS license
• Switch inside Cisco phone supports 802.1q tagging • Feature license – to support X number of phones
o Data packets are sent into the switch untagged • Phone user license – to support X number Cisco IP phones
o Voice packets are sent from the phone switch tagged for the voice vlan
• Configuration Steps
o Configure and verify VTP
§ All switches must run same version
§ Configuration revision
§ VTP operating mode
§ VTP domain name
o Configure and verify 802.1q trunks
§ (Config-‐if)# switchport trunk encapsulation dot1q
§ (Config-‐if)# switchport mode {trunk|dynamic desirable|dynamic auto}
§ Best practice: AUTO-‐NOT-‐USE anything auto. Stick with static configs.
§ Verify: show interfaces fastEthernet 0/24 switchport
o Create VLANs PBX Model
§ (Config)# vlan 10 • Users can dial each other using unique extension numbers
§ (Config-‐vlan)# name VOICE • Dial 9 to get an outside line
§ (Config-‐vlan)# exit • Includes a receptionist that can be used to manually route calls from the PSTN
§ (Config)# vlan 50
§ (Config-‐vlan)# name DATA

§ (Config-‐vlan)# exit
§ Verify: show vlan brief, show vtp status

Keyswitch Model Basic CUCME IP Phone Configuration (c. 5)

• Interoffice calls are rare, since the business is small enough to walk and talk to someone 1. Configure VLANs -‐ Enable portfast on switchports that will connect directly to phones.
• Incoming calls from the PSTN ring the corresponding line of all the phones in the office 2. Configure DHCP with Option 150 (TFTP)
• Each line maps to a PSTN line 3. Configure TFTP server and specify the TFTP files that can be downloaded
• “Hey Beth, pick up line one!” a. cnf-‐file location tftp://{IP of TFTP server} >> specify different TFTP server
• Taking a line off-‐hook will tie up one of the PSTN lines so that the configuration files are not saved on the CUCME router flash

Hybrid Model Ephone and Ephone-‐DN – The Keys to Ringing Phones
• Mixture of PBX and Keyswitch where phones have 1:1 mapping of PSTN lines, but also have Ephone-‐DN in its simplest form is just a directory number
internal extensions for calling each other • Single-‐line ephone-‐dn – Make/receive only one call at a time. If an additional call comes
• Using an internal lines as the default/active line, it doesn’t tie up the PSTN lines in, it will ring busy.
o (Config)# ephone-‐dn 1
o (Config-‐ephone-‐dn)# number 1000
Installing Unified CME on a Cisco Router
o (Config-‐ephone-‐dn)# exit
Files required for a CME install:
• Dual-‐line ephone-‐dn – Can handle two simultaneous calls. Useful for supporting call
• Basic files – core files for CME, including the firmware files that IP phones need
waiting, conferencing, and consultative transfers.
• GUI files – files to power the web-‐interface
o (Config)# ephone-‐dn 2 {<cr> | dual-‐line}
• XML template file – dictates the structure of the CME web-‐based management
utility. Editing this file allows you to create different levels of administrators
o You can also configure a secondary number on a line like below:
(such as a CME administrator who can only modify IP phone configurations
(Config-‐ephone-‐dn)# number 1001 secondary 5551231001
through the web-‐based utility)
• The number of ephone-‐dn’s added per CUCME install is based on the max-‐dn command
• MOH files

• Script files – various Tool Command Language (Tcl) script files to provide more
advanced functionality to CME (e.g. auto-‐attendant and automatic call
Understanding and Configuring Ephones
The following syntax adds an ephone to the CUCME router:
distributor [ACD] functions)
• (Config)# ephone 1
• Miscellaneous files – custom ringtones, backgrounds, etc.
• (Config-‐ephone)# mac-‐address aaaa.bbbb.cccc
archive tar /xtract tftp://172.16.2.5/cme-‐full-‐4.3.0.0.tar flash: • (Verify) show ephone

Max ePhone >> Max ePhone-‐DNs >> TFTP Alias >> TFTP File >> Load >> IP Source >> Create CNF-‐Files

Configuring the Cisco Unified CME Router as a TFTP Server
dir flash:/phone/7940-‐7960/P00308000500.bin alias P00308000500.bin
The alias command allows the phone to simply request the alias name on not the full location

(Config)# tftp-‐server flash:/phone/7940-‐7960/P00308000500.bin alias P00308000500.bin
Allows the firmware file to be setup for access through TFTP using the alias name Associating Ephones and Ephone-‐DNs
Basic syntax is as follows:
Configuring the Cisco Unified CME System-‐Level Functions • (Config-‐ephone)# button {phys. button} {separator} {ephone-‐dn}
You can access these features with the telephony-‐service global command: o (Config)# ephone 1
• Maximum phones and directory numbers o (Config-‐ephone)# button 1:2
o (Config)# telephony-‐service o (Config-‐ephone)# restart >> warm reboot, re-‐download config file from TFTP
o (Config-‐telephony)# max-‐ephones {#} >> max phones o (Optional) -‐ (Config-‐ephone)# button 1:2 2:3 >> assigns two buttons at once
o (Config-‐telephony)# max-‐dn {#} >> max directory numbers o (Verify) show ephone
• Firmware load files
o Make sure you have aliased phone loads for TFTP first (see above)
o (Config-‐telephony)# load {phone model no.} {firmware filename}
• Source IP address information
o (Config-‐telephony)# ip source-‐address 172.16.1.1
o Assigns IP to CME and will work as long as FastEthernet 0/0.10
interface with the IP of 172.16.1.1 is online. Otherwise, it will be
down. You can configure a loopback interface instead, which will Separators to use with ephone assignment:
make it accessible as long as the router is up.
o (Config)# interface loopback 0 : Normal ring: line rings normally and handset flashes
o (Config)# ip address 172.16.254.254 255.255.255.255 f Feature ring: line performs a triple ring for incoming calls
o (Config-‐if)# exit b Call waiting beep, no ring: line ringer suppressed, handset light still flashes
o (Config)# telephony-‐service s Silent ring: no ring or call waiting beep. Visual lights still flash.
o (Config-‐telephony)# ip source-‐address 172.16.254.254
m Monitor mode: line does not ring for incoming calls, cannot place outbound calls
o This would only be advantageous when the IP phones have more
w Watch mode: same as monitor mode, but watches ALL lines on an ephone
than one path to reach the CME router.
• Generated configuration files o Overlay line (no call waiting): shared-‐line between other ephones
o IP phone uses configuration file to determine which IP address to c Overlay line (with call waiting): same as above, but with call waiting
use when contacting the CME router (specified via the ip source-‐ x Overlay expansion/rollover: calls can rollover to empty, available lines on ephone
address command in the previous section), the firmware file it
should download, the IP phone’s extension number(s), etc. Working with Shared Lines and Overlay Options
o (Config-‐telephony)# create cnf-‐files • To configure a basic shared line, assign the same ephone-‐dn to multiple ephones. Downside is
o CNF file creation is already On that if one person is using that line, no one else can.
Updating CNF files • To share lines, but avoid having them tied up if someone is on the phone, configure multiple
CNF files update complete ephone-‐dns with same extension number. CUCME will random pick which phone to ring first.
o Once the router is configured to support IP phones, the CME • (Config-‐ephone-‐dn)# preference {0-‐10} – Sets preference of phones in a hunt group with 0
router generates a unique configuration file for each phone with being the highest preference and 10 being lowest. Preference 0 is default for all ephone-‐dns
the information specific to the IP phone. • Problem: If you configure one phone with a lower (better) preference, when the second call
o show telephony-‐service tftp-‐bindings – Shows all the files the comes in, it will typically go to the occupied phone as call-‐waiting.
CME service itself is service to the IP phones via TFTP. This is in o (Config-‐ephone-‐dn)# huntstop – Stop hunting for other matches with this same
nd
addition to the files manually specified using the tftp-‐server ephone-‐dn number. Problem, though, is that when 2 call comes in, it will
command in global config mode. simply go to busy.
nd
o XMLDefault.cnf.xml is the default configuration file handed out to o (Config-‐ephone-‐dn)# huntstop-‐channel – Stop hunting on 2 channel of a dual
nd
any IP phone that does not have an existing configuration in the line DN. When 2 call comes in, it will skip over first ephone-‐dn and send calls
CME router. Stored in system: drive, which is the router’s RAM. to next available ephone. However, it must be used in conjunction with no
Upon reboot, it is cleared out and rebuilt. huntstop. Otherwise, the huntstop command would play a busy signal before
§ more system:/its/XMLDefault.cnf.xml – Allows the call could be transitioned to another phone.
you to view the contents of the file.
§ Provides correct Ethernet port to use (always 2000,
by default), CME router IP address, phone firmware
information
§ If phone loads with the CME router generic
configuration file, it will have no lines and say
“Cisco CME” in the bottom left hand corner.

• (Config-‐ephone)# button 1o10,11 – Overlays multiple ephone-‐dns onto the same button of an
ephone. This way, when calls some in, all lines configured this way will ring simultaneously.
nd
o If you configure ephone-‐dn {DN} dual-‐line, when the 2 call comes in, it will try
nd
to reach the 2 channel of the ephone that is currently on a call. Even if you set
the separator to “o” (overlay, no call waiting), the agent would hear the call
waiting beeps. Function of dual-‐line and not the separator.

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)# number 3001
• (Config-‐ephone-‐dn)# name Maintenance
• (Config-‐ephone-‐dn)# park-‐slot
• (Config-‐ephone-‐dn)# exit

DETAILED MODE
• (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.

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)# 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)# after-‐hour exempt
• (Config-‐ephone)# exit

Configure exemptions with the pin code feature
• (Config-‐ephone)# ping {4-‐8 digit pin}
• (Config-‐ephone)# exit


Configuring a Router-‐Based DHCP Server Configuring CDRs and Call Accounting

(Config)# ip dhcp excluded-‐address 172.16.1.1 172.16.1.9 Option One: Storing CDR data in buffered memory (RAM)
(Config)# ip dhcp pool VOICE_SCOPE (Config)# logging buffered 512000
(dhcp-‐config)# network 172.16.1.0 255.255.255.0 (Config)# dial-‐control-‐mib {max-‐side | retain-‐timer}
(dhcp-‐config)# dns-‐server 4.2.2.2 1. (Config)# dial-‐control-‐mib max-‐size 700
(dhcp-‐config)# default-‐router 172.16.1.1 2. (Config)# dial-‐control-‐mib retain-‐timer 10080
(dhcp-‐config)# option 150 ip 172.16.1.1 (Verify) show logging

NTP Server Setup Option Two: Storing CDR data to a Syslog Server (e.g. Kiwi Syslog Server)
(Config)# ntp server 64.209.210.20 (Config)# gw-‐accounting syslog
(Config)# clock timezone ARIZONA -‐7 (Config)# logging 172.30.100.101
(Config)# clock summer-‐time CA1_DST recurring 2 Sunday March 02:00 1 Sunday November 02:00
Configuring Music on Hold (MoH)
System Setup (Config)# telephony-‐service
(Config-‐telephony)# date-‐format {mm | dd | yy} (Config-‐telephony)# moh {filename.wav | filename.au}
(Config-‐telephony)# time-‐format {12 | 24} (Config-‐telephony)# multicast moh 239.1.1.55 port 2123
(Config-‐telephony)# system message {Text}
Configuring Manual Local Directory Entries
Call Forward (Config-‐telephony)# directory {entry | first-‐name-‐first | last-‐name-‐last}
(Config-‐ephone-‐dn)# call-‐forward {all | busy | max-‐length | night-‐service | noan } (Config-‐telephony)# directory entry {1-‐100} {digits} name {directory name}
(Example) (Config-‐ephone-‐dn)# call-‐forward noan 1599 timeout 25
Analog Voice Ports
Call Transfer (Config)# voice-‐port 0/0/0
System Wide (Config-‐telephony)# transfer-‐system {full-‐blind | full-‐consult | local-‐consult } (Config-‐voiceport)# signal {groundStart | loopStart}
Line Specific (Config-‐ephone-‐dn)# transfer-‐mode {blind | consult} (Config-‐voiceport)# cptone {US | UK | …}
(Config-‐voiceport)# station-‐id name 3rd Floor Fax
Call Park (Config-‐voiceport)# station-‐id number 5551000
(Config-‐ephone-‐dn)# park-‐slot {reserved-‐for <dn> | timeout <sec> }
(Config-‐ephone-‐dn)# park-‐slot timeout 60 limit 10 {recall | notify | transfer | <cr>} FXO Commands
(Config)# voice-‐port 0/0/0
Call Pickup (Config-‐voiceport)# dial-‐type {dtmf | pulse}
(Config)# ephone-‐dn 1 (Config-‐voiceport)# ring number <number> (def. 1 ring)
(Config-‐ephone-‐dn)# pickup-‐group 5509
no service directed-‐pickup disables directed pickup, PickUp f(x) as local group pickup Digital Voice Ports
show controllers t1
Intercom (Config)# controller t1 1/0
intercom {ephone-‐dn number} (barge-‐in | no-‐auto-‐answer | no-‐mute} label {description} (Config-‐controller)# framing {esf | sf}
(Config-‐controller)# linecode {ami | b8zs}
(Config)# ephone-‐dn 60 (Config-‐controller)# clock source {free-‐running | internal | line}
(Config-‐ephone-‐dn)# number A100 (Config-‐controller)# ds0-‐group {1-‐23} timeslots {1-‐24} type {<cr> | … | fxo-‐loop-‐start}
(Config-‐ephone-‐dn)# intercom A101 label “Manager” ds0-‐group command is useful for fractional T1s
(Config-‐ephone-‐dn)# exit show voice port summary
(Config)# ephone-‐dn 61 Many US providers use ESF and B8ZS
(Config-‐ephone-‐dn)# number A101
(Config-‐ephone-‐dn)# intercom A100 label “Assistant” T1 CCS PSTN Interface
(Config-‐ephone-‐dn)# exit (Config)# isdn switch-‐type {primary-‐5ess | …}
Must match switch of service provider
Configure Paging (Config)# controller t1 1/0
(Config)# ephone-‐dn 81 (Config-‐controller)# pri-‐group timeslots 1-‐24 {nfas_d | service | <cr>}
(Config-‐ephone-‐dn)# number 5555 (Config-‐controller)# pri-‐group {nfas_d | service | timeslots | <cr>}
(Config-‐ephone-‐dn)# paging show voice port summary
(Config-‐ephone-‐dn)# exit
(Config)# ephone 1 POTS Dial Peers for FXS ports
(Config-‐ephone)# paging-‐dn 80 (Config)# dial-‐peer voice {#} {mmoip | pots | vofr | voip}
(Config-‐dial-‐peer)# destination-‐pattern 1101
Configure Paging (Multicast) (Config-‐dial-‐peer)# port 0/0/0
(Config)# ephone-‐dn 82 (Config-‐dial-‐peer)# exit
(Config-‐ephone-‐dn)# number 6666 (Config)# dial-‐peer voice 1102 pots
(Config-‐ephone-‐dn)# paging ip 239.1.1.100 port 2000 (Config-‐dial-‐peer)# destination-‐pattern 1102
(Config-‐dial-‐peer)# port 0/0/1
Paging Groups (Verify) show dial-‐peer voice {<cr> | summary}
(Config-‐ephone-‐dn)# paging group {PG1, PG2, …, PG10} (Verify) debug voip dialpeer

After-‐Hours Call Blocking POTS Dial Peers for a T1 interface
(Config)# dial-‐peer voice 2000 pots
DEFINE AFTER-‐HOURS (Config-‐dial-‐peer)# destination-‐pattern 2…
(Config)# telephone service (Config-‐dial-‐peer)# no digit-‐strip
(Config-‐telephony)# after-‐hours day mon 17:00 08:00 (Config-‐dial-‐peer)# port 1/0:23
(Config-‐telephony)# after-‐hours date jan 1 00:00 00:00 This will send all 2XXX extensions out port 1/0:23 (the T1 PRI interface)
(Verify) show dial-‐peer voice {<cr> | summary}
SPECIFY BLOCKING PATTERNS (Verify) debug voip dialpeer
(Config)# telephone service
(Config-‐telephony)# after-‐hours block pattern {1-‐32} 91.......... VoIP Dial Peers
(Config-‐telephony)# after-‐hours block pattern {1-‐32} 9011T • (Config)# dial-‐peer voice 2000 voip
(Config-‐telephony)# after-‐hours block pattern {1-‐32} 91900....... 7-‐24 • (Config-‐dial-‐peer)# destination-‐pattern 2...
• 7-‐24 blocks this pattern at all times, 24 hours, 7 days a week • (Config-‐dial-‐peer)# session target ipv4:10.1.1.2
• No exceptions when using this keyword • (Config-‐dial-‐peer)# codec g711ulaw

CREATE EXEMPTIONS Primary difference between the POTS and VoIP dial peer configuration is the use of the session target
(Config)# ephone 1 command rather than the port command.
(Config-‐ephone)# after-‐hour exempt • ipv4:ip address
(Config-‐ephone)# exit • dns:name (H.323 gatekeeprs, SIP proxy servers, etc.)

CREATE PIN CODES FXS/FXO PLAR Configuration
(Config)# ephone 2 • (Config)# voice-‐port 0/0/0
(Config-‐ephone)# ping {4-‐8 digit pin} • (Config-‐voiceport)# connection plar 1102
(Config-‐ephone)# exit

SHOW/DEBUGS
CREATE PIN TIMEOUTS
show telephony-‐service tftp-‐bindings
(Config)# telephony-‐service
debug tftp events
(Config-‐telephony)# login timeout 120 clear 23:00
debug ephone register
• clear: absolute time when pin timeout, users can then log back in show voice port summary
• Requires a phone reset show dial-‐peer voice {<cr> | summary}
debug voip dialpeer
show dialplan number number -‐ determine which dial peer will match a specific string


a. Voice Activity Detection (VAD) – Router can detect silence in

Exam Topics Covered in Part III: conversation and not send RTP data (35%)
b. RTP Header Compression – Sends first packet as full packet, then strips
Describe VoIP components and technologies the redundant information out of future packets. Compresses 40-‐byte
Describe the process of voice packetization IP/RTP/UDP header down to aprox. 2-‐5 bytes (40% w/ G.729(
Describe RTP and RTCP
Describe the function of and differences between codecs The Role of Digital Signal Processors
Describe H.323, MGCP, SIP, and SCCP signalling protocols 16 = 1 DSP chip
DSP: Performs all the sampling, encoding, and compression functions on audio coming into router.
Describe and configure gateways, voice ports, and dial peers to connect to the PSTN and service • PVDM2-‐8 >> 0.5 DSP chips
provider networks • PVDM2-‐16 >> 1 DSP chip
Describe the function and application of a dial plan • PVDM2-‐32 >> 2 DSP chips
Describe the function and application of voice Gateways
• PVDM2-‐48 >> 3 DSP chips
Describe the function and application of voice ports in a Gateway
• PVDM2-‐64 >> 4 DSP chips
Describe the function and operation of call-‐legs

Describe and configure voice dial peers
High Complexity requires
Describe the differences between PSTN and Internet Telephony Service Provider circuits
two times as many DSPs

as the medium
Describe and configure a Cisco network to support VoIP
complexity codecs.
Describe the purpose of VLANs in a VoIP environment

Describe the environmental considerations to support VoIP

Configure switched infrastructure to support voice and data VLANs

Describe the purpose and operation of PoE
Identify the factors that impact voice quality Understanding RTP and RTCP
Describe how QoS addresses voice quality issues Protocols of voice:
Identify where QoS is deployed in the UC infrastructure • Real-‐time Transport Protocol (RTP)
• Real-‐time Transport Control Protocol (RTCP)

• RTP operates at the transport layer on top of UDP
Gateway and Trunk Concepts (c. 7) o UDP: port numbers (i.e. session multiplexing), header checksums
o RTP: payload type (audio/video), sequence number, time stamp
Choosing a Voice Codec
Questions to ask when choosing a voice codec for your network:
• Is the codec support on all VoIP devices in my network?
• How many Digital Signal Processor (DSP) resources needed to code audio using codec?
• Does codec meet satisfactory quality levels for my network for all audio types?
• How much bandwidth does the codec consume?
• How does codec handle packet loss?
• Does codec support multiple sample sizes?

• RTP engages and chooses a random port number (16,384-‐32,767) for each RTP stream
The Process of Converting Voice to Packets • RTP streams are one-‐way. Therefore, two RTP streams per call.
• Cisco designed all its phones to use G.711 or G.729 • Once call established, RTCP engages (statistics reporting) and sends packets at least
• G.711 is “common ground” codec that ‘s compatible cross-‐platform (7-‐11s everywhere) once every 5 seconds >> not as important as RTP
• G.729 comes in two different variants: o Packet count
o G.729a (annex A) – Sacrifices some quality to achieve a more processor-‐ o Packet delay
efficient coding process o Packet loss
o G.729b (annex B) – Support for Voice Activity Detection (VAD), which o Jitter (delay variations)
makes voice transmission more efficient. • RTP uses even-‐numbered ports
• RTCP uses the odd-‐numbered port following the RTP port.

Internet Low Bitrate Codec: One Codec to Rule Them All
Internet Low Bitrate Codec (iLBC)
• Low Bit rate: 15.2 kbps (20 ms sample size) or 13.3 kbps (30 ms sample size)
• High codec complexity
• Best Quality: Comparable to G.711; one of best quality levels for any compressed codec
• Loss Covering Methods: Dropped packets impact less than any other codec
• Limited Support: Only the newest, such as 7906G, 7942G, 7962G, 7979G

Trunking CME to PSTN
Understanding Analog Connections:

• Foreign Exchange Office (FXO) connect to the PSTN CO or legacy PBX
o Controls number of concurrent, active PSTN phone calls
A = Audio Bandwidth (in bytes)
• Foreign Exchange Station (FXS) – connect to analog devices (i.e. telephones, fax, etc.)
(E * Codec Bandwidth)/8 • Ear and Mouth (E&M) – connects directly to a PBX system with E&M interfaces
Where codec bandwidth is in bps and sample size is in sec (ex. 64000 * 0.03)
Understanding Digital Connections:
B = Layer 2 Overhead • Channel associated signaling (CAS) T1/E1: attempt to squeeze signaling into same
• Ethernet: 20 bytes channel as the audio. Able to support 24 DS0 audio channels out of a T1 interface, 30
• Frame Relay: 4-‐6 bytes channels in Europe, etc.
• PPP: 6 bytes • Common channel signaling (CCS) or Primary Rate Interface (PRI) T1/E1: separate
C = Layer 3 Overhead (40 bytes) signaling into a dedicated channel, leaving the full 64 kbps of each DS0 for audio only.
• IP: 20 bytes o 23 DS0 audio channels in US, E1 still 30 channels
• UDP: 8 bytes o PRI describes ISDN implementation of CCS, more commonly used by PSTN
• RTP: 12 bytes Cos and PBX vendors
D = Additional Overhead – such as VPN connection
• GR3/L2TP: 24 bytes BOSON and CISCO PRESS
• MPLS: 4 bytes • Dial peer used to connect to a PBX system using a digital T1 CAS config = POTS
• IPsec: 50-‐57 bytes • Default codec used by a VoIP dial peer = G.729
E = Sample Size – specific time interval, default 20 ms • Apply a translation rule with a translation profile
• Digit manipulation command that will work under a VoIP dial peer = translation-‐profile\
Total Bandwidth Requirements (in kbps) • Frequency of human speech = 200-‐9000 Hz
• Frequency reproduced by Nyquist theorem = 300-‐4000 Hz
((8 * (A+B+C+D)) * (1000/E)) – Bandwidth Savings • Mandatory parts of an outbound POTS dial peer: destination pattern and port
• For best voice quality, delay should be maximum of 150ms delay
Example: G.729, sample size 20 ms, using Ethernet without VPN = 32 kbps
Example: G.711, sample size 30 ms, using Ethernet without VPN = 80 kbps

Calculating Codec Bandwidth Requirements
1. Determine audio bandwidth required for audio codec itself
a. Bigger sample size >> Fewer packets >> Less bandwidth
2. Determine data link, network, and transport layer overhead
3. Add any additional overhead amounts
4. Add it all together
5. Subtract bandwidth savings measures

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

• 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
b. (Config-‐voiceport)# dial-‐type {dtmf | pulse}
2. Rings before router answers cal
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

;


POTS dial peers Private Line Automatic Ringdown

Defines voice reachability information for any traditional voice connection Automatically dial a number when the phone goes off-‐hook
• If no IP address >> POTS connection
• Configuring for FXS ports FXS PLAR Configuration
o (Config)# dial-‐peer voice {#} {mmoip | pots | vofr | voip} • (Config)# voice-‐port 0/0/0
o (Config-‐dial-‐peer)# destination-‐pattern 1101 • (Config-‐voiceport)# connection plar 1102
o (Config-‐dial-‐peer)# port 0/0/0
o (Config-‐dial-‐peer)# exit FXO PSTN Connections
o (Config)# dial-‐peer voice 1102 pots destination pattern command does not assist with directing inbound calls. If a call were to
nd
o (Config-‐dial-‐peer)# destination-‐pattern 1102 enter the CME router, they would hear a 2 dial tone. The router would be waiting for the
o (Config-‐dial-‐peer)# port 0/0/1 caller to enter a valid extension number. Confusing, so we can configure FXO PLAR
o (Verify) show dial-‐peer voice {<cr> | summary}
o (Verify) debug voip dialpeer
• Configuring for a T1 interface
o (Config)# dial-‐peer voice 2000 pots
o (Config-‐dial-‐peer)# destination-‐pattern 2…
o (Config-‐dial-‐peer)# no digit-‐strip FXO PLAR Configuration
o (Config-‐dial-‐peer)# port 1/0:23 NOTE: Configuring PLAR connections for incoming calls is something you would only need to do for
o This will send all 2XXX extensions out port 1/0:23 (the T1 PRI interface) analog FXO trunks. Digital PSTN connections (such as T1 or E1) receive DNIS information for incoming
o (Verify) show dial-‐peer voice {<cr> | summary} calls, which the router can use for Direct Inward Dial (DID) services.
o (Verify) debug voip dialpeer • (Config)# voice-‐port 2/0/0
• (Config-‐voiceport)# connection plar 1500
• Digits Stripping Rule of POTS Dial Peers: The router will automatically strip any explicitly • (Config-‐voiceport)# exit
defined digit from a POTS dial peer before forwarding the call. • (Config)# voice-‐port 2/0/1
o (Config-‐dial-‐peer)# destination-‐pattern 9………. • (Config-‐voiceport)# connection plar 1500
o Strip the 9 before forwarding • (Config-‐voiceport)# exit

VoIP dial peers Understanding Router Call Processing and Digit Manipulation
Defines voice reachability information for any VoIP connection accessed via IP address
• (Config)# dial-‐peer voice 2000 voip Important Rules:
• (Config-‐dial-‐peer)# destination-‐pattern 2... 1. The most specific destination pattern always wins
• (Config-‐dial-‐peer)# session target ipv4:10.1.1.2 2. Once a match is found, the router immediately processes the call
• (Config-‐dial-‐peer)# codec g711ulaw 3. When a router receives a voice call, it must always match a dial peer in some way in
order for the router to process the call.
Primary difference between the POTS and VoIP dial peer configuration is the use of the session
target command rather than the port command. show dialplan number number -‐ determine which dial peer will match a specific string
• ipv4:ip address
• dns:name (H.323 gatekeeprs, SIP proxy servers, etc.) Methods routers use to match inbound dialer peers
1. Match dialed number (DNIS) using incoming called-‐number dial-‐peer config command
If the codec values do not match between the two routers, the call will fail and return a reorder tone 2. Matchcaller ID information (ANI) using answer-‐address dial-‐peer config command
(fast busy signal). This is commonly called a codec mismatch. The default codec value for VoIP dial 3. Match caller ID information (ANI) using destination-‐pattern dial-‐peer config command
peers is G.729. 4. Match an incoming POTS dial peer by using port dial-‐peer config command
5. If no match has been found using the previous four methods, use dial peer 0.

Using Dial-‐Peer Wildcards Dial Peer 0 Characteristics
Period (.) Matches any dialed digit from 0-‐9 or the * key (20.. = 2011, 2022, 2045)
• Any voice codec
Plus (+) Matches one or more instances of the preceding digit (5+23 = 5523, 55523, 555523)
• No DTMF relay
Brackets ([ ]) Matches a range of digits ([1-‐3]22 = 122, 222, 322)
• IP Precedence 0 – strips all QOS markings, treated same as data traffic
Caret (^) Does NOT match (^[1-‐3]22 = 022, 422, 522, 622)
T Matches any number of dialed digits (from 0-‐32 digits) • Voice Activity Detection (VAD) enabled – when silent >> no traffic >> saves bandwidth
Comma (,) Inserts a one-‐second pause between dialed digits • No Resource Reservation Protocol (RSVP) support – no bandwidth reserved
• Fax-‐rate voice – router will limit bandwidth available to fax signals to maximum allowed
• The pound symbol (#) is not wildcard symbol. It processes a dialed numbered without waiting. by VoIP code >> could devastate fax calls
• No application support – cannot interact with applications like IVR
• If you make a dial peer using only the T wildcard, Cisco recommends using .T so that it requires
• No Direct Inward Dial (DID) support – cannot use DID feature to automatically forward
user to dial at least one number. Otherwise, a phone left off-‐hook for too long without a dialed calls from an outside PSTN to internal devices.
digit will match the destination pattern.
• If you define a dial peer of 9T, you would have to wait for the interdigit timeout (T302 timer)
Using Digit Manipulation
before the call would be processed. You can use the pound symbol (#) to get around this, but it
Adding/Removing digits from dialed number to help a call reach an intended destination
would be easier to simple make better dial peers.
POTS prefix [digits[

POTS forward-‐digits [number]
Sample PSTN Destination Patterns for North America
POTS [no] digit-‐strip
• [2-‐9]......... >> Used for 7-‐digit dialing areas
Global num-‐exp [match] [set]
• [2-‐9]..[2-‐9]…… >> Used for 10-‐digit dialing areas
Global/VOIP/POTS voice translation-‐profile [name]
• 1[2-‐9]..[2-‐9]…… >> Used for 11-‐digit dialing areas
• [469]11 >> Used for service numbers such as 411, 611, 911
• 011T >> Used for international dialing

Configuring a North American PSTN Dial Plan
• (Config)# dial-‐peer voice 90 pots
• (Config-‐dial-‐peer)# description Service Dialing
• (Config-‐dial-‐peer)# destination-‐pattern 9[469]11
• (Config-‐dial-‐peer)# forward-‐digits 3
o Specify number of right-‐justified digits to forward
• (Config-‐dial-‐peer)# port 1/0:1
• (Config-‐dial-‐peer)# exit
• (Config-‐dial-‐peer)# description 10-‐Digit Dialing
• (Config-‐dial-‐peer)# destination-‐pattern 9[2-‐9]..[2-‐9]……
• (Config-‐dial-‐peer)# description 11-‐Digit Dialing
• (Config-‐dial-‐peer)# destination-‐pattern 91[2-‐9]..[2-‐9]……
• (Config-‐dial-‐peer)# forward-‐digits 11
• (Config)# dial-‐peer voice 93
• (Config-‐dial-‐peer)# description International Dialing
• (Config-‐dial-‐peer)# destination-‐pattern 9011T
• (Config-‐dial-‐peer)# prefix 011
o Add any specified digits to the front of the dialed number


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

NEVER DO VERY POOR FRAMING

Cisco AutoQoS Configuration on Router Interfaces
• (Config)# interface fa0/1
• (Config-‐if)# auto qos voip trust


Subscriber Mailbox versus General Delivery Mailbox

Exam Topics Covered in Part IV: • Subscriber Mailbox – user mailbox, served individual
• General Delivery Mailboxes (GDM) – assigned to group of people, no pin assigned for
Implement voicemail features using Cisco Unity Express login, group member must login to their own mailbox which gives access to GDM
Describe the Cisco Unity Express hardware platforms
Configure the foundational elements required for Cisco Unified Communications Manager Express to Mailbox Subscriber Features
support Cisco Unity Express
• Mailbox login, passwords, and PINs
Describe the features available in Cisco Unity Express
Configure AutoAttendant services using Cisco Unity Express
Configure basic voicemail features using Cisco Unity Express

Cisco Unity Express Concepts (c. 9)

• 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

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


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

o (Config-‐dial-‐peer)# no vad

§ Must be disabled for CME dial peer to talk with CUE
o (Config-‐dial-‐peer)# description AutoAttendant
o (Config-‐dial-‐peer)# description AVT
• Telephony Service Configurations for CUE
o (Config)# telephony-‐service
o (Config-‐telephony)# voicemail 7000
o (Config-‐telephony)# web admin system name cmeadmin password cisco
§ Defines credentials CUE will use to access CUCME GUI
o (Config-‐telephony)# dn-‐webedit
o (Config-‐telephony)# time-‐webedit
• MWI Configuration
When CUE sends the digits #401000 to CUCME, CME will match digits to ephone-‐dn 19.
The ephone-‐dn will strip the static digits #40, leaving the digits 1000. Then, CUCME will
turn on MWI for extension 1000.

o (Config-‐ephone-‐dn)# number #40...
o (Config-‐telephony)# mwi on MWI
o (Config)# ephone-‐dn 19 • Does the MWI light turn on when a message is left?
o (Config-‐ephone-‐dn)# number #40...
• Does the MWI light turn off when a message has been heard?
o (Config-‐telephony)# mwi on
o Verify that the ephone-‐dn configuration for the MWI in CUCME matches

the MWI configuration within CUE.
CUE Initialization Wizard (via GUI)
o debug ephone mwi – to see MWI process in real-‐time
• Set hostname, user name, password
• Import CUCME Users
• Set user defaults
• Set CUE call handling

CUE Initialization Wizard Verification (via CLI)
• show run

CUE Troubleshooting

Compatibility
• Is CUE recognized by CUCME?
o Does the Cisco Service Engine show up in the config?
o show version
CUE Trace Files
Cisco Unity Express Configuration • Debugs are not supported on CUE. Instead, they use trace files.
• Is CUE up and running in CUCME? • Traces must be initiated from the CUE console, not CUCME
• Are both service-‐engine and line protocol in an up state? • trace ccn subsystemsip all – initiates tracing by module, entity, and activity
o show interface service-‐Engine 1/0 • show trace buffer tail -‐ view trace information in real-‐time
o Check configuration of the service-‐engine • show trace buffer long – view trace information stored in buffer
• Is the service-‐module in a steady state? • show trace buffer long paged – makes output easier to read
o service-‐module service-‐engine 1/0 status • show log name messages.log – displays contents of the messages.log file
o service-‐module service-‐engine 1/0 reload – If the service engine is not in • show logs – view current atrace.log and messages.log files
a steady state, and the config appears to be correct, you can reload the • copy log atrace.log url ftp://10.1.1.11/atrace.log -‐ upload atrace.log for Cisco TAC
module
o service-‐module service-‐engine 1/0 reset – If service in a failed state and Restoring CUE to Factory Defaults
will not recover after reload. May result in data loss. All data and configuration will be lost when factory defaults are restored
IP Connectivity
• Can I ping the IP address of the service-‐engine?
• Is there a route to the IP address of the service-‐engine?
Call Routing
• Are calls to the voicemail, auto-‐attendant, and AVT pilot numbers answered by CUE?
o Verify that dial peers are configured correctly in CUCME
o Verify that dial peer numbers in CUCME match the pilot numbers in CUE
o show dial-‐peer voice 7000

o debug ccsip calls


Exam Topics Covered in Part V:

Implement UC500 using Cisco Configuration Assistant
Describe the function and operation of Cisco Configuration Assistant
Configure UC500 device parameters
Configure UC500 network parameters
Configure UC500 dial plan and voicemail parameters
Configure UC500 SIP trunk parameters
Configure UC500 voice system features
Configure UC500 user parameters

Perform basic maintenance and operations tasks to support the VoIP solution
Describe basic troubleshooting methods for Cisco Unified Communications Manager Express
Explain basic troubleshooting methods for Cisco Unity Express
Explain basic maintenance and troubleshooting methods for UC500

Introducing Smart Business Communications System (c. 11)

Simplicity
• Predefined hardware and licensing configuration
• Plug and Play in a Feature-‐Rich Suite – Each member of the SBCS suite designed to be
recognized/managed with Cisco Configuration Assistance (CCA, GUI-‐based tool to
configure the UC5XX platform)
• Predefined Use Case Assumptions -‐ Includes best-‐practice configurations for QoS,
security, VLAN segmentation, IOS firewall, wireless security
• Cisco Configuration Assistant Tool

Manageability -‐ The CCA contains a list of features coined “Cisco Smart Assist”
• Automatic assigned of phone extensions
• Dial plan setup
• Voice and data VLAN settings
• Firewall policy setup and activation
• Cisco best-‐practice QoS policies

Complete

Affordability

Components of the Smart Business Communications System
Common Deployment Scenarios

• 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.


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

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Assign ports to VLANs

Keyswitch Model Basic CUCME IP Phone Configuration (c. 5)

Configuring a Router-‐Based DHCP Server Configuring CDRs and Call Accounting

a. Voice Activity Detection (VAD) – Router can detect silence in

POTS dial peers Private Line Automatic Ringdown

Subscriber Mailbox versus General Delivery Mailbox

o (Config-‐dial-‐peer)# no vad

Exam Topics Covered in Part V:

Você também pode gostar

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640 460 Iiuc

Enviado por

Direitos autorais:

Formatos disponíveis

Assign ports to VLANs

Keyswitch Model Basic CUCME IP Phone Configuration (c. 5)

Configuring a Router-­‐Based DHCP Server Configuring CDRs and Call Accounting

a. Voice Activity Detection (VAD) – Router can detect silence in

POTS dial peers Private Line Automatic Ringdown

Subscriber Mailbox versus General Delivery Mailbox

o (Config-­‐dial-­‐peer)# no vad

Exam Topics Covered in Part V:

Você também pode gostar

Configuring a Router-‐Based DHCP Server Configuring CDRs and Call Accounting

o (Config-‐dial-‐peer)# no vad