Cisco Ccna Voice.pdf 4q6p5z

No cdp enable No cdp run

Frequencies of Audio Signals

Human Ear Human Speech Telephone Channel

Lower Limit (Hz)

Upper Limit (Hz)

20 200 300

20000 9000 3400

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

http://www.studyshorts.co.uk

M Morgan ©2010

Page 7 of 50

CCNA Voice IIUC (640-460)

POTS Technologies Analogue Connections Two connectionsGround / Tip – 0v Battery / Ring – -48v

PSTN Signalling Signalling  Ground Start – The station/PBX will ground both ring and tip to request a dial tone.  Loop Start –When a phone is on hook the loop is open, when taken off hook the station will close the loop to the exchange to request a dial tone. Typically used in home environments as this is susceptible to glare.  Glare – If an incoming call happens at the same time as an outgoing line is requested in a PBX environment, they can become connected causing confusion to the outgoing caller.

Supervisory Signalling  On-hook – When the phone is on-hook, the connection between the tip and ring wires is broken and no electrical signal es between them.  Off-hook – When the phone is off-hook, the phone connects the tip and ring wires, completing the circuit and allowing electrical signal to .  Ringing – To cause an analogue phone to ring, the phone company sends an alternating current (AC).

Informational Signalling  Dial tone – Indicates the phone company is ready to receive digits  Busy – Indicates the remote phone is already in use  Ringback – Indicates the remote phone is currently ringing  Congestion – Indicates the long-distance telephone network is not able to complete the call  Reorder – Indicates the local telephone company is not able to complete the call  Receiver off-hook – Indicates the local receiver has been off-hook for an extended period of time  No such number –Indicates the dialed number is invalid  Confirmation – Indicates the telephone company is attempting to complete the call


M Morgan ©2010

Page 8 of 50

CCNA Voice IIUC (640-460) Address Signalling  Dual-tone multi frequency (DTMF) – The buttons on a telephone keypad use a pair of high and low electrical frequencies (thus “dual-tone”) to generate a signal each time a caller presses a digit.  Pulse – The rotary-dial wheel of a phone connects and disconnects the local loop circuit as it rotates to signal specific digits.

E1 / T1 Signalling T1 CAS – Robbed Bit Signalling Least significant bit in every 6th frame is signalling. Reduces quality very slightly. Frame 1 ... Frame 5 Frame 6

1st DS0 ... 1st DS0 1st DS0 S

2nd DS0 ... 2nd DS0 2nd DS0 S

3rd DS0 ... 3rd DS0 3rd DS0 S

... ... ... ...

24th DS0 ... 24th DS0 24th DS0 S

T1 “Giganto” Frame – a set of 24 DS0 (T1). 193 bits at a time, 192 for data and 1 for framing. T1 Super Frame (SF) – 12 Giganto frames at a time. For each SF there is two signalling bits per channel (A & B) T1 Extended Super Frame (ESF) – 24 Giganto frames at a time. For each ESF there are four signalling bits (A, B, C & D). This is currently used for most if not all T1 providers Frame 6 Frame 12 Frame 18 Frame 24

1st DS0 A 1st DS0 B 1st DS0 C 1st DS0 D

2nd DS0 A 2nd DS0 B 2nd DS0 C 2nd DS0 D

3rd DS0 A 3rd DS0 B 3rd DS0 C 3rd DS0 D

E1 CAS Signalling Dedicated Framing and Signalling channels (DS0). Channel 0 (1st timeslot) is framing and channel 16 (17th timeslot) is Signalling, channels 1-15 & 17-31 are voice. Every signalling DS0 is broken down into two nibbles two provide signalling (A, B, C & D) for two DS0 voice channels. The first frame contains signalling for DSO 1 and DS0 31, the next contains signalling for DS0 2 and DS0 30 etc. Signalling is compatible with T1 CAS but very rarely used.


M Morgan ©2010

Page 9 of 50

CCNA Voice IIUC (640-460) T1 and E1 CCS Signalling Like E1 CAS a dedicated DS0 channel (17th timeslot) is used for Signalling. Uses a signalling protocol (Typically ISDN Q931, SS7) rather than four bit signalling. CCS leaves 23 channels available for voice on T1 and 30 channels on E1.


M Morgan ©2010

Page 10 of 50


IP Voice Technologies Cisco Voice Infrastructure Model Layer 1 2 3 4

Purpose

Examples

Endpoints Applications Call Processing Infrastructure

IP Phone, Cell Phone, Video Phone, IM Client Voice Mail, Conference Call apps, Call Centre Apps, 911 Series Unified Communications Manager, UCME, UC500 ASA Firewall Voice Router/Gateway, Voice Switch

Call Processing Layer Max s Redundancy Host

UC500

CME

CCMBE

CCM

48 no Router

250 No Router

500 No Server

30000+ Yes Server

Cisco Unified Communications 500 (UC500) – Appliance providing firewall, NAT, Integrated Voicemail & Auto Attendant, Built in FX0 & FXS Ports, VPN, Optional Wireless and Music on Hold. This is a part of the Cisco Smart Business Communications System (SBCS) range. Cisco Unified Communications Manager Express (CME) – Next step up from the UC500. Cisco Unified Communications Manager Business Edition (CCMBE) – Provides CCM call processing, Cisco Unity Connection and Cisco Unified Mobility applications. Cisco Unified Communications Manager (CCM) – Call processing only. s redundancy and clustering.

Applications Layer Cisco Unity Express – Voicemail hardware (Network module or AIM) physically installed into a ing router. s up to 250 s. This unit provides limited IVR capabilities in order to provide an Automated Attendant system. Cisco Unity Connection – Cut down Cisco Unity ing up to 500 s (7500 dedicated server). Also provides Advanced Call Routing facilities to calls can be routed based on rules, time of day, caller ID etc. Cisco Unity – Full unified solution integrating with Exchange, Lotus Notes & Novell GroupWise. Up to 7500 s per server. s redundancy. Cisco Unified Centre – Provides ACD functionality to a call centre environment. Cisco Unified Meeting Place - Provides a multimedia conference solution that gives you the capability to conference voice, video, and data into a single conference call. For example, multiple offices could participate in a conference call using IP phones, live video feeds, and instant messenger clients. The http://www.studyshorts.co.uk

M Morgan ©2010

Page 11 of 50

CCNA Voice IIUC (640-460) conference call could include PowerPoint presentations, shared whiteboards, or live demonstrations. The organization could also choose to record the conference call for playback at a later time. Cisco

Unified Presence - Provides status and reach ability information for the s of the voice network. For example, Joe might check the status for Samantha and find that she is available on an instant messenger client but is currently engaged in a video call. Cisco Unified Mobility - Allows s to have a single phone number that they can link to multiple devices. For example, Mike could have the phone number +442920 454343 that links to his desk phone, cell phone, and instant messenger client. Cisco Emergency Responder - Because VoIP clients have the ability to “roam around” the network using wireless phones, Soft Phones, or extension mobility functionality, emergency calls (911/999) could pose a location problem. Cisco Emergency Responder (ER) dynamically updates location information for a based on the current position in the network and feeds that information to the emergency service provider if an emergency call is placed. The Cisco ER product also helps manage emergency calls in a centralized IP telephony deployment, ensuring that branch office.

Infrastructure Layer The Infrastructure layer consists of the IP infrastructure to enable a VoIP telephone network (switched, routers etc). The uptime of a traditional PBS system if 99.999 percent so as a result the main factors in the IP infrastructure layer is redundancy and QoS to ensure good uptime and good quality speech.

Signalling SIP - Developed by the IETF. This uses text strings similar to HTML for signalling. SIP itself is only responsible for setting up and tearing down sessions between endpoints, the actual session is transferred typically using RTP over UDP. Registrar, Redirect, Location and Proxy servers can be used. H.323 - Created by the ITU-T to allow simultaneous voice, video and data transmission primarily across ISDN links. The signalling is derived from Q.931 signalling and as a consequence is very difficult to interpret. This is a peer to peer protocol so each gateway in the system is fully independent of any other and needs full configuration for all other gateways. This istrative burden can be reduced by incorporating a H.323 Gatekeeper, where the gatekeeper would have the full knowledge of the infrastructure and all Gateways would ask the Gatekeeper how to find other non local extensions. The Gatekeeper can also perform other tasks such as CAC (Call ission Control) and bandwidth management. H.232 is also responsible for the transport of the media stream. This is the only signalling protocol that s Fax connected to a Cisco ATA. MG - Developed by Cisco and the IETF is a system which puts voice gateways under control of a centralised call agent. The gateway is considered a dumb device, every action such as a phone going off hook or a button pressed is relayed to the MG call agent to ask what to do next such as play a dial tone. This is not ed by CME. http://www.studyshorts.co.uk

M Morgan ©2010

Page 12 of 50

CCNA Voice IIUC (640-460) SC - Cisco proprietary protocol used to control Cisco endpoints (IP Phones, ATA 186 etc). Works in a similar fashion to MG, the end device communicates with CME for every action

Body Industry Used on Gateways Used on Cisco phones Architecture

H.323

MG

SIP

SC

ITU Excellent Yes No Peer to Peer

IETF Fair Yes Limited Client / Server

IETF Very Good Yes Yes Peer to Peer

Proprietary Limited Yes Client / Server

IP Transport RTP - The media stream is carried using RTP on a negotiated UTP port between 16384 and 32767 (Even numbers). RT – A RT session is created at the same time as the RTP session, this is used to relay statistics between the participating devices (and CME). Typically Packet count, Packet delay, Packet loss and Jitter statistics is transmitted. Uses odd number UTP ports

IP Overhead As raw voice data is sent across a network link, layer 2 and layer 3 frame headers are added to the stream as below. Layer 2 Ethernet – 18 bytes Frame Relay – 4 to 6 bytes Point to Point Protocol (PPP) – 6 bytes Layer 3 Total of 40 Bytes IP – 20 bytes Version

Header Length Identification

TTL

Type of Service Flags Protocol

Total Length Fragment Offset Header Checksum

Source IP Address Destination Source Address

UDP – 8 bytes Source Port (16bits) Length (16bits)

Destination Port (16bits) Checksum (16bits)

Real-time Transport Protocol (RTP) – 12 bytes Ver P X CC M PT http://www.studyshorts.co.uk M Morgan ©2010

Sequence Number Page 13 of 50

CCNA Voice IIUC (640-460) Timestamp SSRC Identifier CSRC Identifiers

Compressed RTP Compresses the network and transport layer headers from 40 bytes down to 2 bytes (without checksum) or 4 bytes (with checksum). This is considered very processor intensive so is only used on low bandwidth links (T1 or lower)

Problems with Digital Voice Bandwidth – 21kbps to 320kbps per call depending on codec. QoS can help prioritise voice during bandwidth use peaks. Delay – A maximum one-way delay of 150ms, 200ms is considered the ultimate limit. Jitter – Change of delay between packets, usually caused when there are multiple data paths available between the endpoints. A maximum one-way jitter delay of 30ms is advisable. A “De-Jitter Buffer” can be used to reduce the impact of jitter by buffering a small amount of speech in the device before playing it. Cisco devices implement a variable sized de-jitter buffer to tune to the connection quality. As a downside it introduces additional delay. Packet Loss – As packets are lost there will be holes in the speech. Less than 1% is advisable.

Causes of Delay Transmission delay – The physical time it takes for the packet to travel the wire (Fixed). Serialization delay – The time it takes to place the bits on the wire (Fixed). Codec delay – The time the codec takes to convert voice into a PCM stream. Queuing delay – The time the packet remains in a queue waiting for transmission. QoS can influence this by putting packets in to a high priority queue.

QoS Data applications classes Mission critical – Critical to the running of the business. Transactional – Applications interact with the s and required rapid response times. http://www.studyshorts.co.uk

M Morgan ©2010

Page 14 of 50

CCNA Voice IIUC (640-460) Best Effort – Noncritical – web browsing, email, ftp etc. Scavenger – Non productive and no business need. P2P apps etc.

Trust Boundary All devices are capable if marking packets for priority. Upstream devices can either trust these markings or generate new marking by inspecting the traffic. The most efficient way is to mark the traffic at the closest point to the end device, this allows more efficient transport of the packet throughout the network and avoids the Distribution and especially the Core switches classifying traffic. When configuring AutoQoS it is possible to control whether the downstream devices marking are to be trusted.

Queuing Allows changing the default queuing method on Cisco devices (routers and switches). By default traffic is sent on a FIFO basis. Low Latency Queuing (LLQ) is the most popular. A single “priority queue” and many “custom queues”.

AutoQoS Switch (config-if) # auto qos voip (config-if) # auto qos voip cisco-phone (config-if) # auto qos voip cisco-softphone (config-if) # auto qos voip trust

The first three options will only enable the trust boundary if a Cisco phone is detected using CDP. The last command will trust any marking regardless, typically used where non Cisco phones are used. Router (config-if) # auto qos voip (config-if) # auto qos voip trust

NotesEnsure serial links have a defined bandwidth using the ‘bandwidth XXX’ command under the interface as routers cannot automatically detect it.

MQC – Modular QoS CLI Class map Used to identify and classify traffic. Matches onhttp://www.studyshorts.co.uk

M Morgan ©2010

Page 15 of 50

CCNA Voice IIUC (640-460)   

ACL Input interface NBAR (Network based application recognition). This looks at the up layers to find the application

Mode

Description

Command

(config) (config) (config) (config-cmap) (config-cmap) (config-cmap)

Create a match all class map (default) Create a match any class map Create a match all class map Match on an ACL Match on an input interface Match based on NBAR application signature

Class-map classname Class-map match-any classname Class-map match-all classname Match access-group Match input-interface Match protocol protocol

Match-any signifies an OR condition between statements Match-all signifies an AND condition between statements Policy-map Controls what to do with traffic

Mode

Description

Command

(config) (config-pmap) (config-cmap-c) (config-cmap-c) (config-cmap-c)

Create a policy map Set a class map for this policy Set a priority bandwidth of kbps Set a priority bandwidth of percentage of interface bandwidth Set bandwidth of kbps

Policy-map type policyname Class classname Priority kbps Priority percent percent Bandwidth kbps

Example(config) # Class-map match-any WEB_TRAFFIC (config-cmap) # Match protocol http (config-cmap) # Match protocol https

- Class map to match on either HTTP or HTTPS

(config) # Class-map match-all VOIP (config-cmap) # Match protocol rtp

- Class map to match on RTP traffic

(config) # policy-map VOIP (config-pmap) # class VOIP (config-pmap-c) # priority 4000

- Policy map to give priority bandwidth to VOIP

(config) # interface Ethernet 0 (config-if) # service-policy output VOIP

- Set the QoS on an interface


M Morgan ©2010

Page 16 of 50


Analogue to Digital Conversion / Codecs Conversion 1. 2. 3. 4.

Sample the waveform – Pulse Amplitude Modulation Calculate the number representing each sample (quantisation) Convert to binary – Pulse Code Modulation (G711a etc) Compress if required

Codec Summary Codec

Bandwidth

MOS

iLBC G.711 G.729 G.723.1 G.723.2 G.726 G.726 G.729a G.728

15.2kbps 64kbps 8kbps 6.3kbps 5.3kbps 32kbps 24kbps 8kbps 16kbps

4.1 4.1 3.92 3.9 3.8 3.85 3.7 3.61

Codec Delay

Complexity

20ms Sample Size (bytes)

0.75ms 10ms 30ms

Medium High High

160 20

Notes

Most ed

Medium 10ms

Medium High

Standard PSTN is considered to have a MOS of 4 Comfort Noise - Digital based telephony in some cases introduces a small amount of noise on the call. This avoids the scenario where the listener may believe that the transmission has been lost, and therefore hangs up prematurely. Additionally reduces the effects of VAD introducing sudden change in sound level iLBC – Internet Low Bit rate Codec MOS – Mean Opinion Score. Human based rating which scores the quality of speech between 1 (poor) to 5 (excellent). http://en.wikipedia.org/wiki/Mean_opinion_score PQSM – Perceptual Speech Quality Measurement. Machine based scoring from 6.5 (poor) to 0 (excellent)

G711 Two types µ-law (North America & Japan)  A-law (Europe and reset of World)


M Morgan ©2010

Page 17 of 50

CCNA Voice IIUC (640-460) Similarities Between A-law and u-law  Both are linear approximations of logarithmic input/output relationship. 

Both are implemented using eight-bit code words (256 levels, one for each quantization interval). Eight-bit code words allow for a bit rate of 64 kilobits per second (kbps). This is calculated by multiplying the sampling rate (twice the input frequency) by the size of the code word (2 x 4 kHz x 8 bits = 64 kbps).



Both break a dynamic range into a total of 16 segments: o Eight positive and eight negative segments. o Each segment is twice the length of the preceding one. o Uniform quantization is used within each segment.



Both use a similar approach to coding the eight-bit word: o First (MSB) identifies polarity. o Bits two, three, and four identify segment. o Final four bits quantize the segment are the lower signal levels than A-law.

Differences Between A-law and u-law  Different linear approximations lead to different lengths and slopes.  The numerical assignment of the bit positions in the eight-bit code word to segments and the quantization levels within segments are different.  A-law provides a greater dynamic range than u-law.  u-law provides better signal/distortion performance for low level signals than A-law.  A-law requires 13-bits for a uniform PCM equivalent. u-law requires 14-bits for a uniform PCM equivalent.  An international connection needs to use A-law, u to A conversion is the responsibility of the ulaw country.


M Morgan ©2010

Page 18 of 50


Numbering Plans PSTN Numbering Plan ITU-T E.164   

Country Code National Destination Code Subscriber Number

Example : North American Numbering Plan (NANP) Country Code Area Code Central Office Code Station Code Example - 1 480 555 1212


M Morgan ©2010

Page 19 of 50


Phones Phone Range Lines 7906G 7911G 7914/7915/7916 7920 7921 7931 7936 7937 7940G 7941G 7941G-GE 7942G 7945G 7960G 7961G 7961G-GE 7962G 7965G 7970G 7971G-GE 7975G 7985 ATA 186 ATA 188 VG224 VG248 IP Communicator Unified Personal Communicator

1 1 14 1 1 24 1 1 2 2 2 2 2 6 6 6 6 6 8 8 8 1 2 2 24 48 8

Switch

XML Apps PoE Text Graphics Pre 802.3af

No Yes No No No Yes No No Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes No Yes No No -

Yes Yes No Yes Yes Yes No No Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes No No No No -

No No No No Yes No No No Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes No No No No -

Yes Yes No No Yes No No No Yes Yes No Yes No Yes Yes No Yes No Yes No No No No No No No -

Yes Yes No No Yes Yes No Yes No Yes Yes Yes Yes No Yes Yes Yes Yes Yes Yes Yes Yes No No No No -

Notes

Expansion Module 802.11b Wifi Phone A,B & G Wifi, PTT Conference Station Conference Station High res screen Gig Ethernet High Quality Audio High res screen High res screen Gig Ethernet High Quality Audio High res screen Colour Touch screen Colour Touch screen Colour Touch screen Video Phone Dual FXS Dual FXS Analogue Gateway :FXS Analogue Gateway :FXS Soft Phone

Expansion Module adds an additional 14 lines to a 796x and 797x phones. Up to two units can be added.

Phone Boot Process 1. 2. 3. 4. 5. 6.

Switch detects PoE capabilities and sends power if required. Phone boots software image. Switch sends the Voice VLAN info to the phone using CDP. IP Phone uses DH to get its IP address including ‘option 150’ (TFTP IP Address). Phone s TFTP server and gets configuration file. Phone s with the CME Server listed in the config file.


M Morgan ©2010

Page 20 of 50


Powering Inline Power Cisco Pre-Standard PoE – A switch will send a tone (Fast Link Pulse – FLP) down the network cable, an unpowered Cisco phone will loop the tone back to the switch. The switch then sends a maximum of 6.3 watts to the phone for it to begin powering up. The phone then sends it actual power requirements to the switch using CDP. For non Cisco phones the switch will send the full 15.4 watts. IEEE 802.3AF – The switch sends a constant DC current to the device (does not harm the device because of DC filtering), a 802.3AF device has a specific value resistor allowing the switch to detect the power requirements of the device. This standard is able to send power over Gigabit Ethernet.

Class

Allocated Power

Actual Power Used

0 1 2 3

15.4W 4.0W 7.0W 15.4W

0.44 to 12.95 0.44 – 3.84W 3.84 – 6.49W 6.49 – 12.95W

Midspan Power Power Patch – Sits between the switch and patch to inject power. Avoids cost of replacing switches for PoE switch. Power Injector – Simple power injector, no intelligence. Wall Power -PWR-CUBE-3


M Morgan ©2010

Page 21 of 50


Basic Configuration Switch configuration Mode

Description

Command

# # # #

Show all defined vlans and assigned ports Show total power available / used and port power usage Show directly connected Cisco Device information Show VTP mode and status

Show vlan Show power inline Show cdp neighors Show vtp status

Set Switch Port Trunking Mode (config-if) (config-if) (config-if) (config-if) (config-if)

Set the trunk encapsulation (ISL no used much now) Enable the trunk mode Auto mode. Will aggressively try to raise a trunk. Default Auto. Will not raise trunk but will if the other end does. Set native (untagged) Vlan

(config-if) (config-if) (config-if) (config-if)

Set access port Set the data vlan Set the voice/auxiliary vlan Set STP portfast

Switchport trunk encapsulation dot1q Switchport mode trunk Switchport mode dynamic desirable Switchport mode dynamic auto Switchport trunk native vlan vlan

Set Switch Port Access Mode Switchport mode access Switchport access vlan vlan Switchport voice vlan vlan Spanning-tree porftfast

Configure VLAN (config) (config-vlan)

Create a vlan Assign a name to the vlan

Vlan vlannumber Name name

(config-if) (config-if) (config-if)

Set automatic power mode Turn off PoE Leave power on for second after link goes down

Misc Power inline auto Power inline never Power inline delay shutdown seconds

NotesAs a guideline make the voice VALNs lower in number than data. This allows spanning tree to get the Voice vlan up quicker in the event of a network topology failure. Typically a router will have an access list to stop data and voice traffic crossing the Vlans.

Configuring DH Mode

Description

Command

# (config) (dh-config) (dh-config) (dh-config) (dh-config) (dh-config) (config)

Display DH leases Create a DH pool Define network to enable & issues addressed Set default router Set DNS server Set TFTP server address Set TFTP server name (not recommended) Set dh excluded addresses

Show ip dh binding Ip dh pool pool Network x.x.x.x /24 Default-router x.x.x.x Dns-server x.x.x.x Option 150 ip x.x.x.x Option 66 ascii tftpservername Ip dh excluded-address x.x.x.x y.y.y.y


M Morgan ©2010

Page 22 of 50

CCNA Voice IIUC (640-460) (config-if)

Set helper address for a DH server on an interface

Ip helper-address x.x.x.x

Notes

The ‘Network’ command allows the addition of a mask bit length or network mask. Otherwise is will issue the default class full subnet mask. Common practice is to include the option 150 in data VLANs as well so phones will work if plugged into the data VLAN. ‘Ip helper address’ is used to create a proxy to send a broadcast received on an interface to a unicast address. When the unicast is sent it is sent to the address specified but with a source address of the interface the broadcast was received from. This allows a DH server to identify with DH pool to assign addresses accordingly. For this to work the DH server must have a route to the network requiring DH services.

 

Configuring NTP Mode

Description

Command

# (config) (config)

Show NTP sources and status Set a time server Set a hour zone and hour difference for the time

Show ntp associations Ntp server domainname Clock timezone name x

Configure a router as an NTP Master (config) (config)

Allow other devices to get the time from device Assign an access list to restrict access

Ntp master Ntp access-group list

Stratum 0 – Atomic clock. Stratum 1 – NTP Server directly connection to a radio or atomic clock. Stratum 2 NTP Server gets its time from a stratum 1 server......


M Morgan ©2010

Page 23 of 50


CME Communications Manager Express Licensing IOS – License to run the required IOS (Voice / AdvancedEnterprise etc). Think Windows Server Licence. Feature License – License CME for a specific number of s. Think Windows CAL. Phone License – License the IP phone to interact with CME / CCM. Think Windows XP License.

CME Files While all the functionality for running voice is built into the routers IOS, Cisco provide TAR files to provide additional resources for the phone systemBasic Files – Phone loads / firmware. GUI Files – HTML web front end. XML Template Files – Allows the to edit the GUI such as only allows certain to perform certain actions. MoH Files – Music on hold. Script Files - TCL scripts for advanced functions (auto attendant, ACD etc). Miscellaneous Files – Other files such as Custom ring tones.

Installing 1.

Get the files.

2.

Place the files on a TFTP server

3.

Copy the files to the routers flash memory, either1.

Use the copy command for each file. Takes a long time. or

2.

Use the Archive command to unpack the archive on the router, quick.

Mode

Description

Command

# # #

Show all flash files and free space Think DOS... Install CME from TFTP

Show flash Dir flash: Archive tar /xtract tftp://x.x.x.x/cme..tar flash:


M Morgan ©2010

Page 24 of 50


Basic CME Configuration Mode

Description

Command

#

Show telephony-service

Basic Configuration (config) (config-telephony) (config-telephony) (config-telephony)

Go to telephone service configuration Maximum directory numbers Maximum phones on the system (up licenses purchased) Defines IP address the phones will attempt to

Telephony-service Max-dn x Max-ephones x Ip source-address x.x.x.x

(config-telephony) (config-telephony)

Disable automatically ing ephones Configure ephone-dn to ephone auto assignment

No auto-reg-ephone Auto assign x to y

(config-telephony) (config-telephony)

Allow time from the GUI Allow DN from GUI. Required for CUE

Time-webedit Dn-webedit

Auto Registration and DN assignment

‘Ip source-address’ can be set to a loopback interface if ing phones on more than one interface. The network and phones must have routes to this address.

Phone Loads / files Mode

Description

Command

# # (config) (config-telephony) (config-telephony)

Show the internal telephony service tftp files Display the contents of a text file Define a file for the TFTP server Define what firmware to load to a phone Create the configuration files

show telephony-service tftp-bindings More filename Tftp-server filename alias name Load phonemodel filename Create cnf-files

As the phone only asks for the filename, not the full path the alias element of the ‘tftp-server’ command provides the file alias. ExamplesTftp-server flash:/phone/7940-7960/P00308000500.bin alias P00308000500.bin Tftp-server flash:/phone/7940-7960/P00308000500.loads alias P00308000500.loads Tftp-server flash:/phone/7940-7960/P00308000500.sb2 alias P00308000500.sb2 Tftp-server flash:/phone/7940-7960/P00308000500.sbn alias P00308000500.sbn

To find the filename for the ‘Load’ command referencehttp://www.cisco.com/en/US/docs/voice_ip_comm/cucme/requirements/guide/cme43spc.htm


M Morgan ©2010

Page 25 of 50


Phone configuration files XMLDefault.cnf.xml – Basic phone configuration file, contains what IP address is hosting CME and firmware names to . This can be viewed using the command ‘more system:/its/vrf1/XMLDefault.cnf.xml’

Ephone-dn Represents the phone numbers. Single Line - Only able to handle on call Dual Line - Handles two simultaneous calls – allows call waiting, conferencing, consultative transfers

Mode

Description

Command

(config) (config) (config-ephone-dn) (config-ephone-dn) (config-ephone-dn)

Create a single line dn Create a dual line dn Assign a number Assign a secondary/did number Assign a name for the telephone directory

Ephone-dn tag Ephone-dn tag dual-line Number number Number number secondary number Name name

(config-ephone-dn)

Preference to use when same number assigned to many dn’s. Default is 0. Consider this the last dn in the hunt group. Don’t try to find another dn. If any /line channel on a dual line dn is used, don’t place a second call on the same dn.

Preference x

(config-ephone-dn) (config-ephone-dn)

Huntstop Huntstop channel

EPhone Represents the physical phone.

Mode # # (config) (config-ephone) (config-ephone) (config-ephone) (config-ephone) (config-ephone)

Description

Command

Show ephones trying to . Useful to find phone MAC when setting up phones Create an ephone Assign a MAC Set phone type. Not required as CME will find this out Assign a phone line with a dn Cold reset phone Warm reset phone


M Morgan ©2010

Show ephone Show ephone attempted-registrations Ephone no Mac-address xxxx.xxxx.xxxx Type phonemodel Button x:y Reset Restart

Page 26 of 50

CCNA Voice IIUC (640-460) Auto Registration & Assignment Auto Registration - By default ephones will automatically with CME, they won’t automatically be created in the running config. Disabled with the ‘No auto-reg-ephone’ telephony service command. Auto assignment – CME will automatically assign ephone-dn’s to ephones. Configured with ’Auto assign x to y’ where x is the start dn and y is the end dn.

Button command options Button assignments link a DN to a physical button on a telephone. A number of methods can be used on the assignments-

Mode From IOS Help : S B F M

Normal ring Silent ring Call waiting beep, no ring Feature Mode Monitor Mode

W O C X

Watch Mode Overlay Line (no call waiting) Overlay Line (with call waiting) Overlay Expansion / Overlay

Description Silent ring but beep on call waiting Alternate ring tone for a incoming call Creates a button which shows the status of the ephone-dn. Also acts as a speed dial button. Ideal for receptionist As monitor button but watches the whole phone assigned to the dn Allows multiple phones at the same time to ring on incoming call Allows multiple phones at the same time to ring on incoming call

Single telephone number multiple ephones Some scenarios require a single extension number to be assigned to more than one telephone, such as in a call centre environment, a number of approaches are availableSingle dn assigned multiple ephones Using ‘button x:y’ – All ephones share the one DN/line. Not good for call centre type applications, if one person receives a call all the ephones will be unable to use that DN / number for both incoming and outgoing calls. Multiple ‘ephone-dn’ using same incoming number Multiple DNs are created with the same extension number, with each DN assigned to a single ephone. As each phone has a unique DN, multiple phones can both receive and make calls using the number. Incoming calls are randomly distributed among ephones (only a single phone will ring). If required the ‘Preference’ command allows control of the phone ring sequence, e.g. to always make the phone assigned to DN 10 ring first followed by the phone assigned to DN 11 if the first phone is busy(config) # Ephone-dn 10 dual-line (config-ephone-dn) # Number 1010 (config-ephone-dn) # Preference 0 (config) # Ephone-dn 11 dual-line (config-ephone-dn) # Number 1010 (config-ephone-dn) # Preference 1


M Morgan ©2010

Page 27 of 50

CCNA Voice IIUC (640-460) There is a problem using this approach when using a dual line ephone-dn, as each DN can handle two calls, a second call to shared number could go to the second line of the DN resulting in a call waiting scenario. The ‘Huntstop’ command stops a second call hitting a dn currently in use (huntstop channel) and places it on the next dn (no huntstop) Note the last dn has doesn’t have a ‘no huntstop’ command . (config) # Ephone-dn 10 dual-line (config-ephone-dn) # Number 1010 (config-ephone-dn) # Preference 0 (config-ephone-dn) # Huntstop channel (config-ephone-dn) # No huntstop (config) # Ephone-dn 11 dual-line (config-ephone-dn) # Number 1010 (config-ephone-dn) # Preference 1 (config-ephone-dn) # Huntstop channel

See the Cisco Website for more information. Overlay Line buttons Allows an incoming call to ring multiple phones simultaneously i.e. the incoming call will be overlayed to multiple ephones. (config) # Ephone-dn 10 (config-ephone-dn) # Number 1010 (config-ephone-dn) # Preference 0 (config-ephone-dn) # No huntstop (config-ephone-dn) # Exit (config) #Ephone-dn 11 (config-ephone-dn) # Number 1010 (config-ephone-dn) # Preference 1 (config-ephone-dn) # Exit (config) #Ephone 8 (config-ephone) # Button 1o10,11 (config-ephone) # Exit (config) #Ephone 8 (config-ephone) # Button 1o10,11 (config-ephone) # Exit

In this example multiple DNs are created allowing the shared number 1010 to be used multiple times for incoming and outgoing calls. The DNs are then overlayed to the telephone buttons, in effect a phone button will have multiple assigned DNs. ‘C’ Overlay Line (with call waiting). If the buttons are configured with ‘C’ instead of ‘O’, the first call will ring ephone 8 & 9. A second call will ring the inactive phone but the active will receive a call waiting beep. Although the ephone-dn’s are single line and don’t call waiting, the second call will come in on the inactive dn, dn 11 which will generate the call waiting beep.. Recommendation is to not use dual lines with ‘O’ and ‘C’.


M Morgan ©2010

Page 28 of 50


Additional functions Voice network Directory (Local Directory on phone) Mode

Description

Command

(config) (config-ephone-dn) (config) (config--dn) (config) (config-telephony) (config-telephony) (config-telephony)

Select Cisco ephone dn Assign a name for the telephone directory Select SIP dn (for attached sip phones) Assign a name for the telephone directory Select telephony service config mode Set directory sort order (default) Set directory sort order Create an entry (for non dn entries – up to 100)

ephone-dn dn Name name voice dn dn Name name Telephony-service Directory first-name-first Directory last-name-first Directory entry id number name name

Call forwarding call forward ‘CFwdAll’ phone soft key allows a to enter an extension to forward all calls to. System call forward A DN can be configured with the command ‘Call-forward all XXX’ & ‘Call-forward busy XXX’ to define where to forward calls.

Configuring-

Mode

Description

Command

(config-ephone-dn) (config-ephone-dn) (config-ephone-dn) (config-ephone-dn) (config-ephone-dn) (config--dn) (config--dn) (config-telephony)

Forward all calls Set forward when phone busy Set forward when phone not answered Forward call on activated night service Restrict length of a forward number Set forward when phone busy Set forward when phone not answered Set valid call forward destinations

Call-forward all number Call-forward busy number Call-forward noan number timeout seconds Call-forward night-service number Call-forward max-length length call-forward b2bua busy number call-forward b2bua noan number timeout seconds Call-forward pattern pattern

‘Call-forward pattern pattern’ and ‘Call-forward max-length length’ are used to control what number calls can be forwarded to, this helps avoid call toll fraud. H.450.3 - Allows the original caller and the recipient of the forward to handle the transferred call directly

rather than via the intermediate party handling the media stream (call hair-pinning). This is enabled when a ‘call-forward pattern pattern’ is specified.

Call transfer Consulted transfer – presses the ‘Transfer’ soft key and dials the number to be transferred to. The then consults the transfer recipient informing them of the call. The ‘Transfer’ soft key is then pressed to connect the two parties. This is the default. http://www.studyshorts.co.uk

M Morgan ©2010

Page 29 of 50

CCNA Voice IIUC (640-460) Blind transfer – The call is transferred as soon as the transfer number is entered. H.450.2 – Allows the original caller and the recipient of the transfer to handle the transferred call directly rather than via the intermediate party handling the media stream (call hair-pinning).

Mode

Description

Command

(config-telephony) (config-telephony) (config-telephony) (config-telephony)

Sets blind transfer system using H.450.2 Sets consult transfer system using H.450.2 Sets consult transfer system using proprietary method Sets the pattern for valid transfers

Transfer-system full-blind Transfer-system full-consult Transfer-system local-consult Transfer-pattern pattern

By default call transfers can only take place between phones in the system. Setting a transfer pattern allows calls to be transferred to external numbers. This is means to reduce the possibility of toll fraud.

Call Park Example config to create a park slot(config) # ephone-dn 20 (config-ephone-dn) # number 399 (config-ephone-dn) # park-slot

park-slot timeout command Basic form(config-ephone-dn) # park-slot timeout x limit y

The person who sent the call to the park slot is notified every x seconds for a maximum of y times before taking action. Notify a second extension of the parked call(config-ephone-dn) # park-slot timeout x limit y notify number

Recall the parked call back to the originator(config-ephone-dn) # park-slot timeout x limit y recall

Transfer the timed out parked call to an extension. If that extension is busy transfer to the alternate number(config-ephone-dn) # park-slot timeout x limit y transfer number alternate number

Park Slot reservation It is possible to assign a reservation group to a park slot. Only ephones configured with the same reservation group can pick up the parked call. (config) # ephone-dn 30 (config-ephone-dn) # park-slot reservation-group 1 timeout 10 limit 3 transfer 700


M Morgan ©2010

Page 30 of 50

CCNA Voice IIUC (640-460) (config) # ephone 1 (config-ephone-dn) # park reservation-group 1

NotesOnce a park slot has been created the ‘Park’ button becomes available on the phones. To pick the call up simply call the parked call number or press this ‘PickUp’ softkey then dial the call park no. Additionally the person who parked the call can pick up the call by pressing ‘PickUp’ soft key then press the * key.

Call Pickup Directed Pickup – Pressing the Pickup button results in the phone sounding a dial tone waiting for the to enter the extension number of a ringing phone to pickup. Local Group Pickup – Pressing the GPickup button picks up a ringing phone in the same pickup group. Other Group Pickup - Pressing the GPickup button results in the phone sounding a dial tone waiting for the to enter the group number a ringing phone to pickup. To assign a dn to a group use the command(config-ephone-dn) # pickup-group xxxx

NotesThe GPickUp softkey functions differently depending on the call pickup configuration in CME. If there is only one group configured in CME, pressing the GPickUp button automatically answers the call from your own group number. You will not hear a second dial tone and you do not need to dial an asterisk to signify your own group, because only one group is defined. Once you have configured multiple groups in CME, you will hear a second dial tone after pressing the GPickUp softkey, at which point you can dial either an asterisk for the local group or another group number. Directed Pickup can be disabled by entering ‘no service directed-pickup’ from telephony service configuration mode.

Intercom A two way communication channel using speaker phone. When a presses the button assigned to the intercom the other phone will automatically answer using speaker phone but with the microphone muted in case the other person is saying something secretive. (config) # ephone-dn 20 (config-ephone-dn) # number A100 (config-ephone-dn) # intercom A101 label “Manager”


M Morgan ©2010

Page 31 of 50

CCNA Voice IIUC (640-460) (config) # ephone-dn 21 (config-ephone-dn) # number A101 (config-ephone-dn) # intercom A100 label “Assistant” (config) # ephone 3 (config-ephone) # button 2:20 (config) # ephone 4 (config-ephone) # button 2:21

Further options for the Intercom commandBarge-in – the intercom will force all other calls into the HOLD state and connect tyhe intercom call No-auto-answer – Disable the intercom auto answer No-mute – Disable the auto mute.

Paging A one way speakerphone based announcement. There are two methods, unicast or multicast. As unicast requires a single stream per page group member the group is limited to a maximum of 10 . If using multicast the network must be capable/configurable of ing multicast streams. A phone can only be a member of one paging group but a paging group can be a member of another parent paging group. Create a paging group(config) # ephone-dn 25 (config-ephone-dn) # number 3000 (config-ephone-dn) # paging (config-ephone-dn) # paging ip 239.4.3.4 port 200 (config-ephone-dn) # paging group dnlist

- Unicast paging or - Multicast paging (cannot use 224.) - Associate a child paging group

Assign a phone to the paging group(config) # ephone 3 (config-ephone) # paging-dn 25

After hours call blocking Ability to block specified number outside of hours.

Mode

Description

Command

(config-telephony) (config-telephony) (config-telephony) (config-telephony)

Define outside of hours on particular day O of H on day between start & endtime Define blocked number pattern (up to 100) Permanent block (24-7) - no exceptions

After-hours date month dayno After-hours day day starttime endtime After-hours block pattern no pattern After-hours block pattern no pattern 7-24


M Morgan ©2010

Page 32 of 50

CCNA Voice IIUC (640-460) (config-ephone) (config-ephone) (config-telephony)

Exempt phone from out of hours block Set a pin for temporarily removing blocks Enable for pins. Parameters not required

After-hours exempt Pin xxxx timeout mins clear time

ExampleAfter-hours day mon 17:00 8:00 After-hours day tue 17:00 8:00 After-hours day wed 17:00 8:00 After-hours day thu 17:00 8:00 After-hours day fri 17:00 8:00 After-hours date dec 25 00:00 00:00 After-hours block pattern 1 90..........

- Block all non local calls

Music on Hold Stream a wav or au files in the routers flash memory using unicast (up-to 10 like paging) or multicast. Example(config-telephony) # moh music.wav (config-telephony) # multicast moh 239.4.3.2 port 2100

- Multicast if required

CME GUI Provided the GIU Files have been installed on the router, the HTML front end can be enabled using the following commands(config) # ip http server - Enable http server (config) # ip http secure-server - Enable https server (config) # ip http path flash:/gui - Set the location of the gui files (config) # ip http authentication local - Set local authentication database Additional commands to control the front end(config-telephony) # web system name mike secret (config-telephony) # dn-webedit (config-telephony) # time-webedit

- Enable changing dn through the gui - Enable changing time through the gui

To use simply browse to ‘http://x.x.x.x/ccme.html’


M Morgan ©2010

Page 33 of 50


Gateways A Gateway is a link from the VoIP telephone system (CME) to a traditional PBS / PSTN or another VoIP system. A number of gateway types can be employed-

Analogue gateways – Single call per port FXO (Foreign Exchange Office) Acts as an analogue telephone socket, connecting to the PSTN exchange / Telco central office. These facilitate Analogue trunks to the telco. FXS (Foreign Exchange Station) Acts as an analogue PSTN exchange allowing analogue stations / devices (phones, faxes etc) to be connected to the CME infrastructure. Typical devices for FXS ports - VIC2-2FXS / ATA186 / ATA188 / VG224 / VG248 E&M (Ear & Mouth / Earth & Magneto) Specific analogue module purely for trunking. Typically used to connect two PBX systems together

Digital gateways – Multiple calls per port T1 & E1 CAS Example cards are VWIC-MFT-E1 / VWIC-1MFT-T1, typically used to connect to Telcos. T1 & E1 CCS (Primary Rate Interface PRI) Example cards are VWIC-MFT-E1 / VWIC-1MFT-T1 Basic Rate Interface (BRI)

Dial Peers A Dial peer defines how a call enters / leaves CME, there are two types– POTS Dial Peer connects to a traditional voice system, the call is sent out a voice port where the voice port is an FXO, PRI etc. VoIP Dial Peer IP Based, calls are sent to an IP address, another CME system or SIP server can be used.

Mode

Description

Command

# # # # #

Show the summary and status of all voice ports Show the summary and status of all dial peers Debug the dial peer process Show all voice calls

Show voice port summary Show dial-peer voice summary Debug voip dialpeer Show voice call summary show call active voice

Create POTS FXS Dial Peer extension (config) (config-dial-peer) (config-dial-peer)

Create a dial peer Define the numbers to assign to the port Assign a physical port to the dial peer


M Morgan ©2010

Dial-peer voice tag pots Destination-pattern number Port port

Page 34 of 50

CCNA Voice IIUC (640-460) (config-dial-peer)

Description description

Create VoIP Outbound Dial Peer (config) (config-dial-peer) (config-dial-peer) (config-dial-peer)

Create a dial peer Set the destination pattern Send matching calls to the remove voip server

Dial-peer voice tag voip Destination-pattern pattern Session target ipv4:x.x.x.x Description description

Create a T1/E1 Outbound Dial Peer (config) (config-dial-peer) (config-dial-peer) (config-dial-peer)

Create a dial peer Set the destination pattern

Dial-peer voice tab pots Destination-pattern pattern Description description Port x/x:z

Set the destination port

Destination-patterns When sending a call out through a dial peer a destination pattern must be created to define what calls should be sent through the dial peer. Various options are available to define the pattern as below-

Wildcard Meaning . + []

A single digit One or more instances of Range of digits

T

Anything

Example Matches 50. 1+ [1-3]111 [14-6]11 [6789].. 9T

500, 501 ... 509 11, 111, 11111111 1111, 2111, 3111 111, 411, 511, 611 6xx, 7xx, 8xx, 9xx Anything starting with a 9. Wait for inter-digit time out to dial

Call Legs When a call enters or leaves CME, a call leg is required, so for example if a call comes in on an FXO port a call leg will be created for that call. An extreme example could be where a call comes in to CME via an FXO port, CME then sends the call out to another CME system via an IP trunk then finally the call is sent out an FXS port. The legs in this example would beLeg 1 – Telco exchange to FXO port on voice switch (In to CME ‘A’) Leg 2 – Voice switch to IP trunk over a Wan (Out of CME ‘A’) Leg 3 – IP Wan trunk to voice switch (In to CME ‘B’) Leg 4 – Voice switch FXS to analogue station (Out of CME ‘B’)

A call leg is basically a matching dial peer, as seen above to make an outbound call from CME a dial peer is required to define the target/port and the destination pattern. Inbound calls ideally require a matching dial peer as well, dial peers will be matched using the following criteria and order1.

Matched the dialled number (DNIS) using the ‘incoming called-number’ dial peer configuration command.


M Morgan ©2010

Page 35 of 50

CCNA Voice IIUC (640-460) 2.

Match the caller-id information (ANI) using the ‘answer-address’ dial peer configuration command.

3.

Match the caller-id information (ANI) using the ‘destination-pattern’ dial peer configuration command.

4.

Match an incoming pots dial peer by using the ‘port’ dial peer configuration command.

5.

If no match has been found using the previous four methods, use dial peer 0.

Dial Peer 0 An implicit dial peer for all inbound calls with no matching dial peer. While this functions fine there are benefits to have an explicitly defined matching dial peer for incoming calls as additional options can be defined such as valid codecs, disabling vad etc.


M Morgan ©2010

Page 36 of 50


Digit Manipulation POTS Auto stripping Pots dial peers automatically strip any explicitly defined number from the destination pattern before sending the call. Examples Destination-pattern 9[2-9]....... The 9 will be stripped Destination-pattern 9[469]11 The 9 & 11 will be stripped Destination-pattern 91[2-9]....... The 9 & 1 will be stripped Destination-pattern 9011T The 9011 will be stripped

Prefix Add the prefix to the remaining dialled digits. Forward-digits forward number of right most digits, including any digits automatically stripped. Digit-strip Default action. Turn off auto stripping using the command no digit-strip. Num-exp <match> <set> Effectively search and replace. Global config command.

Example PSTN Failover Example - sending calls for 6... to a remote phone system using an IP trunk. If the trunk fails the calls will be sent out a POTS voice port to numbers relating to the DID numbers of the extensions, eg 6001 will get sent to the PSTN number 02920116001 which the receiving phone system will forward to the extension 6001. (config) # Dial-peer voice 6000 voip (config-dial-peer) # Destination-pattern 6... (config-dial-peer) # Session-target ipv4:10.1.1.2 (config-dial-peer) # Preference 0 (config) # Dial-peer voice 6001 pots (config-dial-peer) # Destination-pattern 6... (config-dial-peer) # No digit-strip (config-dial-peer) # Prefix 0292011 (config-dial-peer) # Port 1/0:15 (config-dial-peer) # Preference 1

Example 0 for operator (config) # Num-exp 0 5000


M Morgan ©2010

Page 37 of 50


Configuring Voice Ports Configuring VWIC T1 & E1 cards Mode

Description

Commands

(config)

Select interface

Controller interface

(config-controller) (config-controller) (config-controller)

Set framing (esf most common) Set coding (b8zs used with esf) Configure CAS

(config) (config-controller)

Set the ISDN switch type Configure CCS

CAS Framing <sf / esf> Linecoding Ds0-group groupnumber timeslots x-y type signalling

CCS Isdn switch-type ..... Pri-group timeslots x-y

Examples Configure all 24 channels of a T1 line using loop start (config) # controller t1 1/0 (config-controller) # Ds0-group 5 timeslots 1-24 type fxo-loop-start (config) # Dial-peer voice 6001 pots (config-dial-peer) # Destination-pattern 6... (config-dial-peer) # No digit-strip (config-dial-peer) # Prefix 0292011 (config-dial-peer) # Port 1/0:5 (config-dial-peer) # Preference 1

- Same as tag number

Configure PRI CCS on an E1 line (config) # controller E1 0/1/0 (config-controller) # pri-group timeslots 1-6

All calls are directed through 1/0:15 for E1 and 23 for T1

Configuring FXO/FXS ports FXS

Mode

Description

Command

(config-voiceport)

Set start method. Loopstart is default. Used when trunking to a pbx Set the dial tone. Also changes the ring cadence accordingly Change the ringing AC frequency Set the ring pattern Set custom ring cadence

Signal tone Ring frequency <25 / 50> Ring cadence patternxx Ring cadence x y z . . . . . Busyout Station-id name Timeouts .....

(config-voiceport) (config-voiceport) (config-voiceport) (config-voiceport) (config-voiceport) (config-voiceport)

Set the caller ID Name


M Morgan ©2010

Page 38 of 50


FXO

Mode

Description

Command

(config-voiceport)

Set start method. Loopstart is default. Used when trunking to a pbx Set the dialling signalling method Length of time before the router answers the call ?????

Signal

(config-voiceport) (config-voiceport)


M Morgan ©2010

Dial-type Ring number <1 – 10>

Page 39 of 50


Unity Unity Range Max Mailboxes Voice Mail Integrated Messaging Unified Messaging Auto Attendant Platform PBX / TDM Redundancy

Unity Express

Unity Connection

Unity

250 Yes Yes No Yes Linux router based No No

7500 Yes Yes No Yes Windows / Linux Server No No

7500 per server Yes Yes Yes Yes Windows Server Yes Yes

Unity Express AIM-CUE Max Mailboxes Voice Ports Installation Storage (hrs) Concurrent languages

50 6 Internal 14 2

NM-CUE

N-CUE-EC

NME-CUE

100 8 NM Slot 100 5

250 16 NM Slot 300 5

250 24 NM Slot 300 5

CUE Features Voicemail ( Mailbox). A /subscriber has his/her own personal mailbox. A pin is required to . Voicemail (General Delivery Mailbox) is a shared mailbox accessible by many subscribers. Subscribers must be made a member of the GDM to access it and will be prompted to access it when checking their own personal mailbox. A pin is not required. IVR (Interactive Voice Prompt) is a system where the system the phone system plays a prompt then waits for a to respond. Typical uses are an auto attendant and bank automated balance enquiry. Auto Attendant allows s to direct themselves to the correct person eg ‘Press 1 for Sales, 2 for s’. Two scripts are provided with the system ‘Auto Attendant Script’ & ‘Auto Attendant Simple Script’. By default the following greetings are available ‘Welcome prompt’, ‘Business Open prompt’, ‘Business Closed prompt’ & ‘Holiday prompt’. istration via Telephone (AVT) allows an to record greetings and prompts. Backup and restore functionality is provided making use of an FTP server. This requires access to the web gui. Message Waiting indicator alerts the there is a message waiting by flashing a red light and displaying an envelope on the phone display. http://www.studyshorts.co.uk M Morgan ©2010 Page 40 of 50

CCNA Voice IIUC (640-460) Message Notifications are additional methods of alerting the there is a message. The notification can be to ring a phone or send an email.

Troubleshooting From IOSShow interface service-engine 1/0 Service-module service-engine 1/0 status Show dial-peer voice Debug ephone mwi

- Should be in a steady state

From CUE Trace Show trace buffer

Setup Process 1. 2. 3. 4. 5.

Configure IOS service-engine and service-module for IP connectivity. Create SIP dial peer for CUE. Create MWI notification ephone dn’s. Perform initial config – domain name, hostname, NTP servers & credentials. Run Initialisation Wizard (import s, MWI methods, voicemail access number, istration by telephone number etc).

Initial Engine Setup Once installed a ‘service-engine x/y’ interface appears in the routers config, this is the routers interface to the Unity Express module. There are two methods of asg it an IP addressMethod 1 (config) # interface service-engine0/1 (config-if) # ip address 192.168.100.1 255.255.255.252 (config-if) # service-module ip address 192.168.100.2 255.255.255.252 (config-if) # service-module ip default-gateway 192.168.100.1 (config-if) # no shutdown

Method 2 (config ) # interface Loopback1 (config-if) # ip address 192.168.1.1 255.255.255.0 (config) # interface Service-engine0/1 (config-if) # ip unnumbered Loopback1 (config-if) # service-module ip address y.y.y.y y.y.y.y (config-if) # no shutdown (config) # ip route y.y.y.y Loopback1 (config) # Ip route 192.168.1.2 255.255.255.255 Service-engine0/1


M Morgan ©2010

Page 41 of 50


Controlling / Connecting to the module # service-module service –engine0/1 sessions # service-module service –engine0/1 reload # service-module service –engine0/1 reset # service-module service –engine0/1 shutdown # service-module service –engine0/1 status

- Connect to the module using the specific engine - Reload the module - Reset the module - Shutdown the module (before powering off router) - Show the status of the CUE module

Initial Configuration of the Module # service-module service –engine0/1 sessions > enable # offline # restore factory default

- Initiates a telnet connection to the module - enter privileged mode - Take module offline

Once restored the unit will reboot and show the prompt‘Do you wish to start configuration now (y,n)?’ Enter Host Name? Enter Domain Name? Would you like to use DNS for CUE (y,n)? Enter IP Address of the Primary NTP Server? Enter IP Address of the Secondary Server? Please Identify a location so that time zone rules can be set correctly? 1) Africa, 2) Americas ....... Please select a country? 1) Anguilla, 2) Antigua & Barbados ...... Please select one of the following time zones regions. 1) Eastern Time, 2) Eastern Time – Michigan.... ** Is the above information OK? 1) Yes, 2) No Waiting xxx .....

After booting it prompts for the details Enter ID: Enter for :

** US Additional Option

Upgrading CUE CUE # software install clean url ftp://x.x.x.x/cue-vm-k9.nm-aim.4.2.1.pkg * Language Installation Menu : 1 ITA, 2 ESP ........ ** # enter the number for the language to sellec one R # - remove the language for given # I # - more information about the language for a given ‘ x- Done with language selection Enter Command:


M Morgan ©2010

Page 42 of 50

CCNA Voice IIUC (640-460) CUE # software install clean url ftp://x.x.x.x/license

*CUE uses a name and of ‘anonymous’. Ensure the FTP server has this setup. ** Corresponding language file must be ed as well.

NOTE an upgrade can be performed using the command software upgrade only from version 2.3.4

Configure CME to access CUE CME communicates with the CUE using a SIP dial-peer.(config) # dial-peer voice 700 voip (config-dial-peer) # Destination-pattern 7.. (config-dial-peer) # session target ipv4:192.68.100.2 (config-dial-peer) # session protocol sipv2 (config-dial-peer) # dtmf-relay sip-notify (config-dial-peer) # codec g711ulaw (config-dial-peer) # no vad

- out of band DTMF - Essential

Create the MWI dns’(config) # ephone-dn 120 (config-ephone-dn) # number 1999... (config-ephone-dn) # mwi on (config) # ephone-dn 120 (config-ephone-dn) # number 1998... (config-ephone-dn) # mwi off

The CUE module will call 1999<ext> to turn the MWI on for this dn. The CUE module will call 1998<ext> to turn the MWI on for this dn. # Debug ephone mwi Trace debugging


M Morgan ©2010

Page 43 of 50


CUE Web Interface


M Morgan ©2010

Page 44 of 50


Initialisation Wizard

The Web name and allows the CUE Module to get the current dn config from CME and ister it.


M Morgan ©2010

Page 45 of 50


– Web Interface (GUI) PIN – Telephone interface(TUI)

Voice Mail Number – This configure the CUE voicemail number and configure the phones message button to this number. Auto Attendant Access Number- Configures the CUE AA number. http://www.studyshorts.co.uk

M Morgan ©2010

Page 46 of 50

CCNA Voice IIUC (640-460) via Telephone Number (ATN) – Allows istering the CUE using a telephone number (message prompt recording etc). SIP MWI Notification Mechanism – Other options are ‘Subscribe – Notify’ .....


M Morgan ©2010

Page 47 of 50


Smart Business Communication System UC520 – Central point of the IP based system. Provides routing, security, VPN (10 s), call processing for 8-48 phones, voicemail & auto-attendant. This is based on CUCME 4.2 and CUE 3.1 for voice mail CE520 – Catalyst Express Switch family Cisco 521 – Wireless Express Access Point. This can operate in either standalone mode (mode one) or Controller based mode (mode two). CCA can manage up to three independent access points. Cisco 526 – Wireless Express Mobility Controller. Can control up to 6 Cisco 521 Access Points. CCA can control two controllers allowing for up to 12 AP in a single SBCS deployment. CCA – Cisco Configuration Assistant, the configuration tool for SBC devices. Default name / ‘cisco’ & ‘cisco’

Typical UC520 Models s Music on Hold 10/100 PoE LAN Expansion 10/100 WAN 10/100 FXS FXO BRI T1 / E1 VWIC Integrated AP

UC520

UC520 Model

UC520

8 or 16 1.5u desktop 3.5mm Jack 8 (Max 80 watt) 1

8 or 16 3.5mm Jack 8 (Max 80 watt) 1

24,32 or 48 2u rack 3.5mm Jack 8 (Max 80 watt) 1

1 4 4 0 0 1 Yes

1 4 0 2 0 1 Yes

1 4 4 4 1 1 No

Typical CE520 Models CE520-8PC 10/100 10/100 PoE 10/100/1000

8 2

CE520-24TT

CE520-24LC

CE520-24PC CE520G-24T

24

20 4 2

24 2

2

24 + 2

The UC520 has the following default configuration    

Data Lan : 192.168.10.0 / 24 VLAN1 Voice Lan : 10.1.1.0 / 24 VLAN100 Telephone Ext length : 3 Out of the box Extensions : 201 – xxx http://www.studyshorts.co.uk

M Morgan ©2010

Page 48 of 50


CCA Communities CCA can discover devices using three methods FQDN  IP Address  Subnet search

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


M Morgan ©2010

Page 49 of 50


Additional Resources The Techexams Forumshttp://www.techexams.net/forums/ccna-voice/

Cisco Communications Manager Express Web sitehttp://www.cisco.com/en/US/products/sw/voicesw/ps4625/tsd_products__series_home.html

Various LABs for Cisco certificationshttp://configurethenetwork.com/


M Morgan ©2010

Page 50 of 50

Cisco Ccna Voice.pdf 4q6p5z

Overview 4q3b3c

More details 26j3b

Related Documents 171j1w

Cisco Ccna (bastante Completo) 42106j

Pe Ccna Cisco V5.0 2a6k5k

Manual Cisco Ccna 1 294e6e

Comandos Cisco Ccna Security 103d3r

Cisco Ccna 200-120 1x476p

Cisco Ccna Voice.pdf 4q6p5z

More Documents from "Satyapriya Panigrahi" 312gn

Cisco Ccna Voice.pdf 4q6p5z

006 - Case Study Summary - Heinz Ketchup 73643o

Crm - Lesson 04 - Types Of Customers And Customer Value 5o3r25

Ne-yo - Let Me Love You Lyrics 6r605r

Electronics Fundamentals And Applications_d. Chattopadhyay And P. C. Rakshit 611e70

Business Ethics - A Case Study On Nike 2p5t1a