cisco ccna voice
DESCRIPTION
44TRANSCRIPT
Legal notice and disclaimer .......................................................................................................................... 5 Introduction .................................................................................................................................................. 6
Definitions ................................................................................................................................................ 6
Well Known Ports ..................................................................................................................................... 6
Miscellaneous commands ........................................................................................................................ 7
POTS Technologies ....................................................................................................................................... 8
Analogue Connections .............................................................................................................................. 8 PSTN Signalling ......................................................................................................................................... 8
E1 / T1 Signalling ...................................................................................................................................... 9
IP Voice Technologies ................................................................................................................................. 11
Cisco Voice Infrastructure Model ........................................................................................................... 11
Signalling ................................................................................................................................................. 12 IP Transport ............................................................................................................................................ 13
IP Overhead ............................................................................................................................................ 13
Compressed RTP ..................................................................................................................................... 14
Problems with Digital Voice .................................................................................................................... 14 Causes of Delay ....................................................................................................................................... 14 QoS ......................................................................................................................................................... 14
AutoQoS .................................................................................................................................................. 15 MQC – Modular QoS CLI ......................................................................................................................... 15
Analogue to Digital Conversion / Codecs ................................................................................................... 17
Conversion .............................................................................................................................................. 17
Codec Summary ...................................................................................................................................... 17 G711 ....................................................................................................................................................... 17
Numbering Plans ........................................................................................................................................ 19
PSTN Numbering Plan ............................................................................................................................. 19
Phones ........................................................................................................................................................ 20
Phone Range ........................................................................................................................................... 20 Phone Boot Process ................................................................................................................................ 20 Powering ................................................................................................................................................. 21
Basic Configuration ..................................................................................................................................... 22
Switch configuration ............................................................................................................................... 22
Configuring DHCP ................................................................................................................................... 22
Configuring NTP ...................................................................................................................................... 23
CME Communications Manager Express .................................................................................................... 24 Licensing ................................................................................................................................................. 24
CME Files ................................................................................................................................................ 24
Installing ................................................................................................................................................. 24
Basic CME Configuration ........................................................................................................................ 25
Phone Loads / files ................................................................................................................................. 25
Phone configuration files ........................................................................................................................ 26 Ephone-dn .............................................................................................................................................. 26
EPhone .................................................................................................................................................... 26
Additional functions ................................................................................................................................... 29
Voice network Directory (Local Directory on phone) ............................................................................. 29
Call forwarding ....................................................................................................................................... 29
Call transfer ............................................................................................................................................ 29
Call Park .................................................................................................................................................. 30
Call Pickup ............................................................................................................................................... 31 Intercom ................................................................................................................................................. 31
Paging ..................................................................................................................................................... 32
After hours call blocking ......................................................................................................................... 32
Music on Hold ......................................................................................................................................... 33 CME GUI .................................................................................................................................................. 33
Gateways .................................................................................................................................................... 34
Analogue gateways – Single call per port ............................................................................................... 34
Digital gateways – Multiple calls per port .............................................................................................. 34
Dial Peers ................................................................................................................................................ 34
Call Legs .................................................................................................................................................. 35
Digit Manipulation ...................................................................................................................................... 37
POTS Auto stripping ................................................................................................................................ 37 Example PSTN Failover ........................................................................................................................... 37
Example 0 for operator........................................................................................................................... 37
Configuring Voice Ports .............................................................................................................................. 38
Configuring VWIC T1 & E1 cards ............................................................................................................. 38 Configuring FXO/FXS ports ..................................................................................................................... 38
Unity ........................................................................................................................................................... 40
Unity Range ............................................................................................................................................ 40
Unity Express .......................................................................................................................................... 40
CUE Features .......................................................................................................................................... 40
Troubleshooting ..................................................................................................................................... 41
Setup Process ......................................................................................................................................... 41
Initial Engine Setup ................................................................................................................................. 41 Controlling / Connecting to the module ................................................................................................. 42 Initial Configuration of the Module ........................................................................................................ 42
Upgrading CUE ........................................................................................................................................ 42 Configure CME to access CUE ................................................................................................................. 43 CUE Web Interface ................................................................................................................................. 44
Initialisation Wizard ................................................................................................................................ 45
Smart Business Communication System .................................................................................................... 48
Typical UC520 Models ............................................................................................................................ 48
Typical CE520 Models ............................................................................................................................. 48
CCA Communities ................................................................................................................................... 49
Cisco Configuration Assistant Tabs ......................................................................................................... 49
Additional Resources .................................................................................................................................. 50
Legal notice and disclaimer
Version 1.0
Copyright © 2010 Michael Morgan.
All rights reserved. Any redistribution or reproduction of part or all of the contents in any form is prohibited other than printing for personal use. This publication may be used free of charge, selling without prior written consent prohibited. You may not, except with our express written permission, host, distribute, or commercially exploit the content. If this publication is not obtained from http://www.caerffili.co.uk/ or http://www.studyshorts.co.uk/ the publication held is considered a pirated copy and must be destroyed immediately.
StudyShorts guides are intended to provide enough information for last minute exam preparation and reference, and are not a substitute for other training material. They were prepared to assist my studies and passing the associated exam and as such may contain errors and some facts may have been summarised or removed.
Introduction
Term Definition
FXO Foreign Exchange Office – Connects to a Telco central office
FXS Foreign Exchange Station – Connects to a local analogue phone or a fax CO. Telco Central Office
Key Switch
Typically uses analogue PSTN connections, uses shared lines between phones and limited feature sets. Phones tend to have line buttons matching the incoming PSTN lines rather than extension numbers
PBX Private Branch Exchange - Typically uses digital PSTN trunks, provides unique telephone extensions and have a large feature set
Local call A call between to local ports
Off net call A call terminated outside of a local port (PSTN)
DNIS Dialed Number Identification Service. A service provider by the Telco to signal the number dialled by the calling party (Direct Inward Dial)
ANI Automatic Number Identification. Signals the telephone number of the calling party (Caller ID)
Integrated Messaging A subscriber can access both an email box and a voice mail box using a single client
Unified Messaging A subscribers can access both email and voice mail from a single mail box
VAD Voice Activity Detection. Allows the phone system to reduce / stop sending packets during silent periods of a voice call resulting in a bandwidth saving of about 35%
H.450 Avoids hair-pinning forwarded and Transferred calls
TDM Time Division Multiplexing
DS0 A single timeslot / channel. Carries 64kb/s T1 1.544mbps. 1.536mbps actual data, .008mbps framing. 24 x DS0 channels. E1 2.048mbps - 32 DS0 channels CAS Channel Associated Signalling. Signalling is placed in
data carrying DS0 channels. Typically called Robbed Bit Signalling
CCS Common Channel Signalling. A dedicated DS0 timeslot is used for signalling. Commonly called Primary Rate ISDN
ITU-T International Telecommunication Union, Telecommunication Standardization Sector
IETF Internet Engineering Task Force RTP Real-time Transport Protocol. Carries the media stream (even UDP port) RTCP Real-time Transport Control Protocol. Carries statistic information (odd UDP
Protocol Port IP FTP 20, 21 TCP
SHH 22 TCP
Definitions
Well Known Ports
Telnet 23 TCP
SMTP 25 DNS 53 DHCP / BOOTP 67 TFTP 69 NEWS 119 NTP 123 SNMP 161, 162 UDP
Mode Description Command # Show layer 1 & 2 info on all
interfaces Show interfaces
# As above but on specific interface
Show interfaces interface
# Show layer 3 info Show ip interfaces
# As above but on specific interface Show ip interfaces interface
# Show brief interface status Show ip interface brief
# Clear all counters on one or all interfaces
Clear counters
(config) Turn off domain lookups No ip domain-lookup Telnet / Session Management
# Show open sessions from this router Show sessions
# Show open sessions to this router Show users
# Kills one of the open sessions from this router
disconnect
# Kills one of the open sessions to this router
Clear line <x>
(config-line) Timeout on the particular line connection
Exec-timeout minutes seconds
Logging & Debugging
# Redirect status messages to the current session
Terminal monitor
# Turn off all debugging u all / undebug all / no debug all # Show log buffer memory Show logging
Lower Limit (Hz) Upper Limit (Hz) Human Ear 20 20000
Human Speech 200 9000 Telephone Channel 300 3400
Miscellaneous commands
Frequencies of Audio Signals
Nyquist Theorem – Frequency sample must be twice the maximum frequency to accurately reconstruct the original wave form.
POTS Technologies
Analogue Connections
Two connections-
Ground / Tip – 0v Battery / Ring – -48v
PSTN Signalling
Signalling Ground Start – The station/PBX will ground both ring and tip to request a dial tone. Loop Start –When a phone is on hook the loop is open, when taken off hook the station will close the loop to the exchange to request a dial tone. Typically used in home environments as this is susceptible to glare.
Glare – If an incoming call happens at the same time as an outgoing line is requested in a PBX environment, they can become connected causing confusion to the outgoing caller.
Supervisory Signalling
On-hook – When the phone is on-hook, the connection between the tip and ring wires is broken and no electrical signal passes between them.
Off-hook – When the phone is off-hook, the phone connects the tip and ring wires, completing the circuit and allowing electrical signal to pass.
Ringing – To cause an analogue phone to ring, the phone company sends an alternating current (AC).
Informational Signalling Dial tone – Indicates the phone company is ready to receive digits Busy – Indicates the remote phone is already in use Ringback – Indicates the remote phone is currently ringing Congestion – Indicates the long-distance telephone network is not able to complete the call Reorder – Indicates the local telephone company is not able to complete the call Receiver off-hook – Indicates the local receiver has been off-hook for an extended period of time
No such number –Indicates the dialed number is invalid Confirmation – Indicates the telephone company is attempting to complete the call
Address Signalling
Frame 1 1st DS0 2nd DS0 3rd DS0 ... 24th DS0
... ... ... ... ... ...
Frame 5 1st DS0 2nd DS0 3rd DS0 ... 24th DS0 Frame 6 1st DS0 S 2nd DS0 S 3rd DS0 S ... 24th DS0
Frame 6 1st DS0 A 2nd DS0 A 3rd DS0 A
Frame 12 1st DS0 B 2nd DS0 B 3rd DS0 B
Frame 18 1st DS0 C 2nd DS0 C Frame 24 1st DS0 D 2nd DS0 D 3rd DS0 D
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.
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
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.
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.
IP Voice Technologies
Layer Purpose Examples
1 Endpoints IP Phone, Cell Phone, Video Phone, IM Client
2 Applications Voice Mail, Conference Call apps, Call Centre Apps, 911 Series
3 Call Processing Unified Communications Manager,
UC500 CME CCMBE CCM Max users 48 250 500 30000+
Redundancy support
no No No Yes
Cisco Voice Infrastructure Model
Call Processing Layer
Cisco Unified Communications 500 (UC500) – Appliance providing firewall, NAT, Integrated Voicemail & Auto Attendant, Built in FX0 & FXS Ports, VPN, Optional Wireless and Music on Hold. This is a part of the Cisco Smart Business Communications System (SBCS) range.
Cisco Unified Communications Manager Express (CME) – Next step up from the UC500.
Cisco Unified Communications Manager Business Edition (CCMBE) – Provides CCM call processing, Cisco Unity Connection and Cisco Unified Mobility applications.
Cisco Unified Communications Manager (CCM) – Call processing only. Supports redundancy and clustering.
Applications Layer Cisco Unity Express – Voicemail hardware (Network module or AIM) physically installed into a supporting router. Supports up to 250 users. This unit provides limited IVR capabilities in order to provide an Automated Attendant system.
Cisco Unity Connection – Cut down Cisco Unity supporting up to 500 users (7500 dedicated server). Also provides Advanced Call Routing facilities to calls can be routed based on rules, time of day, caller ID etc.
Cisco Unity – Full unified solution integrating with Exchange, Lotus Notes & Novell GroupWise. Up to 7500 users per server. Supports redundancy.
Cisco Unified Contact Centre – Provides ACD functionality to support a call centre environment.
Cisco Unified Meeting Place - Provides a multimedia conference solution that gives you the capability to conference voice, video, and data into a single conference call. For example, multiple offices could participate in a conference call using IP phones, live video feeds, and instant messenger clients. The
conference call could include PowerPoint presentations, shared whiteboards, or live demonstrations. The organization could also choose
to record the conference call for playback at a later time.
Cisco Unified Presence - Provides status and reach ability information for the users of the voice network. For example, Joe might check the status for Samantha and find that she is available on an instant messenger client but is currently engaged in a video call.
Cisco Unified Mobility - Allows users to have a single contact phone number that they can link to multiple devices. For example, Mike could have the phone number +442920 454343 that links to his desk phone, cell phone, and instant messenger client.
Cisco Emergency Responder - Because VoIP clients have the ability to “roam around” the network using wireless phones, Soft Phones, or extension mobility functionality, emergency calls (911/999) could pose a location problem. Cisco Emergency Responder (ER) dynamically updates location information for a user based on the current position in the network and feeds that information to the emergency service provider if an emergency call is placed. The Cisco ER product also helps manage emergency calls in a centralized IP telephony deployment, ensuring that branch office.
Infrastructure Layer The Infrastructure layer consists of the IP infrastructure to enable a VoIP telephone network (switched, routers etc). The uptime of a traditional PBS system if 99.999 percent so as a result the main factors in the IP infrastructure layer is redundancy and QoS to ensure good uptime and good quality speech.
Signalling
SIP - Developed by the IETF. This uses text strings similar to HTML for signalling. SIP itself is only responsible for setting up and tearing down sessions between endpoints, the actual session is transferred typically using RTP over UDP. Registrar, Redirect, Location and Proxy servers can be used.
H.323 - Created by the ITU-T to allow simultaneous voice, video and data transmission primarily across ISDN links. The signalling is derived from Q.931 signalling and as a consequence is very difficult to interpret. This is a peer to peer protocol so each gateway in the system is fully independent of any other and needs full configuration for all other gateways. This administrative burden can be reduced by incorporating a H.323 Gatekeeper, where the gatekeeper would have the full knowledge of the infrastructure and all Gateways would ask the Gatekeeper how to find other non local extensions. The Gatekeeper can also perform other tasks such as CAC (Call Admission Control) and bandwidth management. H.232 is also responsible for the transport of the media stream. This is the only signalling protocol that supports Fax connected to a Cisco ATA.
MGCP - Developed by Cisco and the IETF is a system which puts voice gateways under control of a centralised call agent. The gateway is considered a dumb device, every action such as a phone going off hook or a button pressed is relayed to the MGCP call agent to ask what to do next such as play a dial tone. This is not supported by CME.
SCCP
- Cisco proprietary prot
ocol used to control Cisco endpoints (IP Phones, ATA 186 etc). Works in a similar fashion to MGCP, the end device communicates with CME for every action
H.323 MGCP SIP SCCP Body ITU IETF IETF Industry Support Excellent Fair Very Good Proprietary Used on Gateways Yes Yes Yes Limited Used on Cisco phones
No Limited Yes Yes
Architecture Peer to Peer Client / Server Peer to Peer Client / Server
Version Header Length Type of Service Total Length
Identification Flags Fragment Offset
TTL Protocol Header Checksum Source IP Address
Destination Source Address
Source Port (16bits) Destination Port (16bits)
Length (16bits) Checksum (16bits)
Ver P X CC M PT Sequence Number
IP Transport
RTP - The media stream is carried using RTP on a negotiated UTP port between 16384 and 32767 (Even numbers).
RTCP – A RTCP session is created at the same time as the RTP session, this is used to relay statistics between the participating devices (and CME). Typically Packet count, Packet delay, Packet loss and Jitter statistics is transmitted. Uses odd number UTP ports
IP Overhead
As raw voice data is sent across a network link, layer 2 and layer 3 frame headers are added to the stream as below.
Layer 2 Ethernet – 18 bytes Frame Relay – 4 to 6 bytes Point to Point Protocol (PPP) – 6 bytes
Layer 3 Total of 40 Bytes IP – 20 bytes
UDP – 8 bytes
Real-time Transport Protocol (RTP) – 12 bytes
Timestamp
SSRC Identifier
CSRC Identifiers
Compressed RTP
Compresses the network and transport layer headers from 40 bytes down to 2 bytes (without checksum) or 4 bytes (with checksum). This is considered very processor intensive so is only used on low bandwidth links (T1 or lower)
Problems with Digital Voice
Bandwidth – 21kbps to 320kbps per call depending on codec. QoS can help prioritise voice during bandwidth use peaks.
Delay – A maximum one-way delay of 150ms, 200ms is considered the ultimate limit.
Jitter – Change of delay between packets, usually caused when there are multiple data paths available between the endpoints. A maximum one-way jitter delay of 30ms is advisable. A “De-Jitter Buffer” can be used to reduce the impact of jitter by buffering a small amount of speech in the device before playing it. Cisco devices implement a variable sized de-jitter buffer to tune to the connection quality. As a downside it introduces additional delay.
Packet Loss – As packets are lost there will be holes in the speech. Less than 1% is advisable.
Causes of Delay
Transmission delay – The physical time it takes for the packet to travel the wire (Fixed).
Serialization delay – The time it takes to place the bits on the wire (Fixed).
Codec delay – The time the codec takes to convert voice into a PCM stream.
Queuing delay – The time the packet remains in a queue waiting for transmission. QoS can influence this by putting packets in to a high priority queue.
QoS
Data applications classes Mission critical – Critical to the running of the business.
Transactional – Applications interact with the users and required rapid response times.
Best Effort
–
Noncritical
–
web browsing, email, ftp etc.
Mode Description Command
(config) Create a match all class map (default)
Class-map classname
(config) Create a match any class map
Class-map match-any classname
(config) Create a match all class map
Class-map match-all classname
(config-cmap) Match on an ACL Match access-group
Mode Description Command
(config) Create a policy map Policy-map type policyname
(config-pmap) Set a class map for this policy Class classname
(config-cmap-c) Set a priority bandwidth of kbps Priority kbps
(config-cmap-c) Set a priority bandwidth of Priority percent percent
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
Notes-
Ensure 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 on-
ACL Input interface NBAR (Network based application recognition). This looks at the up layers to find the application
Match-any signifies an OR condition between statements Match-all signifies an AND condition between statements
Policy-map Controls what to do with traffic
Example- (config) # Class-map match-any WEB_TRAFFIC - Class map to match on either HTTP or HTTPS (config-cmap) # Match protocol http (config-cmap) # Match protocol https (config) # Class-map match-all VOIP - Class map to match on RTP traffic (config-cmap) # Match protocol rtp (config) # policy-map VOIP - Policy map to give priority bandwidth to VOIP (config-pmap) # class VOIP (config-pmap-c) # priority 4000 (config) # interface Ethernet 0 - Set the QoS on an interface (config-if) # service-policy output VOIP
Analogue to Digital Conversion / Co
decs
Codec Bandwidth MOS Codec Delay
Complexity 20ms Sample Size (bytes)
Notes
iLBC 15.2kbps 4.1
G.711 64kbps 4.1 0.75ms Medium 160 G.729 8kbps 3.92 10ms High 20 Most Supported G.723.1 6.3kbps 3.9 30ms High
G.723.2 5.3kbps 3.8
G.726 32kbps 3.85 Medium G.726 24kbps G.729a 8kbps 3.7 10ms Medium
Conversion
1. Sample the waveform – Pulse Amplitude Modulation 2. Calculate the number representing each sample (quantisation) 3. Convert to binary – Pulse Code Modulation (G711a etc) 4. Compress if required
Codec Summary
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)
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 u-law country.
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
Phones
Lines Switch XML Apps PoE Notes
Text Graphics Pre 802.3af
7906G 1 No Yes No Yes Yes
7911G 1 Yes Yes No Yes Yes 7914/7915/7916
14 No No No No No Expansion Module
7920 1 No Yes No No No 802.11b Wifi Phone
7921 1 No Yes Yes Yes Yes A,B & G Wifi, PTT
7931 24 Yes Yes No No Yes 7936 1 No No No No No Conference Station 7937 1 No No No No Yes Conference
Station 7940G 2 Yes Yes Yes Yes No
7941G 2 Yes Yes Yes Yes Yes High res screen 7941G-GE 2 Yes Yes Yes No Yes Gig Ethernet 7942G 2 Yes Yes Yes Yes Yes High Quality Audio 7945G 2 Yes Yes Yes No Yes High res screen 7960G 6 Yes Yes Yes Yes No
7961G 6 Yes Yes Yes Yes Yes High res screen
7961G-GE 6 Yes Yes Yes No Yes Gig Ethernet 7962G 6 Yes Yes Yes Yes Yes High Quality Audio 7965G 6 Yes Yes Yes No Yes High res screen 7970G 8 Yes Yes Yes Yes Yes Colour Touch screen 7971G-GE 8 Yes Yes Yes No Yes Colour Touch screen 7975G 8 Yes Yes Yes No Yes Colour Touch screen 7985 1 Yes Yes Yes No Yes Video Phone ATA 186 2 No No No No No Dual FXS
ATA 188 2 Yes No No No No Dual FXS
Phone Range
Expansion Module adds an additional 14 lines to a 796x and 797x phones. Up to two units can be added.
Phone Boot Process
1. Switch detects PoE capabilities and sends power if required. 2. Phone boots software image. 3. Switch sends the Voice VLAN info to the phone using CDP. 4. IP Phone uses DHCP to get its IP address including ‘option 150’ (TFTP IP Address). 5. Phone contacts TFTP server and gets configuration file. 6. Phone registers with the CME Server listed in the config file.
Powering
Class Allocated Power
Actual Power Used
0 15.4W 0.44 to 12.95
1 4.0W 0.44 – 3.84W 2 7.0W 3.84 – 6.49W
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.
Midspan Power Power Patch Panel – Sits between the switch and patch panel to inject power. Avoids cost of replacing switches for PoE switch.
Power Injector – Simple power injector, no intelligence.
Wall Power CP-PWR-CUBE-3
Basic Configuration
Mode Description Command
# Show all defined vlans and assigned ports
Show vlan
# Show total power available / used and port power usage
Show power inline
# Show directly connected Cisco Device information
Show cdp neighors
# Show VTP mode and status Show vtp status
Set Switch Port Trunking Mode
(config-if) Set the trunk encapsulation (ISL no used much now)
Switchport trunk encapsulation dot1q
(config-if) Enable the trunk mode Switchport mode trunk
(config-if) Auto mode. Will aggressively try to raise a trunk. Default
Switchport mode dynamic desirable
(config-if) Auto. Will not raise trunk but will if the other end does.
Switchport mode dynamic auto
(config-if) Set native (untagged) Vlan Switchport trunk native vlan vlan
Set Switch Port Access Mode
(config-if) Set access port Switchport mode access
(config-if) Set the data vlan Switchport access vlan vlan (config-if) Set the voice/auxiliary vlan Switchport voice vlan vlan
Mode Description Command # Display DHCP leases Show ip dhcp binding
(config) Create a DHCP pool Ip dhcp pool pool
(dhcp-config) Define network to enable & issues addressed
Network x.x.x.x /24
(dhcp-config) Set default router Default-router x.x.x.x (dhcp-config) Set DNS server Dns-server x.x.x.x (dhcp-config) Set TFTP server address Option 150 ip x.x.x.x (dhcp-config) Set TFTP server name (not Option 66 ascii tftpservername
Switch configuration
Notes-
As a guideline make the voice VALNs lower in number than data. This allows spanning tree to get the Voice vlan up quicker in the event of a network topology failure.
Typically a router will have an access list to stop data and voice traffic crossing the Vlans.
Configuring DHCP
(config-if) Set helper address for a DHCP server on an interface
Ip helper-address x.x.x.x
Mode Description Command # Show NTP sources and status Show ntp associations
(config) Set a time server Ntp server domainname
(config) Set a hour zone and hour difference for the time
Clock timezone name x
Configure a router as an NTP Master
Notes-
The ‘Network’ command allows the addition of a mask bit length or network mask. Otherwise is will issue the default class full subnet mask. Common practice is to include the option 150 in data VLANs as well so phones will work if plugged into the data VLAN.
‘Ip helper address’ is used to create a proxy to send a broadcast received on an interface to a unicast address. When the unicast is sent it is sent to the address specified but with a source address of the interface the broadcast was received from. This allows a DHCP server to identify with DHCP pool to assign addresses accordingly. For this to work the DHCP server must have a route to the network requiring DHCP services.
Configuring NTP
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......
CME Communications Manager Express
Mode Description Command
# Show all flash files and free space Show flash
# Think DOS... Dir flash: # Install CME from TFTP Archive tar /xtract
Licensing
IOS – License to run the required IOS (Voice / AdvancedEnterprise etc). Think Windows Server Licence.
Feature License – License CME for a specific number of users. Think Windows CAL.
Phone User License – License the IP phone to interact with CME / CCM. Think Windows XP License.
CME Files
While all the functionality for running voice is built into the routers IOS, Cisco provide TAR files to provide additional resources for the phone system-
Basic Files – Phone loads / firmware.
GUI Files – HTML web front end.
XML Template Files – Allows the user to edit the GUI such as only allows certain user to perform certain actions.
MoH Files – Music on hold.
Script Files - TCL scripts for advanced functions (auto attendant, ACD etc).
Miscellaneous Files – Other files such as Custom ring tones.
Installing
1. Get the files.
2. Place the files on a TFTP server
3. Copy the files to the routers flash memory, either-
1. 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.
Basic CME Configuration
Mode Description Command
# Show telephony-service
Basic Configuration
(config) Go to telephone service configuration
Telephony-service
(config-telephony) Maximum directory numbers
Max-dn x
(config-telephony) Maximum phones on the system (up licenses purchased)
Max-ephones x
(config-telephony) Defines IP address the phones will attempt to register
Ip source-address x.x.x.x
Auto Registration and DN assignment
Mode Description Command # Show the internal telephony service
tftp files show telephony-service tftp-bindings
# Display the contents of a text file More filename
(config) Define a file for the TFTP server Tftp-server filename alias name
‘Ip source-address’ can be set to a loopback interface if supporting phones on more than one interface. The network and phones must have routes to this address.
Phone Loads / files
As the phone only asks for the filename, not the full path the alias element of the ‘tftp-server’ command provides the file alias. Examples- Tftp-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 reference- http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/requirements/guide/cme43spc.htm
Phone configuration files
Mode Description Command
(config) Create a single line dn Ephone-dn tag
(config) Create a dual line dn Ephone-dn tag dual-line (config-ephone-dn) Assign a number Number number (config-ephone-dn) Assign a secondary/did
number Number number secondary number
(config-ephone-dn) Assign a name for the telephone directory
Name name
(config-ephone-dn) Preference to use when same
number assigned to many dn’s. Default is 0.
Preference x
(config-ephone-dn) Consider this the last dn in the hunt group. Don’t try to find another
Huntstop
Mode Description Command # Show ephone
# Show ephones trying to register. Useful to find phone MAC when setting up phones
Show ephone attempted-registrations
(config) Create an ephone Ephone no
(config-ephone) Assign a MAC Mac-address xxxx.xxxx.xxxx
(config-ephone) Set phone type. Not required as CME will find this out
Type phonemodel
XMLDefault.cnf.xml – Basic phone configuration file, contains what IP address is hosting CME and firmware names to download. This can be viewed using the command ‘more system:/its/vrf1/XMLDefault.cnf.xml’
Ephone-dn
Represents the phone numbers.
Single Line - Only able to handle on call Dual Line - Handles two simultaneous calls – allows call waiting, conferencing, consultative transfers
EPhone
Represents the physical phone.
Auto Registration & Assignment
Mode From IOS Help Description
: Normal ring
S Silent ring
B Call waiting beep, no ring Silent ring but beep on call waiting
F Feature Mode Alternate ring tone for a incoming call M Monitor Mode Creates a button which shows the
status of the ephone-dn. Also acts as a speed dial button. Ideal for receptionist
W Watch Mode As monitor button but watches the
Auto Registration - By default ephones will automatically register with CME, they won’t automatically be created in the running config. Disabled with the ‘No auto-reg-ephone’ telephony service command.
Auto assignment – CME will automatically assign ephone-dn’s to ephones. Configured with ’Auto assign x to y’ where x is the start dn and y is the end dn.
Button command options Button assignments link a DN to a physical button on a telephone. A number of methods can be used on the assignments-
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 available-
Single 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
There is a problem using this approach when using a dual line ephone-d
n, as each DN can handle two calls, a second call to shared number could go to the second line of the DN resulting in a call waiting scenario.
The ‘Huntstop’ command stops a second call hitting a dn currently in use (huntstop channel) and places it on the next dn (no huntstop) Note the last dn has doesn’t have a ‘no huntstop’ command .
(config) # Ephone-dn 10 dual-line (config-ephone-dn) # Number 1010 (config-ephone-dn) # Preference 0 (config-ephone-dn) # Huntstop channel (config-ephone-dn) # No huntstop (config) # Ephone-dn 11 dual-line (config-ephone-dn) # Number 1010 (config-ephone-dn) # Preference 1 (config-ephone-dn) # Huntstop channel See the Cisco Website for more information.
Overlay Line buttons Allows an incoming call to ring multiple phones simultaneously i.e. the incoming call will be overlayed to multiple ephones.
(config) # Ephone-dn 10 (config-ephone-dn) # Number 1010 (config-ephone-dn) # Preference 0 (config-ephone-dn) # No huntstop (config-ephone-dn) # Exit (config) #Ephone-dn 11 (config-ephone-dn) # Number 1010 (config-ephone-dn) # Preference 1 (config-ephone-dn) # Exit (config) #Ephone 8 (config-ephone) # Button 1o10,11 (config-ephone) # Exit (config) #Ephone 8 (config-ephone) # Button 1o10,11 (config-ephone) # Exit
In this example multiple DNs are created allowing the shared number 1010 to be used multiple times for incoming and outgoing calls. The DNs are then overlayed to the telephone buttons, in effect a phone button will have multiple assigned DNs.
‘C’ Overlay Line (with call waiting). If the buttons are configured with ‘C’ instead of ‘O’, the first call will ring ephone 8 & 9. A second call will ring the inactive phone but the active user will receive a call waiting beep. Although the ephone-dn’s are single line and don’t support call waiting, the second call will come in on the inactive dn, dn 11 which will generate the call waiting beep..
Recommendation is to not use dual lines with ‘O’ and ‘C’.
Additional functions
Mode Description Command
(config) Select Cisco ephone dn ephone-dn dn
(config-ephone-dn) Assign a name for the telephone directory
Name name
(config) Select SIP register dn (for attached sip phones)
voice register dn dn
(config-register-dn) Assign a name for the telephone directory
Name name
(config) Select telephony service config mode
Telephony-service
Mode Description Command (config-ephone-dn) Forward all calls Call-forward all number
(config-ephone-dn) Set forward when phone busy Call-forward busy number
(config-ephone-dn) Set forward when phone not answered
Call-forward noan number timeout seconds
(config-ephone-dn) Forward call on activated night service
Call-forward night-service number
(config-ephone-dn) Restrict length of a forward number Call-forward max-length length
Voice network Directory (Local Directory on phone)
Call forwarding
User call forward
‘CFwdAll’ phone soft key allows a user to enter an extension to forward all calls to.
System call forward A DN can be configured with the command ‘Call-forward all XXX’ & ‘Call-forward busy XXX’ to define where to forward calls. Configuring-
‘Call-forward pattern pattern’ and ‘Call-forward max-length length’ are used to control what number calls can be forwarded to, this helps avoid call toll fraud.
H.450.3 - Allows the original caller and the recipient of the forward to handle the transferred call directly rather than via the intermediate party handling the media stream (call hair-pinning). This is enabled when a ‘call-forward pattern pattern’ is specified.
Call transfer
Consulted transfer – User presses the ‘Transfer’ soft key and dials the number to be transferred to. The user then consults the transfer recipient informing them of the call. The ‘Transfer’ soft key is then pressed to connect the two parties. This is the default.
Blind transfer
–
The call is transferre
d as soon as the transfer number is entered.
Mode Description Command
(config-telephony) Sets blind transfer system using H.450.2
Transfer-system full-blind
(config-telephony) Sets consult transfer system using H.450.2
Transfer-system full-consult
(config-telephony) Sets consult transfer Transfer-system local-consult
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).
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
(config) # ephone 1
(config-ephone-dn) # park reservation-group 1
Notes-
Once a park slot has been created the ‘Park’ button becomes available on the phones.
To pick the call up simply call the parked call number or press this ‘PickUp’ softkey then dial the call park no. Additionally the person who parked the call can pick up the call by pressing ‘PickUp’ soft key then press the * key.
Call Pickup
Directed Pickup – Pressing the Pickup button results in the phone sounding a dial tone waiting for the user to enter the extension number of a ringing phone to pickup.
Local Group Pickup – Pressing the GPickup button picks up a ringing phone in the same pickup group.
Other Group Pickup - Pressing the GPickup button results in the phone sounding a dial tone waiting for the user to enter the group number a ringing phone to pickup.
To assign a dn to a group use the command-
(config-ephone-dn) # pickup-group xxxx
Notes-
The GPickUp softkey functions differently depending on the call pickup configuration in CME. If there is only one group configured in CME, pressing the GPickUp button automatically answers the call from your own group number. You will not hear a second dial tone and you do not need to dial an asterisk to signify your own group, because only one group is defined. Once you have configured multiple groups in CME, you will hear a second dial tone after pressing the GPickUp softkey, at which point you can dial either an asterisk for the local group or another group number.
Directed Pickup can be disabled by entering ‘no service directed-pickup’ from telephony service configuration mode.
Intercom
A two way communication channel using speaker phone. When a user presses the button assigned to the intercom the other phone will automatically answer using speaker phone but with the microphone muted in case the other person is saying something secretive.
(config) # ephone-dn 20
(config-ephone-dn) # number A100 (config-ephone-dn) # intercom A101 label “Manager”
Mode Description Command
(config-telephony) Define outside of hours on particular day
After-hours date month dayno
(config-telephony) O of H on day between start & endtime
After-hours day day starttime endtime
(config-telephony) Define blocked number pattern (up After-hours block pattern no pattern
(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 command-
Barge-in – the intercom will force all other calls into the HOLD state and connect tyhe intercom call
No-auto-answer – Disable the intercom auto answer
No-mute – Disable the auto mute.
Paging
A one way speakerphone based announcement. There are two methods, unicast or multicast. As unicast requires a single stream per page group member the group is limited to a maximum of 10 members. If using multicast the network must be capable/configurable of supporting multicast streams. A phone can only be a member of one paging group but a paging group can be a member of another parent paging group.
Create a paging group- (config) # ephone-dn 25 (config-ephone-dn) # number 3000 (config-ephone-dn) # paging - Unicast paging or (config-ephone-dn) # paging ip 239.4.3.4 port 200 - Multicast paging (cannot use 224.) (config-ephone-dn) # paging group dnlist - 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.
(config-ephone) Exempt phone from out of hours block
After-hours exempt
(config-ephone) Set a pin for temporarily removing blocks
Pin xxxx
Example-
After-hours day mon 17:00 8:00 After-hours day tue 17:00 8:00 After-hours day wed 17:00 8:00 After-hours day thu 17:00 8:00 After-hours day fri 17:00 8:00 After-hours date dec 25 00:00 00:00 After-hours block pattern 1 90.......... - Block all non local calls
Music on Hold
Stream a wav or au files in the routers flash memory using unicast (up-to 10 like paging) or multicast.
Example-
(config-telephony) # moh music.wav (config-telephony) # multicast moh 239.4.3.2 port 2100 - Multicast if required
CME GUI
Provided the GIU Files have been installed on the router, the HTML front end can be enabled using the following commands-
(config) # ip http server - Enable http server (config) # ip http secure-server - Enable https server (config) # ip http path flash:/gui - Set the location of the gui files (config) # ip http authentication local - Set local authentication database Additional commands to control the front end- (config-telephony) # web admin system name mike secret password (config-telephony) # dn-webedit - Enable changing dn through the gui (config-telephony) # time-webedit - Enable changing time through the gui
To use simply browse to ‘http://x.x.x.x/ccme.html’
Gateways
Mode Description Command
# Show the summary and status of all voice ports
Show voice port summary
# Show the summary and status of all dial peers
Show dial-peer voice summary
# Debug the dial peer process
Debug voip dialpeer
# Show all voice calls Show voice call summary # show call active voice Create POTS FXS Dial Peer extension
(config) Create a dial peer Dial-peer voice tag pots
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.
(config-dial-peer) Description description
Create VoIP Outbound Dial Peer
(config) Create a dial peer Dial-peer voice tag voip
(config-dial-peer) Set the destination pattern Destination-pattern pattern
(config-dial-peer) Send matching calls to the remove voip server
Session target ipv4:x.x.x.x
(config-dial-peer) Description description Create a T1/E1 Outbound Dial Peer
(config) Create a dial peer Dial-peer voice tab pots
Wildcard Meaning Example Matches . A single digit 50. 500, 501 ... 509
+ One or more instances of
1+ 11, 111, 11111111
[] Range of digits [1-3]111 1111, 2111, 3111 [14-6]11 111, 411, 511, 611 [6789].. 6xx, 7xx, 8xx, 9xx
T Anything 9T Anything starting with a 9.
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-
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 be-
Leg 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 order-
1. Matched the dialled number (DNIS) using the ‘incoming called-number’ dial peer configuration command.
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.
Digit Manipulation
POTS Auto stripping
Pots dial peers automatically strip any explicitly defined number from the destination pattern before sending the call.
Examples
Destination-pattern 9[2-9]....... The 9 will be stripped Destination-pattern 9[469]11 The 9 & 11 will be stripped Destination-pattern 91[2-9]....... The 9 & 1 will be stripped Destination-pattern 9011T The 9011 will be stripped
Prefix <digits> Add the prefix to the remaining dialled digits.
Forward-digits <number> forward number of right most digits, including any digits automatically stripped.
Digit-strip Default action. Turn off auto stripping using the command no digit-strip.
Num-exp <match> <set> Effectively search and replace. Global config command.
Example PSTN Failover
Example - sending calls for 6... to a remote phone system using an IP trunk. If the trunk fails the calls will be sent out a POTS voice port to numbers relating to the DID numbers of the extensions, eg 6001 will get sent to the PSTN number 02920116001 which the receiving phone system will forward to the extension 6001.
(config) # Dial-peer voice 6000 voip (config-dial-peer) # Destination-pattern 6... (config-dial-peer) # Session-target ipv4:10.1.1.2 (config-dial-peer) # Preference 0 (config) # Dial-peer voice 6001 pots (config-dial-peer) # Destination-pattern 6... (config-dial-peer) # No digit-strip (config-dial-peer) # Prefix 0292011 (config-dial-peer) # Port 1/0:15 (config-dial-peer) # Preference 1
Example 0 for operator
(config) # Num-exp 0 5000
Configuring Voice Ports
Mode Description Commands
(config) Select interface Controller interface
CAS
(config-controller) Set framing (esf most common) Framing <sf / esf>
(config-controller) Set coding (b8zs used with esf)
Linecoding <ami / b8zs>
(config-controller) Configure CAS Ds0-group groupnumber timeslots x-y type signalling CCS
Mode Description Command (config-voiceport) Set start method. Loopstart is
default. Used when trunking to a pbx
Signal <groundstart / loopstart>
(config-voiceport) Set the dial tone. Also changes the ring cadence accordingly
Cptone <countrycode>
(config-voiceport) Change the ringing AC frequency
Ring frequency <25 / 50>
(config-voiceport) Set the ring pattern Ring cadence patternxx (config-voiceport) Set custom ring cadence Ring cadence x y z . . . . . Busyout
Configuring VWIC T1 & E1 cards
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 - Same as tag number (config-dial-peer) # Preference 1 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
Signal <groundstart / loopstart>
(config-voiceport) Set the dialling signalling method Dial-type <dtmf / pulse>
FXO
Unity
Unity Express Unity Connection Unity
Max Mailboxes 250 7500 7500 per server
Voice Mail Yes Yes Yes Integrated Messaging
Yes Yes Yes
Unified Messaging No No Yes
Auto Attendant Yes Yes Yes Platform Linux router based Windows / Linux
Server Windows Server
AIM-CUE NM-CUE N-CUE-EC NME-CUE Max Mailboxes
50 100 250 250
Voice Ports 6 8 16 24 Installation Internal NM Slot NM Slot NM Slot Storage (hrs) 14 100 300 300 Concurrent 2 5 5 5
Unity Range
Unity Express
CUE Features
Voicemail (User Mailbox). A user/subscriber has his/her own personal mailbox. A pin is required to login.
Voicemail (General Delivery Mailbox) is a shared mailbox accessible by many subscribers. Subscribers must be made a member of the GDM to access it and will be prompted to access it when checking their own personal mailbox. A pin is not required. IVR (Interactive Voice Prompt) is a system where the system the phone system plays a prompt then waits for a user to respond. Typical uses are an auto attendant and bank automated balance enquiry.
Auto Attendant allows users to direct themselves to the correct person eg ‘Press 1 for Sales, 2 for Accounts’. Two scripts are provided with the system ‘Auto Attendant Script’ & ‘Auto Attendant Simple Script’. By default the following greetings are available ‘Welcome prompt’, ‘Business Open prompt’, ‘Business Closed prompt’ & ‘Holiday prompt’.
Administration via Telephone (AVT) allows an admin to record greetings and prompts.
Backup and restore functionality is provided making use of an FTP server. This requires administrator access to the web gui.
Message Waiting indicator alerts the user there is a message waiting by flashing a red light and displaying an envelope on the phone display.
Message Notifications
are additional methods of alerting the user there is a message. The notification can be to ring a phone or send an email.
Troubleshooting
From IOS- Show interface service-engine 1/0 Service-module service-engine 1/0 status - Should be in a steady state Show dial-peer voice <tag> Debug ephone mwi From CUE Trace <all/ccn/dns/....> Show trace buffer
Setup Process
1. Configure IOS service-engine and service-module for IP connectivity. 2. Create SIP dial peer for CUE. 3. Create MWI notification ephone dn’s. 4. Perform initial config – domain name, hostname, NTP servers & admin credentials. 5. Run Initialisation Wizard (import users, MWI methods, voicemail access number, administration by telephone number etc).
Initial Engine Setup
Once installed a ‘service-engine x/y’ interface appears in the routers config, this is the routers interface to the Unity Express module. There are two methods of assigning it an IP address-
Method 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
Controlling / Connecting to the module
# service-module service –engine0/1 sessions - Connect to the module using the specific engine # service-module service –engine0/1 reload - Reload the module # service-module service –engine0/1 reset - Reset the module # service-module service –engine0/1 shutdown - Shutdown the module (before powering off router) # service-module service –engine0/1 status - Show the status of the CUE module
Initial Configuration of the Module
# service-module service –engine0/1 sessions - Initiates a telnet connection to the module > enable - enter privileged mode # offline - Take module offline # restore factory default Once restored the unit will reboot and show the prompt- ‘Do you wish to start configuration now (y,n)?’ Enter Host Name? Enter Domain Name? Would you like to use DNS for CUE (y,n)? Enter IP Address of the Primary NTP Server? Enter IP Address of the Secondary Server? Please Identify a location so that time zone rules can be set correctly? 1) Africa, 2) Americas ....... Please select a country? 1) Anguilla, 2) Antigua & Barbados ...... Please select one of the following time zones regions. 1) Eastern Time, 2) Eastern Time – Michigan.... ** Is the above information OK? 1) Yes, 2) No Waiting xxx ..... After booting it prompts for the admin user account details Enter administrator user ID: Enter password for : ** US Additional Option
Upgrading CUE CUE # software install clean url ftp://x.x.x.x/cue-vm-k9.nm-aim.4.2.1.pkg * Language Installation Menu : 1 ITA, 2 ESP ........ ** # enter the number for the language to sellec one R # - remove the language for given # I # - more information about the language for a given ‘ x- Done with language selection Enter Command:
CUE # software install clean url ftp://x.x.x.x/license
*CUE uses a username and password of ‘anonymous’. Ensure the FTP server has this account setup. ** Corresponding language file must be downloaded as well.
NOTE an upgrade can be performed using the command software download upgrade only from version 2.3.4
Configure CME to access CUE
CME communicates with the CUE using a SIP dial-peer.-
(config) # dial-peer voice 700 voip (config-dial-peer) # Destination-pattern 7.. (config-dial-peer) # session target ipv4:192.68.100.2 (config-dial-peer) # session protocol sipv2 (config-dial-peer) # dtmf-relay sip-notify - out of band DTMF (config-dial-peer) # codec g711ulaw (config-dial-peer) # no vad - 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
CUE Web Interface
Initialisation Wizard
The Web username and password allows the CUE Module to get the current dn config from CME and administer it.
Password – Web Interface (GUI)
PIN – Telephone interface(TUI)
Voice Mail Number – This configure the CUE voicemail number and configure the phones message button to this number.
Auto Attendant Access Number- Configures the CUE AA number.
Administrator via Telephone Number (ATN)
–
Allows administering the CUE using a telephone number
(message prompt recording etc).
SIP MWI Notification Mechanism – Other options are ‘Subscribe – Notify’ .....
Smart Business Communication System
UC520 UC520 Model UC520
Users 8 or 16 8 or 16 24,32 or 48
1.5u desktop 2u rack Music on Hold 3.5mm Jack 3.5mm Jack 3.5mm Jack 10/100 PoE 8 (Max 80 watt) 8 (Max 80 watt) 8 (Max 80 watt) LAN Expansion 10/100
1 1 1
WAN 10/100 1 1 1
FXS 4 4 4
FXO 4 0 4 BRI 0 2 4 T1 / E1 0 0 1 VWIC 1 1 1
CE520-8PC CE520-24TT CE520-24LC CE520-24PC CE520G-24T 10/100 24 20
10/100 PoE 8 4 24
UC520 – Central point of the IP based system. Provides routing, security, VPN (10 users), call processing for 8-48 phones, voicemail & auto-attendant. This is based on CUCME 4.2 and CUE 3.1 for voice mail
CE520 – Catalyst Express Switch family
Cisco 521 – Wireless Express Access Point. This can operate in either standalone mode (mode one) or Controller based mode (mode two). CCA can manage up to three independent access points.
Cisco 526 – Wireless Express Mobility Controller. Can control up to 6 Cisco 521 Access Points. CCA can control two controllers allowing for up to 12 AP in a single SBCS deployment.
CCA – Cisco Configuration Assistant, the configuration tool for SBC devices. Default username / password ‘cisco’ & ‘cisco’
Typical UC520 Models
Typical CE520 Models
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
CCA Communities
CCA can discover devices using three methods- FQDN IP Address Subnet search
Cisco Configuration Assistant Tabs
Device Displays the platform and installed interfaces (VIC, Wireless, FXO etc)
Options to ‘Configure as a PBX’ or ‘Configure as a Key system’
System Options for ‘Region’, ‘Phone Language’, ‘Voicemail Language’, Data & Time formats, ‘System Message’ & System Speed Dials
Network IP address, DHCP, Voice Vlan
AA & Voicemail Configure the AA & Voicemail extension pilot number and PSTN numbers. Ability to choose the AA script and number options
SIP Trunk Settings to connect to an ITSP (Internet Telephony Service Provider). Registrar, Proxy & MWI Server.
Voice Features Music on Hold, Paging, Group Pickup, Caller ID Block, Outgoing Call Block Number List, Intercom, Hunt Group, Call Park, Multi-party Conference
Dial Plan Number of digits per extension, Outgoing Call Handling (area code, local number etc size). Outgoing access code (9). Incoming call Handling / DID
Users User Phone assignment (names, numbers etc)
Additional Resources
The Techexams Forums-
http://www.techexams.net/forums/ccna-voice/
Cisco Communications Manager Express Web site-
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/tsd_products_support_series_home.html
Various LABs for Cisco certifications-
http://configurethenetwork.com/