CCNA Collaboration – Notes, Chapter 7

CICD – Ch7 – Enabling Telephony Features with CME

Configuring Voice Network Directory

  • IP Phones support a local directory
    • Updates from CME as devices get configured
  • Names are entered under ephone-dn configuration
    • Used for both building the local directory (internal phone directory) and caller ID information
 number 2001
 name Phone One
 number 2022
 name Super Man
 number 2033
 name Super Women

IP Phones allow you to search the directory. You can do this by tapping the directories button on the phone. It’ll bring up a new display where you can see Missed Calls, Received Calls, Placed Calls and Local Directory (along with some more)

In Local Directory you can search for first name / last name and directly dial from the search. The search sorts alphabetically by first name. This can be changed:

CME(config-telephony)#directory ?
  entry             Define new directory entry
  first-name-first  first name is first in ephone-dn name field
  last-name-first   last name is first in ephone-dn name field

CME(config-telephony)#directory en
CME(config-telephony)#directory entry ?
  <1-250>  Directory entry tag
  clear    clear all directory entries

CME(config-telephony)#directory entry 1 ?
  WORD  A sequence of digits representing dir. number

CME(config-telephony)#directory entry 1 95550123 name John Doe
CME(config-telephony)#directory las                           
CME(config-telephony)#directory last-name-first ?

CME(config-telephony)#directory last-name-first 
CCP - Changing Directory naming schema

CCP – Changing Directory naming schema

Name for an extension

Name for an extension

Configuring Call Forwarding

Call forwarding allows a user to forward any call coming into their extension and have it be sent to another number of their choosing (within parameters of COR)

2 Methods for configuring

  • From the IP Phone – End user method
    • User chooses CFwdAll softkey and enters the phone number they want to forward calls to. If external make sure to enter the access key to get to PSTN
  • From CME CLI – CME Administrator method
CME(config)#ephone-dn 40
CME(config-ephone-dn)#call-forward ?
  all            forward all calls
  busy           forward call on busy
  max-length     max number of digits allowed for CFwdAll from IP phone
  night-service  forward call on activated night-service
  noan           forward call on no-answer

CME(config-ephone-dn)#call-forward all ?
  WORD  A sequence of digits

CME(config-ephone-dn)#call-forward all 91234567 ?

CME(config-ephone-dn)#call-forward all 91234567 

Call-forward Pattern to support H.450.3

H.450.3 standard allows CME to redirect call directly to the final destination. Without this standard a call could come into CME be routed to a remote gateway at site A where the extension lives and then call forwarded to the cfwd destination that exists at site B

With H.450.3 enabled we can bypass the hairpinning affect at site A where CME will directly send the call to the cfwd destination at site B

CCNP Track goes into this standard in much more detail

CME(config-telephony)#call-forward ?
  pattern  H.450.3/SIP match calling-party number forwarding pattern. This is used to enable remote party
           redirect forwarding with H.450.3/SIP. Default is local rotary forwarding only.
  system   Define call forward system parameters. 

CME(config-telephony)#call-forward pattern ?
  WORD  A sequence of digits (including wildcards) - representing calling-party number. Use '.T' to enable
        remote party redirect forwarding for all VoIP calling-party numbers.
CME(config-telephony)#call-forward pattern 44..

Use H.450.3 standard for any call-forward matching 4400-4499

Configuring Call Transfer

Transferring a call allows you to forward a call that came into you to another person. You do this by tapping on the Trnsfer softkey on the phone and entering the extension to send the call to.

3 Transfer modes exist

  • Full-Consult
    • Allows you to talk to receiving caller before completing the transfer
    • Default in CME
    • Requires dual line configuration
    • Based on H.450.2 standard – prevent hairpinning
  • Full-Blind
    • Immediately transfer to call to the dialed extension
    • Works on single line configuration
    • Based on H.450.2 standard – prevent hairpinning
  • Local-Consult
    • Cisco proprietary method of the consult transfer. Works similar to H.450 standard, only difference is when operating with non-Cisco phones in the network
transfer-digit-collect  transfer-pattern  transfer-system  

CME(config-telephony)#transfer-system ?
  full-blind     Perform call transfers without consultation using H.450.2 or SIP REFER standard methods
  full-consult   Perform H.450.2/SIP call transfers with consultation using second phone line if available,
                 fallback to full-blind if second line unavailable. This is the recommended mode for most
                 systems. See also 'supplementary-service' commands under 'voice service voip' and
  local-consult  Perform call transfers with local consultation using second phone line if available,
                 fallback to blind for non-local consultation/transfer target. Uses Cisco  proprietary

CME(config-telephony)#transfer-system full-consult

By default – Cisco only allows transfers within the internal system. This is done to help prevent toll fraud. To allow transfers to outside world use the transfer-pattern command or go under advanced features in CCP.

CME(config-telephony)#transfer-pattern ?
  WORD  digit string pattern for permitted non-local call transfers


Configuring Call Park

Call park allows you to pick up a call that has been placed on hold. Calls on hold are only allowed the be answered again on the original phone. Parking the call works by using a free ephone-dn in the CME config that hasn’t been assigned to a phone and has specifically designated as a call park slot

Park softkey on the phone
Once a call is parked the phone will display an extension. Any other phone may dial this extension to answer the parked call. Can also use the PickUp softkey and dial the park number to retrieve from

CME(config)#ephone-dn 30
CME(config-ephone-dn)#number 3000
CME(config-ephone-dn)#name Park
CME(config-ephone-dn)#park-slot ?
  directed           Configure this park slot to be used for directed park 
  reservation-group  Reserve this park slot for the exclusive use of the phone with the same reservation
  reserved-for       Reserve this park slot for the exclusive use of the phone with the extension indicated
                     by the transfer target extension number
  timeout            Set call park timeout

CME(config-ephone-dn)#park-slot timeout ?
  <0-65535>  Specify the park timeout (seconds) before the call is returned to the number it was parked

CME(config-ephone-dn)#park-slot timeout 60 ?
  limit  Set call park timeout count limit

CME(config-ephone-dn)#park-slot timeout 60 li
CME(config-ephone-dn)#park-slot timeout 60 limit ?
  <1-65535>  Specify the number of park timeout cycles before the call is disconnected

CME(config-ephone-dn)#park-slot timeout 60 limit 2 ?
  notify    Define additional extension number to notify for park timeout
  recall    recall transfer back to originator phone after timeout
  transfer  Transfer to originator or specified destination after timeout limit exceeded

CME(config-ephone-dn)#park-slot timeout 60 limit 2 recall ?
  alternate  Transfer to alternate target if original target is busy
  retry      Set recall/transfer retry interval if target is in use

CME(config-ephone-dn)#park-slot timeout 60 limit 2 recall 

Park-slot command functions

  • reservered-for <dn> – allows to reserve the call park slot for a DN, other phones cannot use this call park slot
  • timeout <seconds> – Number of seconds CME will wait before notifying the phone that parked the call
  • limit <count> – Limit the number of timeout intervals
  • notify <dn> – Notify a different DN than the phone that parked the call
    • only
  • recall – cause the call to ring back to original phone that parked the call
  • alternate <dn> – allows to transfer destination
  • retry <seconds> – Amount of time before CME attempts to transfer a parked call again

Configuring Call Pickup

Allows you to answer another ringing phone in the org from your local phone. This is done by using the PickUp softkey while the other phone is ringing. PickUp groups can be created to divide the phones out. Based on the group the phone is assigned to you can call PickUp for that group

By default phones managed by CME a user can pickup other ringing phones. Can disable this under the telephony-services with the no service directed-pickup
Doing this sets the PickUp softkey to work for local group pickup

3 methods to answer ringing phones

  • Directed PickUp
    • Press PickUp softkey and dial the DN of the ringing phone. CME transfers the call to the phone you are on
  • Local Group PickUp
    • PickUp a call in the same PickUp Group, use the GPickUp softkey
  • Other Group PickUp
    • Answer another groups ringing phone by using the GPickUp softkey and dialing the DN

Configuring Intercom

Allows phones to have a speakerphone ‘tether’ between them. The deployment of intercoms works through speed-dial and auto-answer speed-dial configuration.

Example: Admin Assistant can press the button configured for intercom (speed-dial button) which will dial the Execs phone which is setup for auto-answer. The call is answered muted and the Exec can unmute to respond

To prevent users from mis-dialing the intercom, use something that cannot be dialed from other IP phones

CME#conf t
Enter configuration commands, one per line.  End with CNTL/Z.
CME(config)#ephone-dn 20
CME(config-ephone-dn)#number A200 ?
  no-reg     Set E164 not register
  secondary  secondary dn number

CME(config-ephone-dn)#number A200 
CME(config-ephone-dn)#name Assistant
CME(config-ephone-dn)#intercom ?
  WORD  A sequence of digits - representing intercom/auto-call extension to call

CME(config-ephone-dn)#intercom A201 ?
  barge-in        Allow intercom calls received on this DN to force other calls into the call HOLD state to allow the incoming intercom
                  call to immediately connect without waiting
  label           Define a text label for the intercom
  no-auto-answer  Disable intercom auto-answer
  no-mute         Disable intercom mute-on-answer
  ptt             Define intercom push-to-talk

CME(config-ephone-dn)#intercom A201 label Manager ?
  ptt  Define intercom push-to-talk

CME(config-ephone-dn)#intercom A201 label Manager 
CME(config)#ephone-dn 21
CME(config-ephone-dn)#number A201
CME(config-ephone-dn)#intercom A200 label ?   
  WORD  Intercom text label, use quoted string if including spaces

CME(config-ephone-dn)#intercom A200 label Assistant
CME(config)#ephone 20
CME(config-ephone)#mac-address cccc.cccc.cccc
CME(config-ephone)#type 7975
CME(config-ephone)#button 2:20
CME(config)#ephone 21
CME(config-ephone)#mac-address bbbb.bbbb.bbbb
CME(config-ephone)#type 7965
CME(config-ephone)#button 2:21
restarting BBBB.BBBB.BBBB

Configuring Paging

Similar to intercom except that it only provides one-way automatic path for communication. Broadcast a message

This works by assigning an ephone-dn as a paging number. Calls into the paging DN will broadcast to the phones that have been assigned to the paging group. Phones are only allowed to be assigned to one group, but you can create a paging number that pages multiple paging groups.

CME Paging can be unicast or multicast. CME limits a unicast paging group to 10 phones. Multicast is outside of the scope of the CCNA. The note for multicast is that you must enable to the network to route multicast within the network. When I get back to my CCIE studies I’ll write up a post specifically on the different types of multicast.

Unicast Paging

CME#conf t
Enter configuration commands, one per line.End with CNTL/Z.
CME(config)#ephone-dn 25
CME(config-ephone-dn)#number 2525
CME(config)#ephone 10
CME(config-ephone)#paging-dn 25
Set phone's mac address or device-id first
CME(config-ephone)#mac-add acac.acac.acac
CME(config-ephone)#paging-dn 25
CME(config)#ephone 11
CME(config-ephone)#mac-add caca.caca.caca
CME(config-ephone)#paging-dn 25

Multicast Paging

CME(config)#ephone-dn 26
CME(config-ephone-dn)#number 2626
CME(config-ephone-dn)#paging ip port 2626 
ephone-dn 4  number 2727  name GroupA  paging ip port 2000  exit ephone 21  paging-dn 4 multicast  exit

ephone-dn 4  number 2727  name GroupA  paging ip port 2000  exit ephone 21  paging-dn 4 multicast  exit

Configuring After-Hours Call Blocking

After hours call blocking allows for a range of times to be defined as specific after hours. This can then be used to list number patterns that are not allowed during those times. Calls made during that time will receive a reorder tone and then disconnected.

Exceptions to the rule can be made by creating a back-door. A PIN can be entered on the phone to allow a call to be made. The time is configurable in CME to say how long the PIN is good for

Configuration Steps

  1. Define day and/or hours of the day that are considered off hours
  2. Specific patterns that will be blocked for the times in step 1
  3. Create exemption policy if required

telephony-service  after-hours block pattern 1 1900.......  login timeout 60 clear 0:0  after-hours day mon 18:0 7:1  after-hours day tue 18:0 7:1  after-hours day wed 18:0 7:1  after-hours day thu 18:0 7:1  after-hours day fri 18:0 7:1  after-hours day sat 7:2 7:1  after-hours day sun 7:2 7:1  exit

telephony-service  after-hours block pattern 1 1900…….  login timeout 60 clear 0:0  after-hours day mon 18:0 7:1  after-hours day tue 18:0 7:1  after-hours day wed 18:0 7:1  after-hours day thu 18:0 7:1  after-hours day fri 18:0 7:1  after-hours day sat 7:2 7:1  after-hours day sun 7:2 7:1  exit

Configuring CDRs and Call Accounting

Call Detail Records (CDR) contain information about calls coming into, out and between phones on the network. CDR contains all the information to find who called who and how long the conversation lasted

CME logs CDR to ram and/or syslog. RAM is not the best because it can all be lost if the router reboots. RAM is also a limited resource on the router. 

I do not have a syslog server in my lab, but will sent to a fake address for configuration purposes

CME(config)#gw-accounting syslog ?
  stats  Enable stats as part of accounting.

CME(config)#gw-accounting syslog 

CDR could be used for billing purposes (track who’s calling LD, international, etc), and do a charge back model. CME can make this easier by flagging these calls in CDR with an Accounting code.

Users would press an Acct softkey on the phone, enter a PIN (accounting code) to allow them to dial out to long distance or international phones. After the user presses the Acct softkey, an Acct prompt appears at the bottom of the phone, where the user can enter the accounting code followed by the pound key (#). Entering this number during the ring out or connected call state does not interrupt the call in any way. After the user enters the account code, CME flags the CDR records with the account number dialed. This allows for easy filtering and accurate billing to each department.

Configuring Music on Hold

CME can stream MOH from an audio file that is located on the routers flash. MOH can be either unicast or multicast. CME supports G.711 or G.729 codecs for MOH.

G.711 is recommended for MOH as G.729 would require DSP resources for transcoding

CME(config-telephony)#moh ?
  WORD  music-on-hold filename containing G.711 A-law or u-law 8KHz encoded audio file (.wav or .au format). The file must be loaded into
        the routers flash memory, e.g., flash:/ or flash:/audio/  The minimum supported file size is 100 Kb.

CME(config-telephony)#moh flash:moh.wav
CME(config-telephony)#multicast moh ?
  A.B.C.D  Define music-on-hold IP multicast address from flash

CME(config-telephony)#multicast moh port 2222

Configuring Single Number Reach (SNR)

SNR allows a call to come into a single phone number and ring multiple phone destinations. You can link multiple numbers to a parent number (office extension for example). When a call comes into your office DN, if you do not answer at the desk phone it can start to ring your cell phone as well and if no answer the call is transferred to corp voicemail. 

If the call was picked up on the cell phone and you get back to your desk you can “move” the call to the deskphone by hanging up on the cell and continuing the call on the desk phone.

Vice versa, you can be on a call on your desk phone and press the Mobility softkey to move the call to your cell phone without dropping the call.

When configuring SNR in the real world make sure you have enough PSTN trunks available as SNR will use 2 channels (1 for the inbound call to the desk, 1 for the outbound to the cell).

SNR is enabled on a DN by DN basis

ephone-dn 3  no snr calling-number local  snr 915558605555 delay 6 timeout 30  mobility  exit

ephone-dn 3  no snr calling-number local  snr 915558605555 delay 6 timeout 30  mobility  exit

Configuring Hunt Groups

Hunt groups allow you to call a specific number and have multiple phones ring in a configured sequence. 

Hunt groups are configured by creating a pilot number (ephone-hunt, pilot sub-command). The pilot number is what a person would call into that would then start ringing the phones configured in the hunt group in a specific order.

Types of Hunt Groups

  • Longest Idle – Round Robin distribution. Phones are run in a listed order, starting with the phone that has the longest on-hook
  • Peer – Round robin distribution. Extensions are rung in a listed order. Starts with the number after the one last answered a call 
  • Parallel – All phones ring simultaneously
  • Sequential – Strict top-down ordered list. First phone always get the new call unless it is already busy or answer within the timeout period

ephone-hunt 1 longest-idle  pilot 5550011  list 2001,2033  description HuntPilot  no-reg both  exit

ephone-hunt 1 longest-idle  pilot 5550011  list 2001,2033  description HuntPilot  no-reg both  exit

If no answer is made from the Hunt Pilot the following options are available

  • Forward to a pilot number of another hunt group
  • Voicemail pilot
  • Any ephone-dn
  • Back to ephone-dn that transferred the call to the hunt pilot

Configure Night Service

Night service is a specialized afterhours call forwarding system. Night service defines one or more extensions that are eligible for night service that will ring during this time. 

telephony-service  night-service day sat 07:01 07:00  night-service day fri 19:00 07:00  exit telephony-service  no night-service day sat 07:01 07:00  night-service day sun 07:01 07:00  night-service day sat 07:01 07:00  exit telephony-service  no night-service day sun 07:01 07:00  night-service day mon 19:00 07:00  night-service day sun 07:01 07:00  exit telephony-service  no night-service day mon 19:00 07:00  night-service day tue 19:00 07:00  night-service day mon 19:00 07:00  exit telephony-service  no night-service day tue 19:00 07:00  night-service day wed 19:00 07:00  night-service day tue 19:00 07:00  exit telephony-service  no night-service day wed 19:00 07:00  night-service day thu 19:00 07:00  night-service day wed 19:00 07:00  exit telephony-service  no night-service day fri 19:00 07:00  no night-service day thu 19:00 07:00  night-service day fri 19:00 07:00  night-service day thu 19:00 07:00  exit telephony-service  night-service code *1234

telephony-service  night-service day sat 07:01 07:00  night-service day fri 19:00 07:00  exit telephony-service  no night-service day sat 07:01 07:00  night-service day sun 07:01 07:00  night-service day sat 07:01 07:00  exit telephony-service  no night-service day sun 07:01 07:00  night-service day mon 19:00 07:00  night-service day sun 07:01 07:00  exit telephony-service  no night-service day mon 19:00 07:00  night-service day tue 19:00 07:00  night-service day mon 19:00 07:00  exit telephony-service  no night-service day tue 19:00 07:00  night-service day wed 19:00 07:00  night-service day tue 19:00 07:00  exit telephony-service  no night-service day wed 19:00 07:00  night-service day thu 19:00 07:00  night-service day wed 19:00 07:00  exit telephony-service  no night-service day fri 19:00 07:00  no night-service day thu 19:00 07:00  night-service day fri 19:00 07:00  night-service day thu 19:00 07:00  exit telephony-service  night-service code *1234

ephone-dn 3  night-service bell  exit

ephone-dn 3  night-service bell  exit

ephone 11  mac-address CACA.CACA.CACA  exit ephone 11  night-service bell  exit ephone 11  exit ephone-dn 3  name “”  exit ephone 11  button  1:3  restart  exit ephone 11  restart  exit

Configuring Shared ephone-dn

Shared ephone-dn is putting the same ephone-dn on one or more ephones

ephone 2  mac-address CDCD.CDCD.CDCD  type CIPC  auto-line  exit ephone 2  button  2:3 1:2  restart  exit ephone-dn 3  no name  exit ephone-dn 2  exit

ephone 2  mac-address CDCD.CDCD.CDCD  type CIPC  auto-line  exit ephone 2  button  2:3 1:2  restart  exit ephone-dn 3  no name  exit ephone-dn 2  exit

Extension Mobility in CME

Extension Mobility allows users to log into any IP Phone configured for extension mobility. This allows users to sit at different desks and still use their own extension. These settings follow the users who logs in with Ext Mobility – extension, Caller ID, speed dials, etc.

Ext Mobility is configured by creating logout profiles. This defines which extension a phone will have when a user logs out of the phone. 

Users profiles must include their extension, caller ID and speed dials when setting up Ext Mobility. Ext Mobility prompts users to login with their username and a password using the keypad.

CCNA Collaboration – Notes, Chapter 6

CICD – Ch6 – Understanding the CME Dial Plan

Configuring Analog Voice Ports

  • FXS – Foreign Exchange Station
    • Connects to end station – phone, fax machine, modem
    • Physical ports on router, typically a WIC card
    • show voice port summary


 CME#sh voice port summary

=============== == ============ ===== ==== ======== ======== ==
0/3/0 --fxs-lsupdorm on-hookidle y 
0/3/1 --fxs-lsupdorm on-hookidle y 
50/0/11efxs upup on-hookidle y 
50/0/12efxs upup on-hookidle y 
  • Configuration areas
    • Call Progress Tone
    • Signaling
      • Loop Start
        • Default
        • Signal by completing a circuit
          • Off Hook handset, DC power
        • Typically used with analog devices – phones, fax, modems
      • Ground Start
        • Configured
        • Signal a connection by grounding 2 wires in the cable temporarily
        • Tpyically used when connecting to PBX
CME(config)#voice-port 0/3/0 
CME(config-voiceport)#cptone ?
locale 2 letter ISO-3166 country code

  • Caller ID Information
    • Allows other devices in the system to receive caller ID name and number
CME(config-voiceport)#station-id name ANALOG PHONE
CME(config-voiceport)#station-id number 5553000
  • FXO – Foreign Exchange Office
    • Trunk port to PSTN Central Office (CO) or PBX
    • Uses same commands as FXS for signal, station and ID
    • 2 new configurations
      • Dial-type
        • Choose DTMF (Dual Tone Multifrequency)
        • Pulse dialing or rotary dial
        • dial-type <dtmf/pulse>
      • Ring-number
        • ring number <#>
          • Number of rings that should pass before router answers an incoming call to the FXO port
          • Default – one ring
            • Causes router to answer immediately

Configuring Digital Voice Ports

  • T1 or E1 connection, VWIC interface card on a router
  • Must configure to operate, router needs to know which signaling to use
  • CAS – T1/E1 – ds0-group
  • CCS – ISDN – pri-group


Configuration example – CAS

CME(config)# controller t1 0/0/0
CME(config-controller)# frame esf
CME(config-controller)# linecode b8zs
CME(config-controller)# clock source line
CME(config-controller)# ds0-group 1 timeslots 1-24 type …

  • Config info must match the provider
  • Frame in US is most likely esf
  • Clock source – where to get clocking from
  • ds0-group – configure the line as T1 CAS
    • Allow to enter specific number of time slots to provision
    • Single T1 can be provisioned for different purposes
    • Command automatically create voice ports


Configuration example – CCS

CME(config)# isdn switch-type primary-ni
CME(config)# controller t1 0/1/0
CME(config-controller)# pri-group timeslots 1-24

T1 signaling – slot 24
E1 signaling – slot 17

CLI Delivered interface Serial 0/1/3:23  description T1  shutdown  isdn switch-type primary-ni  no shutdown  isdn protocol-emulate network  exit voice-port 0/1/3:23  description T1  exit card type T1 0 1 network-clock-participate wic 1 isdn switch-type primary-ni network-clock-select 6 T1 0/1/3 controller T1 0/1/3  description T1  shutdown  pri-group timeslots 1-24  no shutdown  exit

CLI Delivered interface Serial 0/1/3:23  description T1  shutdown  isdn switch-type primary-ni  no shutdown  isdn protocol-emulate network  exit voice-port 0/1/3:23  description T1  exit card type T1 0 1 network-clock-participate wic 1 isdn switch-type primary-ni network-clock-select 6 T1 0/1/3 controller T1 0/1/3  description T1  shutdown  pri-group timeslots 1-24  no shutdown  exit

Dial Peers

  • Static route for the voice network

  • Manually enter destinations
  • Define voice reachability information
    • Phone numbers that can be dialed
  • Can assign one or more numbers to analog devices
  • Allows for wildcards to define a range of phone numbers


2 Types of Dial Peers

  • POTS – Plain Old Telephone Service
    • Reachability for traditional voice devices connected to FXS, FXO, E&M
  • VOIP Dial Peer
    • Reachability for VOIP connection
    • Reachable through IP address
POTS Dial Peer from CCP

POTS Dial Peer from CCP

VOIP Dial Peer from CCP

VOIP Dial Peer from CCP

Voice Call Legs

  • Connection to or from a voice gateway from a POTS of VOIP resource


  • Call legs need return paths to provide 2 way calling
  • Dial peers provide reachability information (phone number) and path audio must travel
  • Call legs are matched both inbound and outbound

Configuring POTS Dial Peer

  • Dial-peer voice <tag> pots
    • Tag can be any number. Must be unique on the router
    • Try and have tag match the dial peer phone number


CME(config)#dial-peer voice 1101 pots
CME(config-dial-peer)#destination-pattern 1101
CME(config-dial-peer)#port ?
<0-50>Voice interface slot #
CME(config-dial-peer)#port 0/?
<3-3>Voice interface SubUnit #
CME(config-dial-peer)#port 0/3/0
CME(config)#dial-peer voice 2201 pots
CME(config-dial-peer)#destination-pattern 2201 
CME(config-dial-peer)#port 0/3/1 

Show Output

CME#sh dial-peer voice summary 
dial-peer hunt 0
20001potsup up 2001$0 50/0/1
1101 potsup up 1101 0up 0/3/0
2201 potsup up 2201 0up 0/3/1

Disable Digit Strip

Disable Digit Strip

Configuring VOIP Dial Peers

Example shows wildcard being used. This is explained in next section

CME(config)#dial-peer voice 3000 voip
CME(config-dial-peer)#destination-pattern 3...
CME(config-dial-peer)#session target ipv4:
CME(config-dial-peer)#codec ?
aacldAACLD 90000 bps 
clear-channelClear Channel 64000 bps (No voice capabilities: data transport only)
g711alaw G.711 A Law 64000 bps
g711ulaw G.711 u Law 64000 bps
g722-48G722-48K 64000 bps - Only supported for H.320<->H.323 calls
g722-56G722-56K 64000 bps - Only supported for H.320<->H.323 calls
g722-64G722-64K 64000 bps
g723ar53 G.723.1 ANNEX-A 5300 bps (contains built-in vad that cannot be disabled)
g723ar63 G.723.1 ANNEX-A 6300 bps (contains built-in vad that cannot be disabled)
g723r53G.723.1 5300 bps
g723r63G.723.1 6300 bps
g726r16G.726 16000 bps
g726r24G.726 24000 bps
g726r32G.726 32000 bps
g728 G.728 16000 bps
g729br8G.729 ANNEX-B 8000 bps (contains built-in vad that cannot be disabled)
g729r8 G.729 8000 bps
gsmamr-nbGSM AMR-NB 4750 - 12200 bps (contains built-in vad that cannot be disabled)
ilbc iLBC 13330 or 15200 bps 
isac iSAC 10 to 32 kbps (variable bit-rate)
transparenttransparent; uses the endpoint codec
CME(config-dial-peer)#codec g711ulaw

  • Session Target – Similar to pots, port command
    • Syntax = ipv4:<ip> , dns:<name>
  • Codecs must match between the 2 routers
    • Default on VOIP is G.729


dial-peer voice 3000 voip  no shutdown  no vad  description 3000  destination-pattern 3...  preference 0  session target ipv4:  codec g711ulaw  exit

dial-peer voice 3000 voip  no shutdown  no vad  description 3000  destination-pattern 3…  preference 0  session target ipv4:  codec g711ulaw  exit

Dial Peer Wildcards

Wildcard Description
. (period) Matches any digits dialed 0-9 or *

20.. matches 2000 – 2099
+ (plus) Matches 1 or more instances of preceding digit

5+23 matches 5523, 55523, 55523, etc

up to 32 digits – max length of dialable numbers
[ ] (brackets) Match range of digits

[1-3] matches 122, 222, 322

^ does not match

[^1-3] matches 422,522 – 922, *22
T Matches any number of dialed digits

0-32 digits
, (comma) Insert 1 second pause between dialed digits

PSTN Dial Plan

North America example

Dial Plan



7 digit dialing

6 periods


10 digit dialng

2 periods, followed by 6


11 digit dialing


Service numbers

411, 611, 911


International Dialing

CME(config)# dial-peer voice 90 pots
CME(config-dial-peer)# description Service Dialing
CME(config-dial-peer)# destination-pattern 9[469]11
CME(config-dial-peer)# forward-digits 3
CME(config-dial-peer)# port 1/0:1
CME(config-dial-peer)# exit
CME(config)# dial-peer voice 91 pots
CME(config-dial-peer)# description 10-Digit Dialing
CME(config-dial-peer)# destination-pattern 9[2-9]..[2-9]......
CME(config-dial-peer)# port 1/0:1
CME(config-dial-peer)# exit
CME(config)# dial-peer voice 92 pots
CME(config-dial-peer)# description 11-Digit Dialing
CME(config-dial-peer)# destination-pattern 91[2-9]..[2-9]......
CME(config-dial-peer)# forward-digits 11
CME(config-dial-peer)# port 1/0:1
CME(config-dial-peer)# exit
CME(config)# dial-peer voice 93 pots
CME(config-dial-peer)# description International Dialing
CME(config-dial-peer)# destination-pattern 9011T
CME(config-dial-peer)# prefix 011
CME(config-dial-peer)# port 1/0:1
CME(config-dial-peer)# exit
  • Forward digits <#>
    • Specify the number of right justified digits to forward
    • Ex: 9[469]11, forward-digit 3
      • Only sends 411, 611 or 911. Leading 9 is dropped
  • prefix <#>
    • Add any digit in front of dialed number before routing the call

Private Line Automatic Ringdown

  • PLAR relies on existing dial plan to complete a call
  • Automatically dial a number as soon as the port detects an off hook signal


voice-port 0/0/0
 connection plar <extension>

Router Call Processing and Digit Manipulation

  • Most specific destination pattern always wins
  • When a match is found router immediately processes the call

555[1-3]… –> Matches 3000 numbers
5551… –> Matches 1000 numbers
   This is more specific

If the following 3 dial peers existed and 5551234 is dialed, the 3rd dial peer would process the call

  1. 555[1-3]…
  2. 5551…
  3. 5551
    The 234 would be dropped, this is the most specific match as it was found before 234 were dialed

show dialplan number <dialed digits>
– Test to see which dial peer would be matched

Matching Inbound and Outbound Dial Peers

Dial Peer 0

  • Cannot change
  • Default settings
    • Any codec
    • No DTMF Relay
    • IP Precedence 0
    • Voice Activity Detection Enabled
    • No RSVP Support
    • Fax-rate voice
    • No application support (IVR)
    • No DID support

Digit Manipulation

  • Process of adding or removing digits from a dialed number to help reach a destination
  • POTS
    • prefix-digit > add before dialed digits
    • forward-digits <number> > Forward the number of right most digits
    • [no] digit-strip > Enable/disable digit stripping
  • Global
    • num-exp and digit set digits > Transform any dialed number match the string into digits specified in the string
  • Global, POTS or VOIP
    • Voice translation profile
      • Allow to configure a translation profile consisting of up to 15 rules
      • Created globally
      • Applied to dial peer, similar to an ACL

Additional Commands

preference – if desintation pattern is the same, choose the path with the lower preference
– Used for failover
– 0 is better than 1

num-exp 0 <dn> – Anytime 0 is dialed, send to configured directory number

Digit Manipulation, POTS Dial Peers Order of Operations

  1. num-exp
  2. Automatic digit strip, pots dial peer
  3. Voice translation profile
  4. Prefix digit
  5. Forward digits


CME Class of Restriction (COR)

  • Prevent users from calling certain number
    • International calls
    • High cost numbers (1-900)
    • Certain internal phones from reaching executives numbers

Steps for COR Creation

  1. Define COR Tags to use
  2. Create outbound COR list
  3. Create inbound COR list
  4. Assign outbound COR list
  5. Assign inbound COR list


COR Configuration – CLI

CME#conf t
Enter configuration commands, one per line.End with CNTL/Z.
CME(config)#dial-peer cor CUSTOM
CME(config-dp-cor)#name 911
CME(config-dp-cor)#name LOCAL
CME(config-dp-cor)#name LD
CME(config)#dial-peer cor list 911-CALL 
CME(config-dp-corlist)#member 911
CME(config-dp-corlist)#dial-peer cor list LOCAL-CALL
CME(config-dp-corlist)#member LOCAL 
CME(config-dp-corlist)#dial-peer cor list LD-CALL 
CME(config-dp-corlist)#member LD
CME(config-dp-corlist)#dial-peer cor list 911-ONLY
CME(config-dp-corlist)#member 911
CME(config-dp-corlist)#dial-peer cor list 911-LOCAL
CME(config-dp-corlist)#member 911
CME(config-dp-corlist)#member LOCAL
CME(config-dp-corlist)#dial-peer cor list 911-LOCAL-LD
CME(config-dp-corlist)#member 911
CME(config-dp-corlist)#member LOCAL
CME(config-dp-corlist)#member LD
CME(config)#do sh run | s dial-peer
dial-peer cor custom
 name 911
 name LOCAL
 name LD
dial-peer cor list 911-CALL
 member 911
dial-peer cor list LOCAL-CALL
 member LOCAL
dial-peer cor list LD-CALL
 member LD
dial-peer cor list 911-ONLY
 member 911
dial-peer cor list 911-LOCAL
 member 911
 member LOCAL
dial-peer cor list 911-LOCAL-LD
 member 911
 member LOCAL
 member LD
CME(config)#dial-peer voice 10 pots
CME(config-dial-peer)#cor list outgoing 911-CALL
% Invalid input detected at '^' marker.

CME(config-dial-peer)#corlist outgoing 911-CALL
CME(config-dial-peer)#dial-peer voice 11 pots 
CME(config-dial-peer)#corlist outgoing LOCAL-CALL
CME(config-dial-peer)#dial-peer voice 12 pots
CME(config-dial-peer)#corlist outgoing LD-CALL 
CME(config)#ephone-dn 1
CME(config-ephone-dn)#corlist incoming 911-ONLY
CME(config-ephone-dn)#ephone-dn 2
CME(config-ephone-dn)#corlist incoming 911-LOCAL
CME(config-ephone-dn)#ephone-dn 3
CME(config-ephone-dn)#corlist incoming 911-LOCAL-LD


COR Rule

  • If no outgoing cor list is applied the call is always routed
  • If no incoming cor list is applied the call is always routed

CCNA Collaboration – Notes, Chapter 5

CICD – Ch5 – Managing Endpoints and End Users in CME


  • CME has 3 levels of users
    • System Admin
      • Authority of all aspects of CME system and phone features
    • Customer Admin
      • Perform MACD (Move, Adds, Changes, Delete) to phones and users
      • No system level access
    • Phone User
      • Customize aspects of their own phone
      • Speed dial, extension mobility, search user directory
      • Login with username and password

Creating Users

  • System admin
    • Must be privilege level 15
    • Can use CLI
    • GUI – CME built in (no screenshots)
router(config)#hostname CME
CME(config)#username cisco priv 15 password cisco
CME(config)#line vty 0 4
CME(config-line)#login local
CME(config-line)#transport input all
CME(config)#int gi0/0
CME(config-if)#ip add
CME(config-if)#no shut
CME(config-if)#ip http server
CME(config)#ip http secure-server 
% Generating 1024 bit RSA keys, keys will be non-exportable...[OK]
CME(config)#ip http authentication local
  • GUI CME – Customer Admin
    • Configure > System Parameter > Administrators Login Account


CME Terminology

  • ephone – ethetnet phone
    • Physical phone hardware
    • How SCCP phones are referenced in the CLI
    • Device type, MAC address
  • ephone-dn – Directory number
    • Number assigned to SCCP phone

Associate user to ephone-dn

  • Provides caller ID for internal calls
  • Builds system directory
  • Presence status monitoring
    • On/off hook, unregistered
  • Applied ephone-dn config to ephone when user logs in with extension mobility


CME Phones

  • CME supports SIP and SCCP

Config – CLI


ephone --> create phone
ephone-dn --> create SCCP DN
telephony-service --> telephony config


voice register pool --> create phone
voice register dn --> create SIP DN
voice register global --> telephony config
  • Add phone manually with CLI or GUI
  • Can enable Auto Registration
    • Phones automatically get added to CME


Home Lab screenshots and CLI

  • Base config has already been shown above or in previous notes. CCP was installed on my Win10 VM.

Configuring CME

CLI sent to router

 no auto-reg-ephone

New Options that are available

Enabling SIP

CLI sent to router

Enabling VOIP Settings, SIP settings

voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 no supplementary-service h450.2
 no supplementary-service h450.3
 no supplementary-service h450.7
 no supplementary-service sip moved-temporarily
 no supplementary-service sip refer
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
registrar server

Telephony Settings

Enable router to support SIP and SCCP endpoints, set number of phones and extensions router can support

Choose IP address for phones to register with. I used the physical interface. Loopback address can also be used. Must be a static address, DHCP does not work

CLI sent to router

 cnf-file location flash:
voice register global
 mode cme
 max-pool 25
 max-dn 50
 tftp-path flash:
 create profile
 create cnf-files
voice register global
 mode cme
 create profile

Phone Images

Not included in the chapter, but CME needs the phone firmware loaded on the flash.

Example Output - done from CLI, TFTP image to router

tftp-server flash:SCCP45.9-4-2SR3-1S.loads alias SCCP45.9-4-2SR3-1S.loads
tftp-server flash:apps45.9-4-2ES26.sbn alias apps45.9-4-2ES26.sbn
tftp-server flash:cnu45.9-4-2ES26.sbn alias cnu45.9-4-2ES26.sbn
tftp-server flash:cvm45sccp.9-4-2ES26.sbn alias cvm45sccp.9-4-2ES26.sbn
tftp-server flash:dsp45.9-4-2ES26.sbn alias dsp45.9-4-2ES26.sbn
tftp-server flash:jar45sccp.9-4-2ES26.sbn alias jar45sccp.9-4-2ES26.sbn
tftp-server flash:term45.default.loads alias term45.default.loads
tftp-server flash:term65.default.loads alias term65.default.loads
 load 7945 SCCP45.9-4-2SR3-1S.loads
 create cnf-files

Creating SCCP Extension and Phone, creating and assigning user to phone

CLI sent to router

ephone-dn 1 dual-line
 number 2001
ephone 1
 mac-address C062.6BD3.33CE
 type 7945
ephone 1
 username phone1
ephone-dn 1
 name Phone One
ephone 1

Creating SIP Extension and Phone, creating and assigning user to phone

CLI sent to router


voice register pool 1 
 type 7945 
 create cnf-files 
voice register global 
 mode cme 
 create profile 
voice register pool 1 
 username Phone2 password password 
 number 1 dn 1 
voice register dn 1 
 name Phone Two 
voice register pool 1 

CCNA Collaboration – Notes, Chapter 4

CICD – Ch4 – CME Administration


  • CCNA Covers configuring CME using CCP (Cisco Configuration Professional)
    • I will include screenshots of the GUI and also show the command line that gets generated
    • CME can also be configured by using it’s own GUI
      • HTML and java based
      • Loaded onto flash with a .tar file
      • Allows add/change of phones, modifying dial plan, configuring hunt groups, etc.
  • CCP configures all major aspects of CME
    • CCP can also be used for LAN, WAN and security features
  • CCP must discover devices before it can configure them
    • Discovery includes finding out information about the hardware, software, interfaces and modules


CCP CME Features

  • CUBE – Cisco Unified Border Element
    • Telepohony gateway for IP to IP services
      • Ex: IP-TSP – IP Telephony Service Provision
    • NAT/PAT services
    • Billing, security, call admision control, qos, SIP negotiation
  • CUCME – Standalone CME
  • SRST – Survivable Remote Site Telephony
    • Allows phones to register to router (gateway) if they lose connectivity to CUCM
  • CME as SRST
    • SRST with full CME features

CCNA Collaboration – Notes, Chapter 3

CICD – Ch3 – Cisco IP Phone


  • IP Phones require the following
    • POE – Power Over Ethernet
    • Voice VLAN
    • DHCP
  • Phone has 3 port switch built into it
    • Port 1, connects to switch
    • Port 2, phone ASIC
    • Port 3, connects to PC


Power Over Ethernet, POE

  • Phones must receive power from a source
    • Switch POE
    • Power patch panel
    • POE injector
    • Power brick
  • POE is the ability to send electricity over ethernet
    • Centralized power distribution
      • Switches are generally on some type of backup power (UPS, generator)
    • Don’t need a power outlet at the phone
      • Outlets may not be where phones are being places
  • Standard, IEEE
    • 802.3af
      • 15-25 watts
    • POE+
      • 802.3at, 51 watts


Output from a switch

Home_Switch#sh power inline 


Interface AdminOper Power DeviceClass Max


--------- ------ ---------- ------- ------------------- ----- ----

Fa0/1 auto off0.0 n/a n/a 15.4 

Fa0/2 auto off0.0 n/a n/a 15.4 

Fa0/3 auto off0.0 n/a n/a 15.4 

Fa0/4 auto off0.0 n/a n/a 15.4 

Fa0/5 auto off0.0 n/a n/a 15.4 

Fa0/6 auto on 12.0IP Phone 7945 3 15.4 

Fa0/7 auto off0.0 n/a n/a 15.4 

Fa0/8 auto off0.0 n/a n/a 15.4 



Voice VLAN

  • Cisco recommends having a dedicated vlan for voice
  • VLAN = Broadcast domain = IP Subnet
  • Trunk
    • Allow multiple vlans across a single physical interface
    • Also known as, tagging
    • 802.1q = standard
    • ISL = Cisco Proprietary
  • Voice vlan allows interface to become a multi-vlan access port
    • PC connects to phone, phone connects to switch
    • PC sends traffic untagged = access vlan
    • Phone sends traffic tagged = voice vlan
  • Phones receive voice vlan information through CDP neighbor
  • Configuration, switch

*Create layer 2 vlan on the switch

vlan <#>

name DATA

vlan <##>

name VOICE


spanning-tree bpduguard enable —> This command is not referenced in the book, but I mention it here as a best practice. This is a global command that will affect portfast enabled ports. BPDU Guard disables any interface that receives a BPDU into the interface. This is helpful is someone decides to create a loop by plugging in both ethernet ports on the phone into the switch


*Configure interface connected to a phone

interface <int> —> Go into the interface configuration

switchport access vlan <#> —> assign the access (data) vlan to the interface

switchport voice vlan <##> —> assign the voice vlan to the interface

spanning-tree portfast —> immediately bring interface into forwarding state, bypass spanning-tree listening and learning states

switchport mode access —> statically configure the interface as an access port. Default is to dynamically determine based on what plugged into the interface. Could either be trunk or access

Home_Switch(config)#vlan 20

Home_Switch(config-vlan)#name VOICE



Phone Boot Process

  1. Phone connects to ethernet, if switch supports POE, the phone powers on
  2. Switch delivers voice vlan to phone through CDP. Phone starts tagging traffic with correct vlan information
  3. Phone broadcasts a DHCP Request
    1. Broadcasts are contained within a layer 3 vlan. Configuration can be added to the layer 3 SVI (Switches Virtual Interface) to relay DHCP request to a DHCP server if the server lives on a different subnet. ip helper-address <ip>
    2. Asks for an IP on it’s voice vlan
  4. DHCP server respinds with DHCP Offerr
    1. Phone access the offer if there is no duplicate address
    2. Offer contains: Default gateway, DNS Information, domain name
    3. Required from DHCP: Option 150. Option 150 contains information on the TFTP server, more on this later.
  5. Phone contacts TFTP server and downloads configuration file. The file contains valid CME or CUCM servers
  6. Phone registers with CME or CUCM


Router DHCP Configuration

  • DHCP is required for phones (and endpoints for that matter) to get an IP address and be able to communicate on the network
  • Below is an example configuration that can be done on a Cisco router

Global config:

ip dhcp excluded-address <start> <end>

ip dhcp pool <name>

network <ip> <subnet>

default router <ip>

dns-server <ip>

option 150 ip <ip>

Interface config:

interface vlan <int>

ip helper-address <ip>

Actual configuration

Home_Switch(config)#int vlan 20

Home_Switch(config-if)#ip add

Home_Switch(config-if)#ip helper-address

Home_Switch(config-if)#no shut

Home_Switch(config)#ip dhcp excluded-address

Home_Switch(config)#ip dhcp pool VOICE




Home_Switch(dhcp-config)#option 150 ip 


Network Time Protocol – NTP

  • Provides a clocking source
  • Display the correct time and date on phones
  • Get the correct date and time for voicemails
  • Accurate Call Detail Records (CDR), explained in later chapters
    • Track calls on the network
  • Security features
  • Tag log messages
  • Stratum levels, how accurate is the time source
    • Level 1 is the best



ntp server <ip> —> where to get source of time from

clock timezone <timezone> —> What timezone is the device in

ntp master <stratum> —> Tells router to provide time

ntp server prefer

clock timezone EST -5

clock summer-time EDT recurring


Phone Registration

  • Phones use SCCP or SIP for signaling
  • SCCP, Skinny
    • Cisco proprietary voice signaling protocol to control phones
  • SIP, Session Initiation Protocol
    • IETF standard voice signaling protocol
    • Lightweight alternative to H.323
  • Phones identify themselves with MAC address
    • Talks to CME or CUCM (call processors)
    • Call processor will send XML file to phone with its configuration
    • Configuration includes: device language, firmware version, call processing IPs, ports #s, etc.
      • Softkey layout
  • Signaling protocol is used for majority of phone functionality
    • Dial tone, digit collecting, on/off hook conditions


Quality of Service – QOS

  • For VOIP to operate successfully, voice must have priority over data traffic
  • QOS definition: Ability for the network to provide better or special service to a set of users and application at the expense of other users and applications
  • Voice traffic is time sensitive
  • Voice should get first access to bandwidth
    • Router queues other traffic in time of congestion
  • Problems QOS is trying to solve
    • Lack of bandwidth
    • Delay
    • Fixed delay
    • Variable delay
    • Jitter (delay variation)
    • Packet loss
  • Voice Traffic Requirements
    • Voice is predictable, if you know which codec is being used you’ll be able to calculate how much bandwidth is required
    • These are the maximum thresholds, lower is better
      • End to end delay – 150ms
      • Jitter – 30ms
      • Packet loss – 1%
    • Video has same requirements, just requires more bandwidth

QOS Mechanisms

  • Best Effort – Default, no QOS
    • First come, first serve
  • IntServ – Reservation Model
    • Resource Reservation Protocol (RSVP)
    • Provides guaranteed bandwidth
    • Has scalability problems, each router must track the traffic flow
  • DiffServ – Most popular and flexible model
    • Configure every device to respond with a variety of QOS methods based on traffic classes
    • DSCP
    • Note: This CCNA does not go into the level of detail that I was expecting. I’ll write up another post that’ll be a more in-depth on QOS

QOS Tools

  • Classification and Marking – Identify and mark packets
  • Congestion Management – QOS Queuing strategies
  • Congestion Avoidance – Drop packets before congestion occurs
  • Policing and Shaping – Give hard or soft limits on how much of a specified traffic is allowed
  • Link Efficiency – compression mechanisms

CCNA Collab book goes into Link Efficency and Queuing Algorithms. If you want to know about the others, drop a comment and I’ll write some more details around the others

Link Efficiency

  • Payload compression
    • Compress app data from being sent across the WAN
  • Header compression
    • Eliminate redundant fields of the header
    • RTP Header Compression, compressed RTP (cRTP). Go from 40 bytes down to 2 bytes, 4 bytes with error correction
  • Link Fragmentation and Interleaving – LFI
    • Addresses serialization delay by chopping larger packets into smaller ones
    • Used on PPP or frame relay connections

Queuing Algorithms

  • WFQ – Weighted Fair Queuing
    • Tries to balance available bandwidth for all senders
    • Default on serial interfaces
  • CBWFQ – Class Based WFQ
    • Guarantees specific amounts of bandwidth for various traffic classes
  • LLQ – Low Latency Queuing
    • Add a priority queue
    • Similar to CBWFQ

Applying QOS

  • Input Actions
    • Classification
    • Marking
    • Policing
  • Output Actions
    • Congestion management
    • Marking
    • Congestion avoidance
    • Shaping
    • Policing
    • Compression
    • Fragmentation and Interleaving



  • Simplified mechanism to deploy QOS
  • Deploys template based on Ciso’s QOS best practice
  • Uses CDP to detect IP phone to apply QOS settings

AutoQOS Benefits

  • Reduced time to deploy
  • Configuration consistency
  • Reduced deployment cost
  • Allows manual tuning

AutoQOS, steps before deployment

  • Establish trust boundary – which endpoints do you trust markings from
  • Devices can mark traffic with different QOS classification
  • Ex: Phone marks all traffic as high priority (EF)
    • Note, DSCP was not covered in this book. I’ll write a future blog post
  • Phone has ability to strip marking PC’s set

AutoQOS Config

  • Single command under interface
  • Does not need to be applied on every device
    • This is according to the book. Real life, deploy QOS everywhere in a controller maner
  • Before commands are entered, check to make sure bandwidth statements are correct
  • AutoQOS uses a LLQ model

Global Config

Home_Switch(config)#auto qos ?

  srnd4QoS configurations based on solution reference network design 4.0


Home_Switch(config-if)#auto qos ?

  classifyConfigure classification for untrusted devices

  trust     Trust the DSCP/CoS marking

  video     Configure AutoQoS for video devices

  voip    Configure AutoQoS for VoIP

Home_Switch(config-if)#auto qos voip ?

  cisco-phone    Trust the QoS marking of Cisco IP Phone

  cisco-softphoneTrust the QoS marking of Cisco IP SoftPhone

  trust          Trust the DSCP/CoS marking

Home_Switch(config-if)#auto qos voip cisco-phone 

Home_Switch(config-if)#do sh run int fa0/7

Building configuration…

Current configuration : 226 bytes


interface FastEthernet0/7

 srr-queue bandwidth share 1 30 35 5

 priority-queue out 

 mls qos trust device cisco-phone

 mls qos trust cos

 auto qos voip cisco-phone 




Additional Output generated

CCNA Collaboration – Notes, Chapter 2

CICD – Ch2 – Unified Collaboration

Unified Collaboration

  • More than just VOIP
  • Bring all collaboration into a seamless framework
    • Includes: voice, video and data
  • CICD covers core solutions
    • CME – Communications Manager Express
    • CUCM – Cisco Unified Communications Manager
    • CUC – Cisco Unity Connection
    • CUIMP – Cisco Unified IM and presences
  • There are more outside of this exam
    • Contact Center
    • WebEx
    • Telepresence
    • Spark

Communications Manager Express

  • Runs IP telephony on an ISR (integrated services router)
  • ISR can terminate analog and digital circuits
    • FXO, FXS, T1
      • FXO – Office, connects to POTS circuits
      • FXS – Station, connects to endpoints, analog phones, fax, modem
    • Supports VOIP endpoints
    • Features: conference calls, video, automatic call distribution
  • Platform dependent on number of supported phones
    • ISR4451, can handle 450 phones
  • All in one device for controlling phones and trunking to PSTN

CME Features

  • Call processing and device control
  • CLI or GUI based configuration
  • Local directory service
  • Computer Telephony Integration (CTI)
    • Control a device remotely. Example, have jabber make calls using your desk phone
  • Trunk to other voice systems
  • Integration with Unity Express
    • Runs on modules installed in router
    • Provides voicemail features

CME and IP Phones

  • CME controls virtually every action performed on phone
  • SCCP (skinny) or SIP used for signaling protocols to allow CME to communicate and control the phone
  • CME instructs phones where to connect, establish calls
    • Not involved in the RTP stream of traffic
    • CME is not required after a call has been setup
    • Exception: If call is going to PSTN that is attached to the CME router


  • Linux based, VMware guest
  • Provides core device control, call routing, permissions, features, connectivity to outside applications
  • GUI based administration

CUCM Key Features

  • Full audio and video support
  • Appliance based operation
  • VMware install, on supported hardware
  • Redundant server cluster
    • Single CUCM cluster can handle 40000 phones running SIP or SCCP
    • Megacluster can handle 80000 phones
  • Intercluster trunk and Voice Gateway control
  • Built in disaster recovery
    • Done through sFTP
  • Directory service support and integration
    • Built-in or LDAP
  • Database replication
    • Publisher replicates read-only database to subscribers in the cluster
  • Run Time Data
    • Intercluster communication signaling (ICCS)
    • All servers in cluster form TCP connectionn for ICCS over TCP ports 8002-8004
    • Assuming 10,000 busy hour calls = 1.5Mbps per server
  • Changes to database are performed on Publisher
  • Standard cluster supports single publisher and 19 subscribers
    • 8 for call processing
  • 11 other subscribers can be TFTP, conference bridge, MOH, annunciater
  • TFTP is critical for phone and gateway operations
    • Bulk of config is downloaded from TFTP
  • Best practice is to pull publisher out of all call processing and let subscribers do the work
  • Phones can list 3 redundant CUCM servers for failover purposes

Unity Connection

  • Make any message retrievable from any voice enabled device or application
  • Supports 20000 voicemail boxes
  • Runs as VMware guest
  • Active / Active setup, publisher, subscriber
  • Features
    • Personal call transfer rules, speech recognition

Unity Connection Key Features

  • Appliance based
  • 20k mailboxes per server (cluster?)
  • Access voicemail from anywhere
  • LDAP integration
  • Microsoft Exchange support
  • Voicemail Profile for Internet Mail (VPIM)
    • Integration of voicemail services from difference vendors
  • Active / active high availability
    • Publisher / subscriber cluster
    • Both accept client requests
  • CUCM + CUC communicate using SCCP or SIP

IM and Presence – IMP

  • Jabber client
  • Shows full presence status
  • Enterprise instant messaging
    • Jabber, Extensible Communication Platform (XCP)
    • Industry standard, works with different IM clients
  • Message compliance
    • Logging
    • Encryption with TLS
  • Interdomain federation
    • Connects to other domains outside of your business
    • Business to business communication
  • Secure Messaging
    • TLS, IPSec
  • Jabber XCP extensibility
    • File sharing, app sharing, video conferencing
  • IMP integrated with CUCM supports 45000 users
  • IMP, IM only – 75000 users
    • No integration with CUCM


  • IM Client: peer to peer client, multiuser chat, persistent chat
  • LDAP integration
  • Integrated with CUCM, able to see On/Off hook status
  • Start voice and video calls (HD)
  • Soft phone
  • CTI support, control desk phone from jabber client

Video Communication Server (VCS) + Telepresence Management Suite (TMS)

  • Call types
    • Internal desktop calls
      • Between 2 endpoints with camera
      • Registered to CUCM
    • Telepresence Calls
      • Using TP endpoints
    • External Video Calls
      • Business to business

VCS Control

  • Self contained video endpoint call control system for customers without CUCM

VCS Expressway

  • 2 servers – communicate as a trusted relationship
    • Core – inside firewall
    • Edge – deployed in DMZ
    • Perform firewall traversal
    • Allows callers to signal CUCM to setup video call with endpoints registered to CUCM

TMS – TP Management Suite

  • Manages scheduling of TP Meetings
  • For administrators
    • Provisioning tool to deploy TP endpoints across multiple locations within organization
    • Central admin of all TP infrastructure resources
    • Real time management of all conferences
    • Phone book sync
    • Reporting
  • For users
    • Ease of scheduling
      • Outlook / exchange integration
      • one button to push
    • Contact – phone book from multiple sources