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CCIE Voice Test

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Answer 50 questions in 15 minutes.
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Match each statement with the correct term.
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This is a study tool. The 3 wrong answers for each question are randomly chosen from answers to other questions. So, you might find at times the answers obvious, but you will see it re-enforces your understanding as you take the test each time.

1. What is the bit rate for a G.729 call excluding layer 2?
8kb/s
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
Route Patterns --> Route Lists --> Route Groups --> Gateways/Trunks|RPs point to Route Lists which contain one or more Route Groups which contain one or more Gateways/Trunks.
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.

2. If you can't use a tech-prefix or default tech-prefix how do you get around the fact that CUCM does not register it's endpoint to GK?
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
Use the alias static configuration||alias static (IP of CUCM) gkid (GK ID) gateway voip ras (IP of CUCM) e164 (dn) e164 (dn) e164 (dn)

3. On an MGCP GW - if you select Top Down under Channel Selection Order - What does that mean?
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
Top Down means that channel 1 will be the first channel used to place outgoing calls.
#voice service voip| #no supplementary-service h225-notify cid-update
Re-enter the mgcp bind control and mgcp bind media commands.

4. What are the base telephony-service commands for CME?
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
Sh ccm-manager
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files

5. How do you configure class of service (CoS) in CUCM? CME?
You need pots dial-peers for incoming calls and you need a voip dial-peer pointing to the CUCM so the H323 GW and connect the call to those phones.
#test voice translation rule 1 <input to test>
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
At the CLI: utils ntp status

6. On an H323 GW - how do you adjust the timers for redundancy hunting?
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
Sh ip rsvp reservation||sh sccp connections
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3

7. How do you verify where MOH is being served up from?

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8. 'By Nov. 16 - 2010||I am happy confident and thankful that I am now a CCIE Voice Certified engineer! My technical knowledge of the Cisco UC portfolio is truly unsurpassed. I have opened up a wealth of opportunities for myself and my family and
...
It matches a route pattern| Which points to a route list| Which point to a route group| Which contains a GW or Trunk|*Then it needs to match on an incoming VOIP diap-peer (incoming called -number .)
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
Assign a logout-profile to the ephone.

9. Is the order of the MRGs in an MRGL significant?
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
It's best to strip digits at the voice port.

10. How do you allow the Calling Name to be sent to the PSTN on a router?
To notify SIP Phones of NTP
#after-hours block pattern
When configuring TEHO.
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie

11. 'If you're not getting a DHCP address from CUCM what then?
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
#voice service voip| #no supplementary-service h225-notify cid-update
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify

12. What terminology translates to SRST?
#voice class h323 1| #h225 timeout tcp establish 3||#dial-peer voice 3000 voip| #destination-pattern 3...| #voice-class h323 1| #session target ipv4:(SUB IP)| #dtmf-relay h245-alpha||#dial-peer voice 30001 voip| #destination-pattern 3...| #voice-clas
Reroute when there is a WAN Outage.
Via the voice hunt-group parallel command
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!

13. On the RDP What is the CSS used for?
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
CUCM Service Parameters --> Call Park Reversion Timer
Mobile Voice Access

14. When can you not use a Standard Local Route Group?
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
Enable MOH on the CME.||#telephony service|#moh music-on-hold.au
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
When configuring TEHO.

15. Where can you assign the AAR Group?
Sh ccm-manager
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.

16. Where do you use VIA zone?
Gatekeeper - zone local GK cisco.com 10.1.6.1 - zone local CUCM cisco.com invia GK outvia GK - zone local BR2 cisco.com invia GK outvia GK
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
voice translation-profile|and|voice translation-rule
Voice register dn 1| number 4002||voice register pool 1|id mac 0022.0022.0022|type 7961ge|number 1 dn 1|dtmf-relay rtp-nte|no vad

17. How do you create a trunk on the switch side?
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
Re-enter the mgcp bind control and mgcp bind media commands.
Create a Route Pattern for the local site and the remote site:||RP-LOC --> PT-REM --> RL-TEHO-LOC||RP-REM --> PT-LOC --> RL-TEHO-REM
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!

18. If the PSTN is presenting on channel 1 - should you configure CUCM to search 'top down' or 'bottom up'?
Allows you to transfer by only pressing the Transfer button once.
Bottom up - to prevent glare.
To enable two-stage dialing.
Ip domain-name ipexpert.com|network-clock-participate wic 0|isdn switch-type primary-ni||controller t1 0/0/0|pri-group timeslots 1-3 - 24 service mgcp|no shut||int serial 0/0/0:23|isdn bind-l3 ccm-manager|ccm-manager mgcp|ccm-manager redundant-host 1

19. When using drop-through-option What is the max number of huntgroups?
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
1 ... this is not optional!

20. What are the commands to configure an H323 GW?
#int lo0|#h323-gateway voip bind srcaddr [IP]||#isdn switch-type primary-ni|#network-clock-particpate wic0||#controller t1 0/0/0|#pri-group timeslots 1-3||#int serial 0/0/0:23|#isdn outgoing display-ie|#isdn b-chan selection order [use opposite of wh
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
Call-manager-fallback or telephony-service mode SRST
24kb/s

21. What are the commands to configure a SIP phone in CUCME?
voice translation-profile|and|voice translation-rule
Service Parameter --> Block OffNet To OffNet Transfer --> True

Voice register dn 1| number 4002||voice register pool 1|id mac 0022.0022.0022|type 7961ge|number 1 dn 1|dtmf-relay rtp-nte|no vad

22. In CME Where is the Calling Name derived from?
Re-enter the mgcp bind control and mgcp bind media commands.
Service Parameter --> Block OffNet To OffNet Transfer --> True
Voice register dn 1|number 3001|voice register pool 1|number 1 dn 1|id mac 001b.ac02.14de|type 7961|dtmf-relay rtp-nte|description 3214-3005 !!! PHONE BANNER !!!|codec g711u
The name configuration field in ephone-dn and voice register dn

23. What CM Service needs to be start in Serviceability for MOH to work?
Sh ccm-manager
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
IP Voice Media Streaming App
Mobile Voice Access

24. How do you use an ephone template?
Apply it to the ephone.
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
When configuring TEHO.
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow

25. How do you ensure an H323 GW always uses the Sub as primary and Pub as a backup?
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
The name configuration field in ephone-dn and voice register dn
Create 2 voip dial-peers - one the Sub and one to the Pub (the Pub with a preference of 1). Then create a voice class h323and set the h225 timeout value so it will hunt between the two.
SIP Dial Rules

26. What are the commands to manually configure an MGCP gateway?
#isdn switch-type primary-ni|#network-clock-participate wic 0|#controller t1 0/0/0|#pri-group timeslots 1-3 service mgcp||#ccm-manager mgcp ccm-manager redundant-host [IP]|#ccm-manager fallback-mgcp (for srst)|#ccm-manager moh||#mgcp|#mgcp call-agent
The + character is not supported on a VOIP dial-peer.||So you'll need to add back on the POTS dial-peer before it gets to the PSTN via a voice translation-rule.
Sh ip rsvp reservation||sh sccp connections
Re-enter the mgcp bind control and mgcp bind media commands.

27. Where can you find the command to restore any of the default IP Phone Services if they happen to have been deleted?
Automatically configures the MGCP GW for you.
Auto Call Pickup Enabled
Allows you to transfer by only pressing the Transfer button once.
In the CUCM 7.0.1 Release Notes ... search for restore.

28. What is call-forward pattern used for?
Create a Route Pattern for the local site and the remote site:||RP-LOC --> PT-REM --> RL-TEHO-LOC||RP-REM --> PT-LOC --> RL-TEHO-REM
Call forwarding between voip to voip (when CUBE is in play)
Call Simulator. You can use to validate path from router to the PSTN.
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.

29. Privacy is enabled system-wide in CUCM by default. (T or F)
Re-enter the mgcp bind control and mgcp bind media commands.
TRUE
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
Create 2 voip dial-peers - one the Sub and one to the Pub (the Pub with a preference of 1). Then create a voice class h323and set the h225 timeout value so it will hunt between the two.

30. How do you configure CUCM redundancy on an H323 gateway?
#voice class h323 1| #h225 timeout tcp establish 3||#dial-peer voice 3000 voip| #destination-pattern 3...| # voice-class h323 1| #session target ipv4:10.1.5.3 (sub)| #dtmf-relay h245-alpha||#dial-peer voice 30001 voip| #destination-pattern 3...| #pre
#voice class h323 1| #h225 timeout tcp establish 3||#dial-peer voice 3000 voip| #destination-pattern 3...| #voice-class h323 1| #session target ipv4:(SUB IP)| #dtmf-relay h245-alpha||#dial-peer voice 30001 voip| #destination-pattern 3...| #voice-clas
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
Put them in a route list.

31. 'How do you inform a SIP phone of NTP information?
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
Create 2 voip dial-peers - one the Sub and one to the Pub (the Pub with a preference of 1). Then create a voice class h323and set the h225 timeout value so it will hunt between the two.
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup

32. How do you change modes in voice register global?
Sends the Calling Name.
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.

33. What are the basic ephone-dn and ephone commands?
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
Ip domain-name ipexpert.com|network-clock-participate wic 0|isdn switch-type primary-ni||controller t1 0/0/0|pri-group timeslots 1-3 - 24 service mgcp|no shut||int serial 0/0/0:23|isdn bind-l3 ccm-manager|ccm-manager mgcp|ccm-manager redundant-host 1
Ephone-dn 1 <dual / octo>|number 3001|description 3214-3001 !!! PHONE BANNER !!!||ephone 1|mac-address 001b.d4c6.ca99|button 1:1|type 7961|max-calls-per-button 5|busy-trigger-per-button 3
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones

34. What should you configure before entering auto qos?
Bandwidth
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Need to Know||Anytime you have an external call - it needs to match on a route pattern.

35. What are the commands to configure NTP in IOS?
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.

Single Number Reach (Mobile Connect)
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify

36. What are two caveats to using the ccm-manager config server</ip> command?

#isdn switch-type primary-ni|#network-clock-participate wic 0|#controller t1 0/0/0|#pri-group timeslots 1-3 service mgcp||#ccm-manager mgcp ccm-manager redundant-host [IP]|#ccm-manager fallback-mgcp (for srst)|#ccm-manager moh||#mgcp|#mgcp call-agent
(config)#isdn switch-type primary-net5|(config)#network-clock-participate wic 0||(config)#Controller e1 0/0/0|(config-controller)#pri-group timeslot 1-3 !!!Get the PRI up!!!|(config-controller)#no shut||(config)#int s0/0/0:15|(config-if)#isdn outgoin
CUCM OS Administration Settings --> NTP Servers

37. How do you disable KPML?
Assign SIP Dial Rules
Create a Route Pattern for the local site and the remote site:||RP-LOC --> PT-REM --> RL-TEHO-LOC||RP-REM --> PT-LOC --> RL-TEHO-REM
Voice register dn 1|number 3001|voice register pool 1|number 1 dn 1|id mac 001b.ac02.14de|type 7961|dtmf-relay rtp-nte|description 3214-3005 !!! PHONE BANNER !!!|codec g711u
#after-hours block pattern

38. How do you block calls under call-manager-fallback?
#voice class h323 1| #h225 timeout tcp establish 3||#dial-peer voice 3000 voip| #destination-pattern 3...| # voice-class h323 1| #session target ipv4:10.1.5.3 (sub)| #dtmf-relay h245-alpha||#dial-peer voice 30001 voip| #destination-pattern 3...| #pre
Voice register dn 1| number 4002||voice register pool 1|id mac 0022.0022.0022|type 7961ge|number 1 dn 1|dtmf-relay rtp-nte|no vad
#after-hours block pattern
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM

39. On an MGCP GW - how could you see the primary and backup CUCM servers?
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
Show ccm-manager
Voice register global|mode cme|source-address <IP>|max-dn <NUM>|max-pool <NUM>|tftp-path flash:/<PATH>|create profile||NOTE: You have to do a create profile every time you change the voice register pool

40. When using GK redundancy - what must you do to ensure both SUB and PUB use port 1720 for call-signalling.

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41. How do do configure TEHO?
1 ... this is not optional!
(config)#ip domain-name <NAME>|(config)#isdn switch-type primary-ni|(config)#network-clock-participate wic 0||(config)#controller t1 0/0/0|(config-controller)#pri-group timeslots 1-3 - 24 service mgcp|(config-controller)#no shut||(config)#interface s
Auto Call Pickup Enabled
Create a Route Pattern for the local site and the remote site:||RP-LOC --> PT-REM --> RL-TEHO-LOC||RP-REM --> PT-LOC --> RL-TEHO-REM

42. How do you prevent toll fraud on CUCM?
voice translation-profile|and|voice translation-rule
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
Put them in a route list.
Trunk Port Method (3750 or ESW)||interface fa0/1|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10|switchport voice vlan 20|spanning-tree portfast

43. What would force you to use telephony-service to configure SRST?
Service Parameters --> Mobile Voice Access Number
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files

44. How do you maintain 4-digit dialing to phones that are running in an SRST scenario (WAN outage)?

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45. How do you ensure no inter-digit timeout fro SIP phones within CUCM?
Transfer to external is blocked by default.|To ensure only transfer to internal is allowed - make sure only the following tx pattern is under telephony-service||#telephony service| #transfer-pattern ....
Sends the Calling Name.
...
SIP Dial Rules

46. Where is it best to manipulate digits for inbound calls?

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47. What is the Enable Enterprise Feature Access Service Parameter used for?
Precede the # with a > ... so 9011*>#
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
To enable two-stage dialing.
Voice register dn 1|number 3001|voice register pool 1|number 1 dn 1|id mac 001b.ac02.14de|type 7961|dtmf-relay rtp-nte|description 3214-3005 !!! PHONE BANNER !!!|codec g711u

48. How do you set up redundancy on outbound dial-peers on an H323 gateway?
The + character is not supported on a VOIP dial-peer.||So you'll need to add back on the POTS dial-peer before it gets to the PSTN via a voice translation-rule.
...
Mobile Voice Access
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)

49. How do you use the # as a string terminator within a SIP Dial Rule?
Precede the # with a > ... so 9011*>#
Voice register dn 1| number 4002||voice register pool 1|id mac 0022.0022.0022|type 7961ge|number 1 dn 1|dtmf-relay rtp-nte|no vad
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST
Dial-peers

50. How do you enable Extension Mobility for a device in CME?
Apply voice-class codec
Assign a logout-profile to the ephone.
1 ... this is not optional!
Sh ccm-manager

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