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CCIE Voice Test

Instructions:

Answer 50 questions in 15 minutes.
If you are not ready to take this test, you can study here.
Match each statement with the correct term.
Don't refresh. All questions and answers are randomly picked and ordered every time you load a test.

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 are the commands to configure NTP in IOS?
Enterprise Parameters --> Organization Top Level Domain|Enterprise Parameters --> Cluster Fully Qualified Domain Name
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.

2. 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 os truly unsurpassed. I have opened up a wealth of opportunities for myself and my family and
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
...

3. What is Mobile Voice Access?
...
Call-manager-fallback or telephony-service mode SRST
#after-hours block pattern
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.

4. What CM Service needs to be start in Serviceability for MOH to work?
Via the voice hunt-group parallel command
IP Voice Media Streaming App
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
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.

5. If you have an H323 Gateway at a siet where the phones are controlled by CUCM - what do you need from a dial-peer perspective?
It's best to strip digits at the voice port.
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
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.
Interface f0/0|no ip address|no shut||interface f0/0.10|encap dot1q 10 native --> !!! DATA VLAN !!!|ip address 10.10.100.1 255.255.255.0|interface f0/0.20|encap dot1q 20 --> !!! VOICE VLAN !!!|ip address 10.10.200.1 255.255.255.0

6. What does Display-IE do?
Sends the Calling Name.
(config)#service dhcp|(config)#no ip dhcp conflict logging|(config)#ip dhcp excluded-address <low IP> <hi IP>|(config)#ip dhcp-pool <name>|(dhcpconfig)#network <network name> </nn or mask>|(dhcpconfig)#domain-name <domain>|(dhcpconfig)#dns-server <IP
SIP Route Pattern over a SIP Trunk.
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)

7. How do you configure SIP phones to be able to call other SIP phones via SIP URI?
#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
SIP Route Pattern over a SIP Trunk.
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary

8. When using drop-through-option What is the max number of huntgroups?
1 ... this is not optional!
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
'Use before the ... so XXXX

9. If you need terminate and reoriginate all calls through the CUBE - what do you need to do.
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
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
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
...

10. How do you use the # as a string terminator within a SIP Dial Rule?
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
Precede the # with a > ... so 9011*>#
At the CLI: utils ntp status
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

11. What are the commands to configure SIP phones in CME?
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
When they are explicitly matched in a destination-pattern in a dial-peer.
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

12. '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
...
Configure it on the route group through the route list - then it will be local to the route list.
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
(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|(config-controller)#no shut||(config)#voice service voip|(conf-voi-serv)#sip|(conf-serv-sip)#bind

13. What does ccm-manager config server [IP]</ip> do?
Automatically configures the MGCP GW for you.
In CUCM - configure CFUR to point to it's E164 number.
When configuring TEHO.
To notify SIP Phones of NTP

14. If you're using Called/Calling Party Transformations at the Route Pattern and tranformations at the route list ... What does the RP affect and What does the RL affect?
Outbound dial-peers
Int vlan 130|ip address 10.10.101.1 255.255.255.0|int vlan 240|ip address 10.10.201.1 255.255.255.0
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
Re-enter the mgcp bind control and mgcp bind media commands.

15. How do you allow the Calling Name to be sent to the PSTN on a router?
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
TRUE
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files

16. How do you disable KPML?
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
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)
Assign SIP Dial Rules
Call Simulator. You can use to validate path from router to the PSTN.

17. What are the commands to set up the PRI?

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18. How do you ensure an H323 GW always uses the Sub as primary and Pub as a backup?
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.
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
voice register pool|and|voice register dn
'Use before the ... so XXXX

19. When dealing with an MGCP GW there is no intelligence on the router. How would you strip an incoming Called Number from 10 to 4 digits?
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
Put them in a route list.
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.

20. When you make a call and it results in a secondary dial-tone - What is this a symptom of?
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.
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
Need to Know||Anytime you have an external call - it needs to match on a route pattern.

21. In CME Where is the Calling Name derived from?
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.
Matches any length dialed number and truncates it to 4 digits.
The name configuration field in ephone-dn and voice register dn
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

22. What terminology translates to SRST?
Reroute when there is a WAN Outage.
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
Dial-peers
TRUE

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

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24. What command is needed on a gateway for it to invoke the DSP to convert multicast MOH stream to TDM?
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.
CUCM Service Parameters --> Call Park Reversion Timer
Ccm-manager music-on-hold

25. How do you configure AAR?
'Use before the ... so XXXX
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
#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

26. What must you do for a BACD script to work on a CME router?
Enable MOH on the CME.||#telephony service|#moh music-on-hold.au
'Service Parameters --> Auto Answer Timer
Phone number followed by domain name.|i.e. 3006@ipxcme.com
Re-enter the mgcp bind control and mgcp bind media commands.

27. How do you support multiple codecs on a dialpeer?
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
In CUCM - configure CFUR to point to it's E164 number.
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
Apply voice-class codec

28. What is the Service Parameter to allow the phone that is picking ip to auto answer?
Auto Call Pickup Enabled
Sh ip rsvp reservation||sh sccp connections
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
Call-manager-fallback or telephony-service mode SRST

29. How do you prioritize route groups?
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.
Interface f0/0|no ip address|no shut||interface f0/0.10|encap dot1q 10 native --> !!! DATA VLAN !!!|ip address 10.10.100.1 255.255.255.0|interface f0/0.20|encap dot1q 20 --> !!! VOICE VLAN !!!|ip address 10.10.200.1 255.255.255.0
Put them in a route list.
IP Voice Media Streaming App

30. How do you set the inter-digit timeout for SIP phones in CME?
(config)#voice register dialplan
Find the GUI Files - show flash | i .html|ip http server|ip http path flash:/<PATH TO FILES>||telephony service|web admin system name admin pass cisco|dn-webedit
On CUCM -- UPTSAL||Create Your Users|Phone - set owner id|Create SIP Trunk|SIP Trunk Security Profile||Application Server|Licensing||||On CUPS||Work Left to right
#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

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

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32. How do you configure an MGCP GW? (router side)
The Called/Calling Transformations are superceded.
(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
On CUCM it's identical to adding an H323 GW.
The name configuration field in ephone-dn and voice register dn

33. What does the Transfer On-hook Enabled Service Parameter do?
Enterprise Parameters --> Organization Top Level Domain|Enterprise Parameters --> Cluster Fully Qualified Domain Name
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
Allows you to transfer by only pressing the Transfer button once.
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False

34. 'How do you inform a SIP phone of NTP information?
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
Allows you to transfer by only pressing the Transfer button once.
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
(config)#isdn switch-type primary-net5 (or whatever's provided)|(config)#network-clock-participate wic 0||(config)#controller e1 0/0/0|(config-controller)#pri-group timeslot 1-3|(config-controller)#no shut||(config)#int s0/0/0:15 (this is created aft

35. How is NTP sych setup in CUCM?
Outbound dial-peers
Enable MOH on the CME.||#telephony service|#moh music-on-hold.au
voice register pool|and|voice register dn
CUCM OS Administration Settings --> NTP Servers

36. In Gatekeeper CAC how do you restrict a specific endpoint?
Put them in a route list.
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
Sends the Calling Name.
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast

37. How do you prevent H323 caller-id updates to CUCM
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
When they are explicitly matched in a destination-pattern in a dial-peer.
#voice service voip| #no supplementary-service h225-notify cid-update
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.

38. How do you ensure that G.711 only is used?
When multiple sites use the same route pattern - do your digit manipulation on a route list.
When they are explicitly matched in a destination-pattern in a dial-peer.
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
Debug ephone moh

39. Does CUCM support RSVP natively?
On CUCM -- UPTSAL||Create Your Users|Phone - set owner id|Create SIP Trunk|SIP Trunk Security Profile||Application Server|Licensing||||On CUPS||Work Left to right
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
Sh ip rsvp reservation||sh sccp connections
Via the voice hunt-group parallel command

40. What does rule 1 /^.*(....$)/ /1/ do?
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
The Called/Calling Transformations are superceded.
Matches any length dialed number and truncates it to 4 digits.
CUCM Service Parameters --> Automated Alternate Routing Enable --> True

41. 'If you want to make changes to any softkeys where do you do it?
...
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
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.
Bottom up - to prevent glare.

42. With the gw-priority command - does higher or lower priority take precedence?
(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

#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
Higher

43. What is call-forward pattern used for?
CUCM OS Administration Settings --> NTP Servers
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.
Interface f0/0|no ip address|no shut||interface f0/0.10|encap dot1q 10 native --> !!! DATA VLAN !!!|ip address 10.10.100.1 255.255.255.0|interface f0/0.20|encap dot1q 20 --> !!! VOICE VLAN !!!|ip address 10.10.200.1 255.255.255.0
Call forwarding between voip to voip (when CUBE is in play)

44. What does KPML do?
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast

45. When do you use translate called? Translate calling?
#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
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
Dial-peers
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

46. Where do you use VIA zone?
sh ephone reg|and|debug voip dialpeer
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
(config)#service dhcp|(config)#no ip dhcp conflict logging|(config)#ip dhcp excluded-address <low IP> <hi IP>|(config)#ip dhcp-pool <name>|(dhcpconfig)#network <network name> </nn or mask>|(dhcpconfig)#domain-name <domain>|(dhcpconfig)#dns-server <IP
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

47. How do you convert a full PRI 9config'd by ccm-manager config server) back to a Fractional PRI?

#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
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
Higher

48. 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.
Enable MOH on the CME.||#telephony service|#moh music-on-hold.au
Bottom up - to prevent glare.
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3

49. Which takes precedence Device Locations or Device Pool Locations?
Max-dn|max-ephone|ip source-address
Device Pool Locations
Configure it on the route group through the route list - then it will be local to the route list.
Assign SIP Dial Rules

50. How do you enable AAR system wide?
In the CUCM 7.0.1 Release Notes ... search for restore.
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
Outbound dial-peers