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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. On an H323 GW - how do you adjust the timers for redundancy hunting?
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 .)
#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
Max-dn|max-ephone|ip source-address
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3

2. Maximum Wait Time for Desk Pickup?
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
Show ccm-manager
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
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.

3. 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
...
In the CUCM 7.0.1 Release Notes ... search for restore.
Assign a logout-profile to the ephone.
Call forwarding between voip to voip (when CUBE is in play)

4. 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?
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
Matches any length dialed number and truncates it to 4 digits.
(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
Put them in a route list.

5. If you need terminate and reoriginate all calls through the CUBE - what do you need to do.
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
Configure it on the route group through the route list - then it will be local to the route list.
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323

6. 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.

Bandwidth
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

7. 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
(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

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

8. Describe the relationship between route patterns and end devices.
...
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
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.

9. Name 4 useful show commands for active calls.

(config)#sh cdp neigh detail
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!

10. When a device is roaming to a different site How does it get the Roaming Sensitive Settings from the Device Pool?
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
Reroute when there is a WAN Outage.
Bottom up - to prevent glare.
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary

11. Calls coming from CUCM to PSTN need what?
Outbound dial-peers
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
'Use before the ... so XXXX
+ [CC 1-3 digits] [National Number max 15 minus CC digits]

12. How do you ensure no inter-digit timeout fro SIP phones within CUCM?
Bandwidth
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
SIP Dial Rules
Reroute when here is WAN congestion.

13. What CM Service needs to be start in Serviceability for MOH to work?
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
IP Voice Media Streaming App
Assign SIP Dial Rules

14. What kind of relationship does an MGCP gateway have with CUCM?
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
Master/Slave relationship. CUCM controls it.
#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

15. How do you configure class of service (CoS) in CUCM? CME?
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
It's best to strip digits at the voice port.
Auto Call Pickup Enabled

16. How do you configure phone ports on an ESW?
In the CUCM 7.0.1 Release Notes ... search for restore.
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
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.
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

17. What does the Transfer On-hook Enabled Service Parameter do?
...
Dtmf-relay h245-alpha
Allows you to transfer by only pressing the Transfer button once.
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

18. On an MGCP GW - if you select Top Down under Channel Selection Order - What does that mean?
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
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.
Bottom up - to prevent glare.
Top Down means that channel 1 will be the first channel used to place outgoing calls.

19. What are the commands to configure a SIP phone in CUCME?
(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
Debug isdn q931
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
To notify SIP Phones of NTP

20. When do you do digit manipulation at the route pattern as opposed to the route list?
When multiple sites use the same route pattern - do your digit manipulation on a route list.
SIP Dial Rules
Allows you to transfer by only pressing the Transfer button once.
...

21. How do you support multiple codecs on a dialpeer?
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
Apply voice-class codec
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.

22. How do you change modes in voice register global?
To notify SIP Phones of NTP
Mobile Voice Access
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False

23. 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?
#test voice translation rule 1 <input to test>
CUCM Service Parameters --> Call Park Reversion Timer
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.
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM

24. Where do you use VIA zone?
Show ccm-manager
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 .)
sh ephone reg|and|debug voip dialpeer
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

25. When setting up SIP URI where do you configure the CUCM's domain name?'
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
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
Enterprise Parameters --> Organization Top Level Domain|Enterprise Parameters --> Cluster Fully Qualified Domain Name
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.

26. 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?
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
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
Sh ccm-manager
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)

27. When are digits stripped in a gateway?
#interface vlan 130 (voice vlan)| #h323-gateway voip interface| #h323-gateway voip id BR2 ipaddr 10.1.6.1| #h323-gateway voip h323-id UK| #h323-gateway voip bind srcaddr 10.1.130.1
When they are explicitly matched in a destination-pattern in a dial-peer.
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
SIP Dial Rules

28. When using drop-through-option What is the max number of huntgroups?
When configuring TEHO.
Master/Slave relationship. CUCM controls it.
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
1 ... this is not optional!

29. How do you prevent toll fraud on CUCM?
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
Re-enter the mgcp bind control and mgcp bind media commands.
24kb/s
Sh ip rsvp reservation||sh sccp connections

30. What is CSIM?
voice translation-profile|and|voice translation-rule
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
Service Parameters --> Mobile Voice Access Number
Call Simulator. You can use to validate path from router to the PSTN.

31. How would you set the the timer for Auto Answer?

32. Where can you assign the AAR Group?
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
...
(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
Assign a logout-profile to the ephone.

33. What is Phone NTP Reference used for?
To notify SIP Phones of NTP
Debug ephone moh
On CUCM it's identical to adding an H323 GW.
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.

34. What are the commands to create vlans on an ESW?
Reroute when there is a WAN Outage.
Sh ip rsvp reservation||sh sccp connections
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
Max-dn|max-ephone|ip source-address

35. What commands are needed to configure the voice register pool in CME?
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
SIP Route Pattern over a SIP Trunk.
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.

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

37. What would force you to use telephony-service to configure SRST?
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
Allows you to transfer by only pressing the Transfer button once.
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable

38. On a gatekeeper trunk - how do you ensure the Sub is used as primary and Pub as backup?
SERVICE PARAMETER > Device Name of GK0controlled Trunk That Will Use Port 1720 > US_Trunk (or whatever you're using)||> Host Name/IP of GK That Will Use RAS UDP Port 1719 > (IP of GK) ||BE SURE TO RESTART THE TRUNK in CUCM and possibly the CUCM Servi
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
Debug isdn q931
#voice service voip| #no supplementary-service h225-notify cid-update

39. How do you enable Extension Mobility for a device in CME?
Call forwarding between voip to voip (when CUBE is in play)
Sh ccm-manager
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
Assign a logout-profile to the ephone.

40. Aside from the Media Resources --> Mobile Voice Access where else do you need to specify the the Mobile Voice Access DN?
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
Service Parameters --> Mobile Voice Access Number
#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

41. In Gatekeeper CAC how do you restrict a specific endpoint?
Matches any length dialed number and truncates it to 4 digits.

Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
(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

42. What do you need to do to activate the CME GUI?
Apply it to the ephone.
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
Reroute when here is WAN congestion.
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False

43. When it comes to H323 gateways - what should you always change - unless explicitly told not to?
Phone number followed by domain name.|i.e. 3006@ipxcme.com
(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
It's best to strip digits at the voice port.
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False

44. How do you prevent H323 caller-id updates to CUCM
#voice service voip| #no supplementary-service h225-notify cid-update
Device Pool Locations
#interface vlan 130 (voice vlan)| #h323-gateway voip interface| #h323-gateway voip id BR2 ipaddr 10.1.6.1| #h323-gateway voip h323-id UK| #h323-gateway voip bind srcaddr 10.1.130.1
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group

45. How do you create a trunk on the router side?
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
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
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
In the CUCM 7.0.1 Release Notes ... search for restore.

46. SNR is also known as?
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
(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
8kb/s
Mobile Connect

47. 'If you want to make changes to any softkeys where do you do it?
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
In CUCM - configure CFUR to point to it's E164 number.
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.

48. On the RDP What is the Rerouting CSS used for?
Debug ephone moh
Single Number Reach (Mobile Connect)
#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
Redundant dial plan. One on CUCM and one on the routers via dial-peers.

49. On the RDP What is the CSS used for?
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
SERVICE PARAMETER> Automated Alternate Routing Enable > True
Mobile Voice Access
#test voice translation rule 1 <input to test>

50. What are the base telephony-service commands for CME?
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
CUCM Service Parameters --> Call Park Reversion Timer
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
voice translation-profile|and|voice translation-rule