Test your basic knowledge |

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 basic SCCP Commands fro telephony-service in CUCME?

Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files

2. Name 4 useful show commands for active calls.
Phone number followed by domain name.|i.e. 3006@ipxcme.com
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
Single Number Reach (Mobile Connect)

3. When do you use translate called? Translate calling?
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
Via the voice hunt-group parallel command
When multiple sites use the same route pattern - do your digit manipulation on a route list.
(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

4. What does KPML do?
Reroute when here is WAN congestion.
Configure it on the route group through the route list - then it will be local to the route list.
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
Dial-peers

5. When can you not use a Standard Local Route Group?
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
When configuring TEHO.
The name configuration field in ephone-dn and voice register dn
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.

6. What is CSIM?
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 Simulator. You can use to validate path from router to the PSTN.

'Use before the ... so XXXX

7. If the PSTN is presenting on channel 1 - should you configure CUCM to search 'top down' or 'bottom up'?
Show ccm-manager
Matches any length dialed number and truncates it to 4 digits.
Configure it on the route group through the route list - then it will be local to the route list.
Bottom up - to prevent glare.

8. What is a best practice for digit manipulation - in regards to H323 GWs?
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
Bottom up - to prevent glare.
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
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.

9. How do you ensure an H323 GW always uses the Sub as primary and Pub as a backup?
Debug isdn q931
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.
Apply voice-class codec
Call Simulator. You can use to validate path from router to the PSTN.

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

11. What are the commands to configure a SIP phone in CUCME?
#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
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.
Dial-peers
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

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

13. How do you create a trunk on the router side?
Bandwidth
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
Device Pool Locations
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

14. What is the full E164 format?
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
Device Pool Locations
Bandwidth

15. 'If you want to make changes to any softkeys where do you do it?
(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
Dtmf-relay h245-alpha
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
(config)#voice register dialplan

16. What are the basic ephone-dn and ephone commands?
1 ... this is not optional!
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
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

17. How would you enable security on a GK?
Call Simulator. You can use to validate path from router to the PSTN.
CUCM OS Administration Settings --> NTP Servers
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
sh ephone reg|and|debug voip dialpeer

18. How do you prioritize route groups?
Assign SIP Dial Rules
Bottom up - to prevent glare.
Put them in a route list.
Assign a logout-profile to the ephone.

19. In CME Where is the Calling Name derived from?
(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
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
(config)#voice register dialplan
The name configuration field in ephone-dn and voice register dn

20. Where can you assign the AAR Group?
1 ... this is not optional!
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
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.
Top Down means that channel 1 will be the first channel used to place outgoing calls.

21. What's the best way to do digit manipulation on an IOS gateway?
Allows you to transfer by only pressing the Transfer button once.
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
voice translation-profile|and|voice translation-rule

22. On the RDP What is the CSS used for?
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
Mobile Voice Access
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!

23. How is NTP sych setup in CUCM?
CUCM OS Administration Settings --> NTP Servers
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
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.
On CUCM it's identical to adding an H323 GW.

24. On a gatekeeper trunk - how do you ensure the Sub is used as primary and Pub as backup?
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)
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
Higher

25. How do you allow H323 calls to be preserved should the primary H323 GW fail?
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
When configuring TEHO.
'Use before the ... so XXXX

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?
To enable two-stage dialing.
'Use before the ... so XXXX
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 ....
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. What dtmf-relay type do you use for an H323 GW?
CUCM Service Parameters --> Call Park Reversion Timer
Dtmf-relay h245-alpha
Debug isdn q931
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 .)

28. How do you configure SIP phones to be able to call other SIP phones via SIP URI?
SIP Route Pattern over a SIP Trunk.
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
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
On CUCM it's identical to adding an H323 GW.

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

Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
voice translation-profile|and|voice translation-rule
Master/Slave relationship. CUCM controls it.

30. How do you configure a SIP Trunk? (router side)
Ccm-manager music-on-hold
(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
(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
CUCM Service Parameters --> Automated Alternate Routing Enable --> True

31. What are the commands to configure SRST in fallback?
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
Service Parameters --> Mobile Voice Access Number
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files

32. What types of digit manipulation can you perform at the route pattern?
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
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 the Calling Name.
CUCM Service Parameters --> Automated Alternate Routing Enable --> True

33. When a CUCM device dials a number - what happens?
'Service Parameters --> Auto Answer Timer
Service Parameter --> Block OffNet To OffNet Transfer --> True
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 .)
#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

34. How do you use the # as a string terminator within a SIP Dial Rule?
Precede the # with a > ... so 9011*>#
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
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
SERVICE PARAMETER> Automated Alternate Routing Enable > True

35. How do you configure phone ports on a 3750?
Max-dn|max-ephone|ip source-address
Enable MOH on the CME.||#telephony service|#moh music-on-hold.au
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.

36. When configuring a Gatekeep-controlled trunk - what do you do with the Wait for Far End H.245 TCS option?
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
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
In the CUCM 7.0.1 Release Notes ... search for restore.

37. What are the 3 mandatory commands within call-manager-fallback?
Dtmf-relay h245-alpha
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
Via the voice hunt-group parallel command
Max-dn|max-ephone|ip source-address

38. How do you configure a gateway to register with gatekeeper?
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
#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
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)

39. If you need terminate and reoriginate all calls through the CUBE - what do you need to do.
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
IP Voice Media Streaming App
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323

40. Describe the relationship between route patterns and end devices.
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.
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
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
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify

41. What is Mobile Voice Access?
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
#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
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
Call forwarding between voip to voip (when CUBE is in play)

42. What are the commands to manually configure an MGCP gateway?
#voice service voip| #no supplementary-service h225-notify cid-update
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
Debug isdn q931
#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

43. What terminology translates to SRST?
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
Reroute when there is a WAN Outage.
Max-dn|max-ephone|ip source-address

44. How do you create a trunk on the switch side?
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
Higher

45. How do you test a Voice Translation Rule?
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.
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.
#test voice translation rule 1 <input to test>
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs

46. How do you configure class of service (CoS) in CUCM? CME?
voice translation-profile|and|voice translation-rule
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
Outbound dial-peers
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

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

48. How do you configure SRST?
Call-manager-fallback or telephony-service mode SRST
It's best to strip digits at the voice port.
Re-enter the mgcp bind control and mgcp bind media commands.
Reroute when there is a WAN Outage.

49. 'If you're not getting a DHCP address from CUCM what then?
'Service Parameters --> Auto Answer Timer
Service Parameters --> Mobile Voice Access Number
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
Matches any length dialed number and truncates it to 4 digits.

50. What is the Enable Enterprise Feature Access Service Parameter used for?
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
Phone number followed by domain name.|i.e. 3006@ipxcme.com
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST
To enable two-stage dialing.