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. How do you place SCCP and SIP phones into a single huntgroup?
Via the voice hunt-group parallel command
#test voice translation rule 1 <input to test>
Allows you to transfer by only pressing the Transfer button once.
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.

2. SNR is also known as?
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
Mobile Connect
Sends the Calling Name.

3. What does an H323 GW require that MGCP GWs do not?
Dial-peers
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 ....
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
Bottom up - to prevent glare.

4. When do you use translate called? Translate calling?
The name configuration field in ephone-dn and voice register dn
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.

5. What are the commands to configure a MGCP Gateway? (router)
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)
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
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
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

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

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7. What does KPML do?
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
Default is SRST||mode cme changes to CME|no mode cme changes back to 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)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie

8. On an MGCP GW - if you select Top Down under Channel Selection Order - What does that mean?
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
Top Down means that channel 1 will be the first channel used to place outgoing calls.
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
Single Number Reach (Mobile Connect)

9. In Gatekeeper CAC how do you restrict a specific endpoint?
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
Single Number Reach (Mobile Connect)
Sh ccm-manager
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.

10. 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?
Bandwidth
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.

11. How do you setup AutoRegistration in CUCM?
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 ipv4:10.1.5.3 (sub)| #dtmf-relay h245-alpha||#dial-peer voice 30001 voip| #destination-pattern 3...| #pre
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP

12. What are the base telephony-service commands for CME?
Reroute when there is a WAN Outage.
Outbound dial-peers
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
At the CLI: utils ntp status

13. How do you see multicast packets being sent?
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
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
Debug ephone moh

14. What dtmf-relay type do you use for an H323 GW?
Higher
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
Dtmf-relay h245-alpha
(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

15. What are the commands to configure SIP phones in CME?
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
voice translation-profile|and|voice translation-rule
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
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False

16. What are the commands to configure a T1/E1 PRI?
(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
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
When multiple sites use the same route pattern - do your digit manipulation on a route list.
#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

17. With the gw-priority command - does higher or lower priority take precedence?
...
#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
Higher
IP Voice Media Streaming App

18. 'How do you inform a SIP phone of NTP information?
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
Apply voice-class codec
Sh ip rsvp reservation||sh sccp connections
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group

19. Maximum Wait Time for Desk Pickup?
'Use before the ... so XXXX
CUCM Service Parameters --> Call Park Reversion Timer
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3

20. How do you enable AAR?
Reroute when here is WAN congestion.
SERVICE PARAMETER> Automated Alternate Routing Enable > True
#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
Call Simulator. You can use to validate path from router to the PSTN.

21. 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?
Call Simulator. You can use to validate path from router to the PSTN.
(config)#sh cdp neigh detail
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.
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.

22. What is the Service Parameter to allow the phone that is picking ip to auto answer?
Sh ccm-manager
'Use before the ... so XXXX
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
Auto Call Pickup Enabled

23. How do you prevent toll fraud on CME?

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)
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 ....
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

24. What are the commands to configure a SIP phone in CUCME?
'Service Parameters --> Auto Answer Timer
Device Pool Locations
Bandwidth
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

25. How do you configure SIP phones to be able to call other SIP phones via SIP URI?
Reroute when here is WAN congestion.
SIP Route Pattern over a SIP Trunk.
Sends the Calling Name.
When multiple sites use the same route pattern - do your digit manipulation on a route list.

26. On the RDP What is the Rerouting CSS used for?
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
Single Number Reach (Mobile Connect)
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
Put them in a route list.

27. What commands are needed to configure the voice register pool in CME?
The Called/Calling Transformations are superceded.
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
'Use before the ... so XXXX
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST

28. On an H323 GW - how do you adjust the timers for redundancy hunting?
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)

29. On an MGCP GW - how could you see the primary and backup CUCM servers?
Show ccm-manager
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.

30. How do you prevent H323 caller-id updates to CUCM
When they are explicitly matched in a destination-pattern in a dial-peer.
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
#voice service voip| #no supplementary-service h225-notify cid-update
TRUE

31. What is Phone NTP Reference used for?
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
To notify SIP Phones of NTP
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 ....
Ccm-manager music-on-hold

32. How do you get around relying on DNS for your CUCMs?
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
CUCM Service Parameters --> Call Park Reversion Timer
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
SERVICE PARAMETER> Automated Alternate Routing Enable > True

33. When can you not use a Standard Local Route 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
SIP Dial Rules
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.
When configuring TEHO.

34. What's the best way to do digit manipulation on an IOS gateway?
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
voice translation-profile|and|voice translation-rule
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files

35. How do you configure SRST?
Apply voice-class codec
Automatically configures the MGCP GW for you.
Call-manager-fallback or telephony-service mode SRST
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files

36. What are the commands to configure SRST in fallback?
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
Enable MOH on the CME.||#telephony service|#moh music-on-hold.au
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
Re-enter the mgcp bind control and mgcp bind media commands.

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

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38. What are the commands to ensure an H323 GW always uses the Sub as primary and Pub as a backup?
(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 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
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

39. How do you ensure that G.711 only is used?
#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
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
Assign SIP Dial Rules
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable

40. Name 4 useful show commands for active calls.
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
SIP Route Pattern over a SIP Trunk.
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary

41. How do you block calls under call-manager-fallback?
Sh ip rsvp reservation||sh sccp connections
Show ccm-manager
#after-hours block pattern
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit

42. How do you configure CUBE?

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43. How do you use the # as a string terminator within a SIP Dial Rule?
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
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*>#
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast

44. An MGCP gateway serving as an SRST router requires what?
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
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
Redundant dial plan. One on CUCM and one on the routers via dial-peers.

45. What CUCM services should you activate?
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
Dtmf-relay h245-alpha
...
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.

46. What do you need to do to activate the CME GUI?
Call forwarding between voip to voip (when CUBE is in play)
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
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.

47. What's the best way to perform digit manipulation on a route group?
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
Max-dn|max-ephone|ip source-address
Configure it on the route group through the route list - then it will be local to the route list.
(config)#voice register dialplan

48. How do you set up redundancy on outbound dial-peers on an H323 gateway?
Sh ip rsvp reservation||sh sccp connections
'Service Parameters --> Auto Answer Timer
voice translation-profile|and|voice translation-rule
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)

49. How do you change modes in voice register global?
Max-dn|max-ephone|ip source-address
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST
SERVICE PARAMETER> Automated Alternate Routing Enable > True

50. What does rule 1 /^.*(....$)/ /1/ do?
Matches any length dialed number and truncates it to 4 digits.
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
Apply it to the ephone.
Call forwarding between voip to voip (when CUBE is in play)