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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. 'If you want to make changes to any softkeys where do you do it?
#test voice translation rule 1 <input to test>
Bandwidth
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
Automatically configures the MGCP GW for you.

2. On an MGCP GW - if you select Top Down under Channel Selection Order - What does that mean?
#voice service voip| #no supplementary-service h225-notify cid-update
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
Top Down means that channel 1 will be the first channel used to place outgoing calls.

3. What are the commands to configure a T1/E1 PRI?
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
(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. In a SIP Dial Rule how do you represent a dialed * instead of it being used as a wildcard?

5. What terminology translates to AAR?
Reroute when here is WAN congestion.
Show ccm-manager
8kb/s
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs

6. How do you configure SIP phones to be able to call other SIP phones via SIP URI?
SIP Route Pattern over a SIP Trunk.
(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
Allows you to transfer by only pressing the Transfer button once.
#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

7. How do you create a trunk on the router side?

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 Simulator. You can use to validate path from router to the PSTN.
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary

8. On an H323 GW - how do you adjust the timers for redundancy hunting?
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
The name configuration field in ephone-dn and voice register dn
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
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)

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

10. How do you support multiple codecs on a dialpeer?
24kb/s
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
Apply voice-class codec
TRUE

11. What is Phone NTP Reference used for?
Single Number Reach (Mobile Connect)
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
To notify SIP Phones of NTP

12. What should you configure before entering auto qos?
When multiple sites use the same route pattern - do your digit manipulation on a route list.
SIP Route Pattern over a SIP Trunk.
#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
Bandwidth

13. With the gw-priority command - does higher or lower priority take precedence?
Higher
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)

14. 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)
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)
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
Debug ephone moh

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

16. What does the Transfer On-hook Enabled Service Parameter do?
8kb/s
Mobile Voice Access
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.
Allows you to transfer by only pressing the Transfer button once.

17. Name 2 commands to verify RSVP functionality.
Sh ip rsvp reservation||sh sccp connections
#after-hours block pattern
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
Dtmf-relay h245-alpha

18. '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
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
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 .)
...
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323

19. When a CUCM device dials a number - what happens?
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 .)
Phone number followed by domain name.|i.e. 3006@ipxcme.com
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP
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.

20. When can you not use a Standard Local Route Group?
When configuring TEHO.
Dial-peers
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
Auto Call Pickup Enabled

21. 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.
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)

22. 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?
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.
On CUCM it's identical to adding an H323 GW.
It's best to strip digits at the voice port.
The Called/Calling Transformations are superceded.

23. How would you enable security on a GK?
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
SERVICE PARAMETER> Automated Alternate Routing Enable > True
#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
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary

24. How do you see the details of calls coming in and out of the PRI?
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
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
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
Debug isdn q931

25. How do you create a trunk on the switch side?
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.
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
(config)#sh cdp neigh detail
Call-manager-fallback or telephony-service mode SRST

26. A show isdn status shows TEI_ASSIGNED. What do you do to get MULTIPLE_FRAME_ESTABLISHED?
Re-enter the mgcp bind control and mgcp bind media commands.
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
When they are explicitly matched in a destination-pattern in a dial-peer.
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.

27. When configuring a Gatekeep-controlled trunk - what do you do with the Wait for Far End H.245 TCS option?
Sh ip rsvp reservation||sh sccp connections
In the CUCM 7.0.1 Release Notes ... search for restore.
(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
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.

28. On the RDP What is the Rerouting CSS used for?
Max-dn|max-ephone|ip source-address
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
Single Number Reach (Mobile Connect)
Matches any length dialed number and truncates it to 4 digits.

29. What terminology translates to SRST?
Mobile Connect
Reroute when there is a WAN Outage.
SIP Route Pattern over a SIP Trunk.
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

30. Need to Know||Anytime you have an external call - it needs to match on a route pattern.
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
Bottom up - to prevent glare.
Need to Know||Anytime you have an external call - it needs to match on a route pattern.

31. What kind of relationship does an MGCP gateway have with CUCM?
Master/Slave relationship. CUCM controls it.
Assign SIP Dial Rules
(config)#voice register dialplan
In CUCM - configure CFUR to point to it's E164 number.

32. What is the full E164 format?
Put them in a route list.
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
(config)#sh cdp neigh detail

33. What does rule 1 /^.*(....$)/ /1/ do?
On CUCM it's identical to adding an H323 GW.
Matches any length dialed number and truncates it to 4 digits.

No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.

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

35. How much bandwidth does a G.729 call including layer 3 require?
24kb/s
'Service Parameters --> Auto Answer Timer
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
The name configuration field in ephone-dn and voice register dn

36. 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?
Precede the # with a > ... so 9011*>#
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)
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
When configuring TEHO.

37. What are the 3 mandatory commands within call-manager-fallback?
Max-dn|max-ephone|ip source-address
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
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.
In CUCM - configure CFUR to point to it's E164 number.

38. What are the commands to configure SRST in fallback?
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
Apply voice-class codec
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.

39. How do you get around relying on DNS for your CUCMs?
Dtmf-relay h245-alpha
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
Ccm-manager music-on-hold
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs

40. On the RDP What is the CSS used for?
#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
'Service Parameters --> Auto Answer Timer
CUCM OS Administration Settings --> NTP Servers
Mobile Voice Access

41. How do you configure a SIP Trunk? (router side)
Apply it to the ephone.
(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
#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
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify

42. What are the commands to ensure an H323 GW always uses the Sub as primary and Pub as a backup?
#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
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
1 ... this is not optional!
#after-hours block pattern

43. How do you configure phone ports on a 3750?
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
Sh ip rsvp reservation||sh sccp connections
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.

44. SNR is also known as?
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
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 ....
Mobile Connect

45. 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?
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)
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
Matches any length dialed number and truncates it to 4 digits.
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323

46. When are digits stripped in a 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.
When they are explicitly matched in a destination-pattern in a dial-peer.
#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
Allows you to transfer by only pressing the Transfer button once.

47. Describe how you configure SIP URI functionality.
Call Simulator. You can use to validate path from router to the PSTN.
Apply voice-class codec

#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM

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

49. How do you allow H323 calls to be preserved should the primary H323 GW fail?
Auto Call Pickup Enabled
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.
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
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False

50. How do you set the inter-digit timeout for SIP phones in CME?
...
Apply it to the ephone.
(config)#voice register dialplan
(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