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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. Where can you assign the AAR Group?
Re-enter the mgcp bind control and mgcp bind media commands.
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
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
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.

2. What's the difference between AAR and SRST?
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
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
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.
...

3. How do you ensure an H323 GW always uses the Sub as primary and Pub as a backup?
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
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.
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
+ [CC 1-3 digits] [National Number max 15 minus CC digits]

4. How do you support multiple codecs on a dialpeer?
Debug ephone moh
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
Apply voice-class codec
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

5. Where do you use VIA zone?
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
Phone number followed by domain name.|i.e. 3006@ipxcme.com
SIP Route Pattern over a SIP Trunk.
Enterprise Parameters --> Organization Top Level Domain|Enterprise Parameters --> Cluster Fully Qualified Domain Name

6. 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 .)
In CUCM - configure CFUR to point to it's E164 number.
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM

7. What CM Service needs to be start in Serviceability for MOH to work?
Sends every digit to CUCM in real-time with digit-analysis being handled by 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 .)
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
IP Voice Media Streaming App

8. How do you configure a SIP Trunk? (router side)
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP
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)#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

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

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

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
Max-dn|max-ephone|ip source-address
Higher

11. 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!
Dtmf-relay h245-alpha
Outbound dial-peers
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 .)

12. How do you configure SIP phones to be able to call other SIP phones via SIP URI?
Sends the Calling Name.
CUCM Service Parameters --> Call Park Reversion Timer
SIP Route Pattern over a SIP Trunk.
Mobile Voice Access

13. When do you use translate called? Translate calling?
To notify SIP Phones of NTP
'Use before the ... so XXXX
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
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.

14. How do you prevent toll fraud on CME?
(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
Mobile Voice Access
Sends the Calling Name.
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 ....

15. 'If you want to make changes to any softkeys where do you do it?
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
Master/Slave relationship. CUCM controls it.
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
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.

16. What does ccm-manager config server [IP]</ip> do?
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
Automatically configures the MGCP GW for you.

17. How do you configure DHCP in IOS?
SIP Route Pattern over a SIP Trunk.
#test voice translation rule 1 <input to test>
(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 (if no response within 3 seconds continue dial-peer hunting)

18. How do you convert a full PRI 9config'd by ccm-manager config server) back to a Fractional PRI?
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.

No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
#voice service voip| #no supplementary-service h225-notify cid-update

19. How do you see the details of calls coming in and out of the PRI?
Debug isdn q931
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.
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.

20. Where can you find the command to restore any of the default IP Phone Services if they happen to have been deleted?
#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
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
Allows you to transfer by only pressing the Transfer button once.
In the CUCM 7.0.1 Release Notes ... search for restore.

21. What commands are needed to configure the voice register pool in CME?
TRUE
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 configuring TEHO.
When they are explicitly matched in a destination-pattern in a dial-peer.

22. How do you prevent H323 caller-id updates to CUCM
#voice service voip| #no supplementary-service h225-notify cid-update
#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
...

23. 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?
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
Reroute when here is WAN congestion.
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).

24. What are the commands to configure a SIP phone in CUCME?
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
Precede the # with a > ... so 9011*>#
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
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

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

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26. How do you configure phone ports on a 3750?
Re-enter the mgcp bind control and mgcp bind media commands.
voice register pool|and|voice register dn
Apply voice-class codec
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast

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

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

Single Number Reach (Mobile Connect)
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

29. How do you create a trunk on the switch side?
(config)#sh cdp neigh detail
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
(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
SIP Route Pattern over a SIP Trunk.

30. If you're using Called/Calling Party Transformations at the Route Pattern - what happens if you do Digit manipulation at the route list?

The Called/Calling Transformations are superceded.
Reroute when there is a WAN Outage.
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.

31. 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)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
Higher
On CUCM it's identical to adding an H323 GW.
...

32. What does rule 1 /^.*(....$)/ /1/ do?
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.
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
Matches any length dialed number and truncates it to 4 digits.

33. Aside from the Media Resources --> Mobile Voice Access where else do you need to specify the the Mobile Voice Access DN?
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
Reroute when there is a WAN Outage.
Service Parameters --> Mobile Voice Access Number
...

34. How do you configure SRST?
In CUCM - configure CFUR to point to it's E164 number.
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
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
Call-manager-fallback or telephony-service mode SRST

35. What CUCM services should you activate?
#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.
#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
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!

36. What are the commands to configure SRST in fallback?
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
Configure it on the route group through the route list - then it will be local to the route list.

37. What are the base telephony-service commands for CME?
Dial-peers
(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
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
Apply voice-class codec

38. What are the commands to create the L3 routing interface for VLANS (SVI)?
The Called/Calling Transformations are superceded.
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
(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
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.

39. Name 2 commands to verify RSVP functionality.
Call forwarding between voip to voip (when CUBE is in play)
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
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

40. How do you configure phone ports on an ESW?
Mobile Connect
Via the voice hunt-group parallel command
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

41. What is the Enable Enterprise Feature Access Service Parameter used for?
Assign a logout-profile to the ephone.
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.
To enable two-stage dialing.
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs

42. How do you test a Voice Translation Rule?
Sh ip rsvp reservation||sh sccp connections
#test voice translation rule 1 <input to test>

When they are explicitly matched in a destination-pattern in a dial-peer.

43. How can you confirm the MGCP GW is registered to CUCM - in IOS?
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
Mobile Connect
Sh ccm-manager
Put them in a route list.

44. When can you not use a Standard Local Route Group?
Mobile Voice Access
When configuring TEHO.
SIP Dial Rules
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary

45. Name 4 useful show commands for active calls.
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
(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
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary

46. How do you ensure that G.711 only is used?
1 ... this is not optional!
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
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.

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

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
The Called/Calling Transformations are superceded.
Service Parameter --> Block OffNet To OffNet Transfer --> True

48. If you need terminate and reoriginate all calls through the CUBE - what do you need to do.
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
Device Pool Locations
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True

49. What are the 3 mandatory commands within call-manager-fallback?
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
CUCM Service Parameters --> Call Park Reversion Timer
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
Max-dn|max-ephone|ip source-address

50. How do you use an ephone template?
Reroute when here is WAN congestion.
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
Apply it to the ephone.
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.