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Test your basic knowledge |
CCIE Voice Test
Start Test
Study First
Subjects
:
cisco
,
it-skills
,
ccie
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. 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
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
Precede the # with a > ... so 9011*>#
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 ....
2. How would you enable security on a GK?
24kb/s
#test voice translation rule 1 <input to test>
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.
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
3. How do you configure DHCP in IOS?
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
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
Show ccm-manager
(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
4. What does <i>rule 1 /^.*(....$)/ /1/</i> do?
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
IP Voice Media Streaming App
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
Matches any length dialed number and truncates it to 4 digits.
5. How do you support multiple codecs on a dialpeer?
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
'Service Parameters --> <i>Auto Answer Timer</i>
Apply <i>voice-class codec</i>
Sh ip rsvp reservation||sh sccp connections
6. How do you prioritize route groups?
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
Call Simulator. You can use to validate path from router to the PSTN.
Put them in a route list.
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
7. How do you configure phone ports on a 3750?
<b>Access Port Method</b>||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
Call forwarding between voip to voip (when CUBE is in play)
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
8. Calls coming from CUCM to PSTN need what?
1 ... this is not optional!
Outbound dial-peers
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
Ccm-manager music-on-hold
9. What terminology translates to SRST?
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.
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
Reroute when there is a WAN Outage.
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
10. How do you configure phone ports on an ESW?
(config)#<i>sh cdp neigh detail
<i>Auto Call Pickup Enabled</i>
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
11. How do you prevent toll fraud on CME?
8kb/s
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 ....
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
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
12. How do you verify where MOH is being served up from?
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13. What are the basic ephone-dn and ephone commands?
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
Top Down means that channel 1 will be the first channel used to place outgoing calls.
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
On CUCM -- UPTSAL||Create Your <b>Users</b>|<b>Phone</b> - set owner id|Create SIP <b>Trunk</b>|<b>SIP Trunk Security Profile|</b>|<b>Application Server</b>|<b>Licensing|</b>|||On CUPS||Work Left to right
14. Where can you assign the AAR Group?
When they are explicitly matched in a destination-pattern in a dial-peer.
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
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
15. How do you convert a full PRI 9config'd by ccm-manager config server) back to a Fractional PRI?
#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
Show ccm-manager
1 ... this is not optional!
16. How do you use an ephone template?
It's best to strip digits at the voice port.
Top Down means that channel 1 will be the first channel used to place outgoing calls.
Apply it to the ephone.
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
17. When a device is roaming to a different site How does it get the <i>Roaming Sensitive Settings</i> from the Device Pool?
On CUCM it's identical to adding an H323 GW.
Apply it to the ephone.
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
18. How do you prevent H323 caller-id updates to CUCM
Max-dn|max-ephone|ip source-address
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
#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
#voice service voip| #no supplementary-service h225-notify cid-update
19. How do you enable AAR system wide?
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 .)
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
Default is SRST||<i>mode cme</i> changes to CME|<i>no mode cme</i> changes back to SRST
20. When using <i>drop-through-option</i> What is the max number of huntgroups?
Use the <i>alias static</i> configuration||alias static (IP of CUCM) gkid (GK ID) gateway voip ras (IP of CUCM) e164 (dn) e164 (dn) e164 (dn)
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
Create 2 voip dial-peers - one the Sub and one to the Pub (the Pub with a preference of 1). Then create a <i>voice class h323</i>and set the <i>h225 timeout</i> value so it will hunt between the two.
1 ... this is not optional!
21. How do you block an external call from being transferred back out to the pstn by an internal user?
Ccm-manager music-on-hold
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
Call forwarding between voip to voip (when CUBE is in play)
22. How can you tell the difference between Proxy Line Mode and Shared Line Mode by just looking at the phone?
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
Single Number Reach (Mobile Connect)
Master/Slave relationship. CUCM controls it.
23. How do you prevent toll fraud on CUCM?
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
TRUE
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
24. What is Phone NTP Reference used for?
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
In CUCM - configure CFUR to point to it's E164 number.
To notify SIP Phones of NTP
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
25. What CM Service needs to be start in Serviceability for MOH to work?
...
Create 2 voip dial-peers - one the Sub and one to the Pub (the Pub with a preference of 1). Then create a <i>voice class h323</i>and set the <i>h225 timeout</i> value so it will hunt between the two.
Default is SRST||<i>mode cme</i> changes to CME|<i>no mode cme</i> changes back to SRST
IP Voice Media Streaming App
26. What does an H323 GW require that MGCP GWs do not?
Dial-peers
(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
<i>SIP Route Pattern</i> over a SIP Trunk.
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
27. How is NTP sych setup in CUCM?
CUCM OS Administration Settings --> NTP Servers
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
<i>#after-hours block pattern</i>
Configure it on the route group through the route list - then it will be local to the route list.
28. How do you configure an MGCP GW? (router side)
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
In CUCM - configure CFUR to point to it's E164 number.
(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
29. What is call-forward pattern used for?
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
Sh ip rsvp reservation||sh sccp connections
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 forwarding between voip to voip (when CUBE is in play)
30. What is Mobile Voice Access?
Allows you to transfer by only pressing the Transfer button once.
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
Use the <i>alias static</i> configuration||alias static (IP of CUCM) gkid (GK ID) gateway voip ras (IP of CUCM) e164 (dn) e164 (dn) e164 (dn)
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
31. In CME Where is the Calling Name derived from?
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
In CUCM - configure CFUR to point to it's E164 number.
Dial-peers
The <i>name</i> configuration field in ephone-dn and voice register dn
32. If you need terminate and reoriginate all calls through the CUBE - what do you need to do.
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
Use the <i>alias static</i> configuration||alias static (IP of CUCM) gkid (GK ID) gateway voip ras (IP of CUCM) e164 (dn) e164 (dn) e164 (dn)
<i>Auto Call Pickup Enabled</i>
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
33. What are the 3 mandatory commands within call-manager-fallback?
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
Max-dn|max-ephone|ip source-address
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
(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
34. Maximum Wait Time for Desk Pickup?
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
#test voice translation rule 1 <input to test>
To enable two-stage dialing.
(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
35. On the RDP What is the CSS used for?
Mobile Voice Access
(config)#<i>sh cdp neigh detail
Reroute when there is a WAN Outage.
'Service Parameters --> <i>Auto Answer Timer</i>
36. Name 2 commands to verify RSVP functionality.
Sh ip rsvp reservation||sh sccp connections
Sends the Calling Name.
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
Assign SIP Dial Rules
37. On a gatekeeper trunk - how do you ensure the Sub is used as primary and Pub as backup?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
SIP Dial Rules
38. 'If you're not getting a DHCP address from CUCM what then?
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
Via the <i>voice hunt-group parallel</i> command
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
(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
39. When are digits stripped in a gateway?
When they are explicitly matched in a destination-pattern in a dial-peer.
At the CLI: <i>utils ntp status</i>
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
1 ... this is not optional!
40. How do you change modes in <i>voice register global</i>?
Assign a logout-profile to the ephone.
Default is SRST||<i>mode cme</i> changes to CME|<i>no mode cme</i> changes back to SRST
To enable two-stage dialing.
Reroute when there is a WAN Outage.
41. How do you configure SIP phones to be able to call other SIP phones via SIP URI?
<i>SIP Route Pattern</i> over a SIP Trunk.
Sh ip rsvp reservation||sh sccp connections
Max-dn|max-ephone|ip source-address
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
42. How much bandwidth does a G.729 call including layer 3 require?
24kb/s
<i>Auto Call Pickup Enabled</i>
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
43. If the PSTN is presenting on channel 1 - should you configure CUCM to search 'top down' or 'bottom up'?
Bottom up - to prevent glare.
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 .)
Via the <i>voice hunt-group parallel</i> command
At the CLI: <i>utils ntp status</i>
44. What would force you to use telephony-service to configure SRST?
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
When they are explicitly matched in a destination-pattern in a dial-peer.
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
45. What kind of relationship does an MGCP gateway have with CUCM?
Dial-peers
Master/Slave relationship. CUCM controls it.
#test voice translation rule 1 <input to test>
(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
46. How do you setup AutoRegistration in CUCM?
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
When configuring TEHO.
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
47. What CUCM services should you activate?
8kb/s
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
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.
48. 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
#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
The Called/Calling Transformations are superceded.
(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
49. What should you configure before entering auto qos?
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
Bandwidth
(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)
50. Is the order of the MRGs in an MRGL significant?
Assign SIP Dial Rules
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
Bottom up - to prevent glare.
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).