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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. How do you configure SRST?
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.
Higher
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-manager-fallback</i> or <i>telephony-service mode SRST</i>
2. How would you set the the timer for Auto Answer?
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3. On an H323 GW - how do you adjust the timers for redundancy hunting?
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
<i>Auto Call Pickup Enabled</i>
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
4. 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
Put them in a route list.
<i>SIP Route Pattern</i> over a SIP Trunk.
'Use before the ... so XXXX
5. <b>By Nov. 16 - 2010</b>||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
...
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
'Use before the ... so XXXX
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)
6. What does <i>rule 1 /^.*(....$)/ /1/</i> do?
Via the <i>voice hunt-group parallel</i> command
Matches any length dialed number and truncates it to 4 digits.
Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</i>
<i>voice register pool</i>|and|<i>voice register dn</i>
7. Need to Know||Anytime you have an external call - it needs to match on a route pattern.
Assign SIP Dial Rules
(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
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
8. How do you ensure no inter-digit timeout fro SIP phones within CUCM?
Reroute when there is a WAN Outage.
(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
Allows you to transfer by only pressing the Transfer button once.
SIP Dial Rules
9. How do you set the inter-digit timeout for SIP phones in CME?
(config)#voice register dialplan
(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
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)
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
10. How do you configure phone ports on a 3750?
(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
<b>Access Port Method</b>||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
IP Voice Media Streaming App
<i>Auto Call Pickup Enabled</i>
11. 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.
Show ccm-manager
1 ... this is not optional!
The <i>name</i> configuration field in ephone-dn and voice register dn
12. How do you allow H323 calls to be preserved should the primary H323 GW fail?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Via the <i>voice hunt-group parallel</i> command
#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 --> <i>Auto Answer Timer</i>
13. What is the <i>Enable Enterprise Feature Access</i> Service Parameter used for?
Phone number followed by domain name.|i.e. 3006@ipxcme.com
To enable two-stage dialing.
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
...
14. Where can you assign the AAR Group?
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
...
'Service Parameters --> <i>Auto Answer Timer</i>
15. What CM Service needs to be start in Serviceability for MOH to work?
#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
#test voice translation rule 1 <input to test>
Higher
IP Voice Media Streaming App
16. SNR is also known as?
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
Call Simulator. You can use to validate path from router to the PSTN.
Mobile Connect
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
17. What is the bit rate for a G.729 call excluding layer 2?
8kb/s
Default is SRST||<i>mode cme</i> changes to CME|<i>no mode cme</i> changes back to SRST
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
1 ... this is not optional!
18. On an MGCP GW - if you select <i>Top Down</i> under Channel Selection Order - What does that mean?
Bottom up - to prevent glare.
Ccm-manager music-on-hold
Top Down means that channel 1 will be the first channel used to place outgoing calls.
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
19. How do you place SCCP and SIP phones into a single huntgroup?
Via the <i>voice hunt-group parallel</i> command
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
SIP Dial Rules
Master/Slave relationship. CUCM controls it.
20. How do you see multicast packets being sent?
8kb/s
Debug ephone moh
<i>voice register pool</i>|and|<i>voice register dn</i>
Ccm-manager music-on-hold
21. How do you enable AAR?
SERVICE PARAMETER> Automated Alternate Routing Enable > True
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
22. What should you configure before entering auto qos?
'Service Parameters --> <i>Auto Answer Timer</i>
Bandwidth
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
When configuring TEHO.
23. How do you support multiple codecs on a dialpeer?
Automatically configures the MGCP GW for you.
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
When configuring TEHO.
Apply <i>voice-class codec</i>
24. How do you set up redundancy on outbound dial-peers on an H323 gateway?
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 (if no response within 3 seconds continue dial-peer hunting)
Automatically configures the MGCP GW for you.
Phone number followed by domain name.|i.e. 3006@ipxcme.com
25. How do you prevent toll fraud on CME?
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
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
1 ... this is not optional!
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 ....
26. How do you prevent toll fraud on CUCM?
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
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
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
27. How do you use the # as a string terminator within a SIP Dial Rule?
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
Precede the # with a > ... so 9011*>#
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.
Configure it on the route group through the route list - then it will be local to the route list.
28. What are the 3 mandatory commands within call-manager-fallback?
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
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
Max-dn|max-ephone|ip source-address
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
29. 'How do you inform a SIP phone of NTP information?
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.
(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 - Partitions and Calling Search Spaces||CME - COR Lists
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
30. How do you block calls under call-manager-fallback?
#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.
<i>#after-hours block pattern</i>
The <i>name</i> configuration field in ephone-dn and voice register dn
31. On a gatekeeper trunk - how do you ensure the Sub is used as primary and Pub as backup?
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
(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
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
32. Is the + character supported on a VOIP dial-peer?
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33. In CUCME what SIP config command is equivalent to ephone and ephone-dn?
<i>voice register pool</i>|and|<i>voice register dn</i>
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
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
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
34. What does KPML do?
Automatically configures the MGCP GW for you.
1 ... this is not optional!
The <i>name</i> configuration field in ephone-dn and voice register dn
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
35. 'If you want to make changes to any softkeys where do you do it?
(config)#voice register dialplan
Sh ip rsvp reservation||sh sccp connections
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
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)
36. How would you enable security on a GK?
TRUE
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
37. A <i>show isdn status</i> shows TEI_ASSIGNED. What do you do to get MULTIPLE_FRAME_ESTABLISHED?
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
38. In a SIP Dial Rule how do you represent a dialed * instead of it being used as a wildcard?
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39. What would force you to use telephony-service to configure SRST?
Debug ephone moh
<i>voice register pool</i>|and|<i>voice register dn</i>
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
40. What is CSIM?
#test voice translation rule 1 <input to test>
Call Simulator. You can use to validate path from router to the PSTN.
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
#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
41. When setting up SIP URI where do you configure the CUCM's domain name?'
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
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
Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</i>
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
42. How do you allow the Calling Name to be sent to the PSTN on a router?
Find the GUI Files - show flash | i .html|<i>ip http server</i>|<i>ip http path flash:/<PATH TO FILES></i>||telephony service|web admin system name admin pass cisco|dn-webedit
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
Call forwarding between voip to voip (when CUBE is in play)
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
43. 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 (if no response within 3 seconds continue dial-peer hunting)
#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
SIP Dial Rules
44. Describe how you configure SIP URI functionality.
Dial-peers
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
Find the GUI Files - show flash | i .html|<i>ip http server</i>|<i>ip http path flash:/<PATH TO FILES></i>||telephony service|web admin system name admin pass cisco|dn-webedit
45. When a device is roaming to a different site How does it get the <i>Roaming Sensitive Settings</i> from the Device Pool?
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
SIP Dial Rules
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
46. How do you change modes in <i>voice register global</i>?
Default is SRST||<i>mode cme</i> changes to CME|<i>no mode cme</i> changes back 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.
TRUE
Sh ip rsvp reservation||sh sccp connections
47. What are the commands to configure SRST in fallback?
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
(config)#<i>sh cdp neigh detail
Mobile Connect
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
48. How do you test a Voice Translation Rule?
Bottom up - to prevent glare.
Single Number Reach (Mobile Connect)
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
#test voice translation rule 1 <input to test>
49. An MGCP gateway serving as an SRST router requires what?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
When multiple sites use the same route pattern - do your digit manipulation on a route list.
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
On CUCM it's identical to adding an H323 GW.
50. What must you do for a BACD script to work on a CME router?
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
The <i>name</i> configuration field in ephone-dn and voice register dn
1 ... this is not optional!
...