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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 CME Where is the Calling Name derived from?
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
The <i>name</i> configuration field in ephone-dn and voice register dn
Assign a logout-profile to the ephone.
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
2. 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?
Allows you to transfer by only pressing the Transfer button once.
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.
Mobile Connect
Dtmf-relay h245-alpha
3. How do you test a Voice Translation Rule?
Higher
<i>Auto Call Pickup Enabled</i>
#test voice translation rule 1 <input to test>
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
4. On an MGCP GW - if you select <i>Top Down</i> under Channel Selection Order - What does that mean?
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.
Outbound dial-peers
Sh ip rsvp reservation||sh sccp connections
Top Down means that channel 1 will be the first channel used to place outgoing calls.
5. How do you configure SIP phones to be able to call other SIP phones via SIP URI?
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
<i>SIP Route Pattern</i> over a SIP Trunk.
Sh ip rsvp reservation||sh sccp connections
Allows you to transfer by only pressing the Transfer button once.
6. What do you need to do to activate the CME GUI?
Automatically configures the MGCP GW for you.
Debug isdn q931
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
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
7. How do you see multicast packets being sent?
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
Debug ephone moh
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
8. How do you allow H323 calls to be preserved should the primary H323 GW fail?
Reroute when there is a WAN Outage.
Dial-peers
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
9. What's the best way to do digit manipulation on an IOS gateway?
(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
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
Bottom up - to prevent glare.
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.
10. How do you configure CUBE?
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11. Two useful troubleshooting commands for CUCME?
Debug isdn q931
In CUCM - configure CFUR to point to it's E164 number.
...
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
12. How do do configure TEHO?
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
'Service Parameters --> <i>Auto Answer Timer</i>
<i>Auto Call Pickup Enabled</i>
13. Is the order of the MRGs in an MRGL significant?
Device Pool Locations
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
Assign a logout-profile to the ephone.
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
14. What are the basic ephone-dn and ephone commands?
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Dial-peers
Outbound dial-peers
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
15. How do you support multiple codecs on a dialpeer?
Matches any length dialed number and truncates it to 4 digits.
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
Apply <i>voice-class codec</i>
16. 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
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
Configure it on the route group through the route list - then it will be local to the route list.
17. What command is needed on a gateway for it to invoke the DSP to convert multicast MOH stream to TDM?
When configuring TEHO.
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
#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
Ccm-manager music-on-hold
18. How would you verify that DHCP is working in IOS?
Debug ephone moh
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
(config)#<i>sh cdp neigh detail
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
19. What are the commands to ensure an H323 GW always uses the Sub as primary and Pub as a backup?
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
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
The Called/Calling Transformations are superceded.
#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
20. How do you set up redundancy on outbound dial-peers on an H323 gateway?
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
It's best to strip digits at the voice port.
24kb/s
(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
21. What is the bit rate for a G.729 call excluding layer 2?
#voice service voip| #no supplementary-service h225-notify cid-update
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
8kb/s
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
22. Where can you assign the AAR Group?
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
(config)#voice register dialplan
23. How is NTP sych setup in CUCM?
(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
Bottom up - to prevent glare.
CUCM OS Administration Settings --> NTP Servers
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
24. If the PSTN is presenting on channel 1 - should you configure CUCM to search 'top down' or 'bottom up'?
Bottom up - to prevent glare.
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
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.
25. What is the Service Parameter to allow the phone that is <i>picking ip</i> to auto answer?
#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
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
Top Down means that channel 1 will be the first channel used to place outgoing calls.
<i>Auto Call Pickup Enabled</i>
26. What are the commands to configure a MGCP Gateway? (router)
TRUE
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
Sends the Calling Name.
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
27. What are the commands to configure NTP in IOS?
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
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</i>
28. What are the commands to configure a T1/E1 PRI?
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
(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
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
Configure it on the route group through the route list - then it will be local to the route list.
29. What would force you to use telephony-service to configure SRST?
Master/Slave relationship. CUCM controls it.
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
30. How do you create a trunk on the router side?
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
Bandwidth
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
Precede the # with a > ... so 9011*>#
31. When setting up SIP URI where do you configure the CUCM's domain name?'
At the CLI: <i>utils ntp status</i>
(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
Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</i>
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
32. What is the full E164 format?
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
Sh ip rsvp reservation||sh sccp connections
When they are explicitly matched in a destination-pattern in a dial-peer.
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
33. If you need terminate and reoriginate all calls through the CUBE - what do you need to do.
IP Voice Media Streaming App
Sh ip rsvp reservation||sh sccp connections
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
34. How would you set the the timer for Auto Answer?
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35. What are the commands to create the L3 routing interface for VLANS (SVI)?
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
<i>#after-hours block pattern</i>
Outbound dial-peers
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
36. What does an H323 GW require that MGCP GWs do not?
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
Ccm-manager music-on-hold
Dial-peers
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
37. What should you configure before entering auto qos?
'Use before the ... so XXXX
#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
Bandwidth
<i>voice register pool</i>|and|<i>voice register dn</i>
38. What dtmf-relay type do you use for an H323 GW?
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.
#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
<i>#after-hours block pattern</i>
Dtmf-relay h245-alpha
39. How do you use the # as a string terminator within a SIP Dial Rule?
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
Bandwidth
Precede the # with a > ... so 9011*>#
Mobile Connect
40. On the RDP What is the CSS used for?
Mobile Voice Access
(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
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
To notify SIP Phones of NTP
41. 'If you want to make changes to any softkeys where do you do it?
Outbound dial-peers
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
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
<i>SIP Route Pattern</i> over a SIP Trunk.
42. Maximum Wait Time for Desk Pickup?
Call forwarding between voip to voip (when CUBE is in play)
Debug ephone moh
#test voice translation rule 1 <input to test>
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
43. 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
Bandwidth
(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
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
44. What's the best way to perform digit manipulation on a route group?
Debug ephone moh
Configure it on the route group through the route list - then it will be local to the route list.
Bandwidth
#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
45. On a gatekeeper trunk - how do you ensure the Sub is used as primary and Pub as backup?
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
#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
46. How do you configure class of service (CoS) in CUCM? CME?
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
47. What are the commands to configure an H323 GW?
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
#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
48. How do you create a trunk on the switch side?
Ccm-manager music-on-hold
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
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
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
49. Where do you use VIA zone?
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
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
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
To notify SIP Phones of NTP
50. What are the steps to configure Single Number Reach and Mobile Voice Access?
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.
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
Mobile Connect