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CCIE Voice Test
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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 a SIP Trunk? (router side)
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
(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
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
Assign SIP Dial Rules
2. How do you prevent H323 caller-id updates to CUCM
Apply <i>voice-class codec</i>
#voice service voip| #no supplementary-service h225-notify cid-update
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
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
3. How do you convert a full PRI 9config'd by ccm-manager config server) back to a Fractional PRI?
<b>Access Port Method</b>||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.
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
4. What CM Service needs to be start in Serviceability for MOH to work?
IP Voice Media Streaming App
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
5. What does <i>rule 1 /^.*(....$)/ /1/</i> do?
SIP Dial Rules
Matches any length dialed number and truncates it to 4 digits.
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
Reroute when here is WAN congestion.
6. Is the order of the MRGs in an MRGL significant?
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
Via the <i>voice hunt-group parallel</i> command
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
7. What is the <i>Enable Enterprise Feature Access</i> Service Parameter used for?
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
To enable two-stage dialing.
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
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
8. Name 4 useful show commands for active calls.
Device Pool Locations
Bandwidth
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
9. How do you configure SIP phones to be able to call other SIP phones via SIP URI?
...
Mobile Connect
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
<i>SIP Route Pattern</i> over a SIP Trunk.
10. What are the basic SCCP Commands fro telephony-service in CUCME?
Higher
...
Single Number Reach (Mobile Connect)
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
11. How do you configure DHCP in IOS?
Precede the # with a > ... so 9011*>#
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
(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
IP Voice Media Streaming App
12. How do you use an ephone template?
Apply it to the ephone.
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
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
Sh ccm-manager
13. How do you configure 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.
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
Debug isdn q931
14. What does an H323 GW require that MGCP GWs do not?
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
Dial-peers
Max-dn|max-ephone|ip source-address
Allows you to transfer by only pressing the Transfer button once.
15. How would you enable security on a GK?
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
16. How do you block an external call from being transferred back out to the pstn by an internal user?
Assign SIP Dial Rules
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
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 .)
17. What is CSIM?
Call Simulator. You can use to validate path from router to the PSTN.
Apply it to the ephone.
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
Call forwarding between voip to voip (when CUBE is in play)
18. What would force you to use telephony-service to configure SRST?
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
Dial-peers
19. What's the difference between AAR and SRST?
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Debug ephone moh
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
#voice service voip| #no supplementary-service h225-notify cid-update
20. What does KPML do?
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
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
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
21. What are the steps to configure Single Number Reach and Mobile Voice Access?
At the CLI: <i>utils ntp status</i>
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
8kb/s
22. 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
Top Down means that channel 1 will be the first channel used to place outgoing calls.
(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
Master/Slave relationship. CUCM controls it.
23. Does CUCM support RSVP natively?
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 ....
Bottom up - to prevent glare.
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
24. 'If you're not getting a DHCP address from CUCM what then?
SERVICE PARAMETER> Automated Alternate Routing Enable > True
TRUE
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
25. Where is it best to manipulate digits for inbound calls?
Warning
: Invalid argument supplied for foreach() in
/var/www/html/basicversity.com/show_quiz.php
on line
183
26. How do you ensure an H323 GW always uses the Sub as primary and Pub as a backup?
(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
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.
The <i>name</i> configuration field in ephone-dn and voice register dn
27. What are the commands to manually configure an MGCP gateway?
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
#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
28. What are the base telephony-service commands for CME?
#test voice translation rule 1 <input to test>
Call forwarding between voip to voip (when CUBE is in play)
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
29. How would you verify that DHCP is working in IOS?
Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</i>
(config)#<i>sh cdp neigh detail
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
(config)#voice register dialplan
30. On the RDP What is the CSS used for?
Mobile Voice Access
Bandwidth
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
When they are explicitly matched in a destination-pattern in a dial-peer.
31. How do you configure CUBE?
Warning
: Invalid argument supplied for foreach() in
/var/www/html/basicversity.com/show_quiz.php
on line
183
32. If you're using Called/Calling Party Transformations at the Route Pattern - what happens if you do Digit manipulation at the route list?
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
The Called/Calling Transformations are superceded.
Call Simulator. You can use to validate path from router to the PSTN.
33. How do you configure AAR?
(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.
Single Number Reach (Mobile Connect)
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
34. When do you do digit manipulation at the route pattern as opposed to the route list?
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
When multiple sites use the same route pattern - do your digit manipulation on a route list.
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
35. How is NTP sych setup in CUCM?
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
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
CUCM OS Administration Settings --> NTP Servers
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.
36. What are two caveats to using the <i>ccm-manager config server</ip> command?
<i>#after-hours block pattern</i>
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
#test voice translation rule 1 <input to test>
37. What are the steps to integrate CUPS with CUCM?
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
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
On CUCM it's identical to adding an H323 GW.
38. How do you block calls under call-manager-fallback?
<i>#after-hours block pattern</i>
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
39. What is SIP URI?
The <i>name</i> configuration field in ephone-dn and voice register dn
Phone number followed by domain name.|i.e. 3006@ipxcme.com
CUCM OS Administration Settings --> NTP Servers
40. How do you prevent toll fraud on CUCM?
IP Voice Media Streaming App
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
41. When can you not use a Standard Local Route Group?
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
...
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
When configuring TEHO.
42. When do you use translate called? Translate calling?
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
#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
(config)#voice register dialplan
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
43. What are the commands to create vlans on an ESW?
(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
CUCM OS Administration Settings --> NTP Servers
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
44. What do you need to do to activate the CME GUI?
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 .)
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
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
IP Voice Media Streaming App
45. What is Mobile Voice Access?
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
When multiple sites use the same route pattern - do your digit manipulation on a route list.
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
46. How do you place SCCP and SIP phones into a single huntgroup?
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
Via the <i>voice hunt-group parallel</i> command
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
(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
47. Name 2 commands to verify RSVP functionality.
Sh ip rsvp reservation||sh sccp connections
When configuring TEHO.
Precede the # with a > ... so 9011*>#
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
48. What terminology translates to AAR?
Mobile Connect
...
1 ... this is not optional!
Reroute when here is WAN congestion.
49. How do you enable AAR?
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
SERVICE PARAMETER> Automated Alternate Routing Enable > True
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
50. How do you create a trunk on the switch side?
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
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
Sorry!:) No result found.
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