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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. What is the full E164 format?
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
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
2. Aside from the <i>Media Resources --> Mobile Voice Access</i> where else do you need to specify the the Mobile Voice Access DN?
Service Parameters --> <i>Mobile Voice Access Number</i>
Max-dn|max-ephone|ip source-address
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
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
3. What does Display-IE do?
#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
Sends the Calling Name.
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.
...
4. What should you configure before entering auto qos?
Bandwidth
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Sh ip rsvp reservation||sh sccp connections
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
5. 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?
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
Precede the # with a > ... so 9011*>#
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
6. How do you maintain 4-digit dialing to phones that are running in an SRST scenario (WAN outage)?
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7. Is the order of the MRGs in an MRGL significant?
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
(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
Allows you to transfer by only pressing the Transfer button once.
8. On the RDP What is the Rerouting CSS used for?
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 ....
Single Number Reach (Mobile Connect)
Via the <i>voice hunt-group parallel</i> command
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
9. 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?
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
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.
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.
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
10. 'How do you inform a SIP phone of NTP information?
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
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
#test voice translation rule 1 <input to test>
#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
11. What is a best practice for digit manipulation - in regards to H323 GWs?
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
(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
Allows you to transfer by only pressing the Transfer button once.
12. How do you create a trunk on the switch side?
<b>Access Port Method</b>||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
13. How can you tell the difference between Proxy Line Mode and Shared Line Mode by just looking at the phone?
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
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
14. What are the basic SCCP Commands fro telephony-service in CUCME?
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
<b>Access Port Method</b>||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
The Called/Calling Transformations are superceded.
15. How do you configure SIP phones to be able to call other SIP phones via SIP URI?
#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>SIP Route Pattern</i> over a SIP Trunk.
At the CLI: <i>utils ntp status</i>
It's best to strip digits at the voice port.
16. What's the best way to do digit manipulation on an IOS gateway?
#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)#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
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
Precede the # with a > ... so 9011*>#
17. Privacy is enabled system-wide in CUCM by default. (T or F)
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
To enable two-stage dialing.
TRUE
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
18. On an MGCP GW - how could you see the primary and backup CUCM servers?
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
Show ccm-manager
19. How do you get around relying on DNS for your CUCMs?
Mobile Connect
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
Matches any length dialed number and truncates it to 4 digits.
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
20. How do you disable KPML?
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 PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Assign SIP Dial Rules
Allows you to transfer by only pressing the Transfer button once.
21. How do you see multicast packets being sent?
Debug ephone moh
1 ... this is not optional!
At the CLI: <i>utils ntp status</i>
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
22. How do you configure CUBE?
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23. How do you allow H323 calls to be preserved should the primary H323 GW fail?
Default is SRST||<i>mode cme</i> changes to CME|<i>no mode cme</i> changes back to SRST
1 ... this is not optional!
24kb/s
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
24. Is the + character supported on a VOIP dial-peer?
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25. If you can't use a tech-prefix or default tech-prefix how do you get around the fact that CUCM does not register it's endpoint to GK?
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)
Debug isdn q931
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
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
26. What does <i>ccm-manager config server [IP]</ip> do?
Automatically configures the MGCP GW for you.
Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</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
(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
27. Need to Know||Anytime you have an external call - it needs to match on a route pattern.
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
When they are explicitly matched in a destination-pattern in a dial-peer.
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 Simulator. You can use to validate path from router to the PSTN.
28. When dealing with an MGCP GW there is no intelligence on the router. How would you strip an incoming Called Number from 10 to 4 digits?
Bottom up - to prevent glare.
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
IP Voice Media Streaming App
29. What are the commands to configure SIP phones in CME?
Show ccm-manager
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
Outbound dial-peers
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
30. When configuring a Gatekeep-controlled trunk - what do you do with the <i>Wait for Far End H.245 TCS</i> option?
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
Dial-peers
31. How do you ensure that G.711 only is used?
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
#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
Debug isdn q931
Call forwarding between voip to voip (when CUBE is in play)
32. In CUCME what SIP config command is equivalent to ephone and ephone-dn?
#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
<i>voice register pool</i>|and|<i>voice register dn</i>
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
33. What would force you to use telephony-service to configure SRST?
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
CUCM OS Administration Settings --> NTP Servers
Service Parameters --> <i>Mobile Voice Access Number</i>
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.
34. How do you verify that NTP is working on the CUCM server?
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 IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
At the CLI: <i>utils ntp status</i>
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
35. What are the commands to create the L3 routing interface for VLANS (SVI)?
It's best to strip digits at the voice port.
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
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
36. How do do configure TEHO?
Bottom up - to prevent glare.
Dtmf-relay h245-alpha
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
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
37. How do you place SCCP and SIP phones into a single huntgroup?
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
Via the <i>voice hunt-group parallel</i> command
Sh ccm-manager
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
38. What does an H323 GW require that MGCP GWs do not?
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
Dial-peers
Mobile Connect
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
39. How do you configure class of service (CoS) in CUCM? CME?
The Called/Calling Transformations are superceded.
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
<b>Access Port Method</b>||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
<i>#after-hours block pattern</i>
40. Where can you find the command to restore any of the default IP Phone Services if they happen to have been deleted?
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
Via the <i>voice hunt-group parallel</i> command
#test voice translation rule 1 <input to test>
41. What kind of relationship does an MGCP gateway have with CUCM?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Master/Slave relationship. CUCM controls it.
<i>voice translation-profile</i>|and|<i>voice translation-rule</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
42. How do you support multiple codecs on a dialpeer?
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.
Apply <i>voice-class codec</i>
Sh ccm-manager
Configure it on the route group through the route list - then it will be local to the route list.
43. Where do you use VIA zone?
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
Apply it 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
Reroute when here is WAN congestion.
44. What CUCM services should you activate?
(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
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
Higher
45. An MGCP gateway serving as an SRST router requires what?
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
46. What are the commands to manually configure an MGCP gateway?
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
8kb/s
#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
47. How do you see the details of calls coming in and out of the PRI?
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
Outbound dial-peers
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
Debug isdn q931
48. 'If you want to make changes to any softkeys where do you do it?
Mobile Connect
#test voice translation rule 1 <input to test>
SERVICE PARAMETER> Automated Alternate Routing Enable > True
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
49. 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.
Mobile Voice Access
When configuring TEHO.
Configure it on the route group through the route list - then it will be local to the route list.
50. How do you verify where MOH is being served up from?
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