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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 would force you to use telephony-service to configure SRST?
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
Precede the # with a > ... so 9011*>#
(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
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
2. How do you see the details of calls coming in and out of the PRI?
Debug isdn q931
Bandwidth
Put them in a route list.
Device Pool Locations
3. Name 2 commands to verify RSVP functionality.
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
Sh ip rsvp reservation||sh sccp connections
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
4. What kind of relationship does an MGCP gateway have with CUCM?
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
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
Configure it on the route group through the route list - then it will be local to the route list.
Master/Slave relationship. CUCM controls it.
5. How do you configure CUCM redundancy on an H323 gateway?
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
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
#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
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
6. How do you ensure no inter-digit timeout fro SIP phones within CUCM?
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
Debug isdn q931
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
SIP Dial Rules
7. 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
<i>#after-hours block pattern</i>
8kb/s
1 ... this is not optional!
8. When can you not use a Standard Local Route Group?
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
Apply <i>voice-class codec</i>
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
When configuring TEHO.
9. What are the basic SCCP Commands fro telephony-service in CUCME?
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
Device Pool Locations
24kb/s
(config)#<i>sh cdp neigh detail
10. What is the Service Parameter to allow the phone that is <i>picking ip</i> to auto answer?
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
<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
Ccm-manager music-on-hold
11. How do you enable AAR?
Mobile Connect
SERVICE PARAMETER> Automated Alternate Routing Enable > True
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
TRUE
12. How do you configure SRST?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
Call forwarding between voip to voip (when CUBE is in play)
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
13. How do you verify that NTP is working on the CUCM server?
Dtmf-relay h245-alpha
At the CLI: <i>utils ntp status</i>
#test voice translation rule 1 <input to test>
Matches any length dialed number and truncates it to 4 digits.
14. What are the commands to configure SIP phones in CME?
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
#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
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
Master/Slave relationship. CUCM controls it.
15. What's the best way to perform digit manipulation on a route group?
Configure it on the route group through the route list - then it will be local to the route list.
When they are explicitly matched in a destination-pattern in a dial-peer.
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
(config)#voice register dialplan
16. How can you tell the difference between Proxy Line Mode and Shared Line Mode by just looking at the phone?
Show ccm-manager
Master/Slave relationship. CUCM controls it.
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
17. 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
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
18. What terminology translates to SRST?
Reroute when there is a WAN Outage.
#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
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
1 ... this is not optional!
19. How do you change modes in <i>voice register global</i>?
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
In CUCM - configure CFUR to point to it's E164 number.
Default is SRST||<i>mode cme</i> changes to CME|<i>no mode cme</i> changes back to SRST
...
20. Two useful troubleshooting commands for CUCME?
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
#voice service voip| #no supplementary-service h225-notify cid-update
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
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
21. How do you prevent toll fraud on CME?
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
Show ccm-manager
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 ....
TRUE
22. When configuring a Gatekeep-controlled trunk - what do you do with the <i>Wait for Far End H.245 TCS</i> option?
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
8kb/s
Matches any length dialed number and truncates it to 4 digits.
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
23. How do you prevent H323 caller-id updates to CUCM
#voice service voip| #no supplementary-service h225-notify cid-update
#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
Device Pool Locations
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
24. What is call-forward pattern used for?
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
#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
Call forwarding between voip to voip (when CUBE is in play)
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.
25. How do you set the inter-digit timeout for SIP phones in CME?
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
Mobile Connect
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
(config)#voice register dialplan
26. 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)
<i>voice register pool</i>|and|<i>voice register dn</i>
Ccm-manager music-on-hold
Automatically configures the MGCP GW for you.
27. How do you configure phone ports on an ESW?
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
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
Mobile Voice Access
Debug ephone moh
28. In Gatekeeper CAC how do you restrict a specific endpoint?
Precede the # with a > ... so 9011*>#
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
Bottom up - to prevent glare.
29. On an H323 GW - how do you adjust the timers for redundancy hunting?
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
30. What are the base telephony-service commands for CME?
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
Service Parameters --> <i>Mobile Voice Access Number</i>
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
31. If the PSTN is presenting on channel 1 - should you configure CUCM to search 'top down' or 'bottom up'?
Bandwidth
Single Number Reach (Mobile Connect)
Bottom up - to prevent glare.
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
32. How do do configure TEHO?
Sh ip rsvp reservation||sh sccp connections
Phone number followed by domain name.|i.e. 3006@ipxcme.com
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
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
33. What must you do for a BACD script to work on a CME router?
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
Phone number followed by domain name.|i.e. 3006@ipxcme.com
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
34. What should you configure before entering auto qos?
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
Bandwidth
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.
In CUCM - configure CFUR to point to it's E164 number.
35. On an MGCP GW - how could you see the primary and backup CUCM servers?
Put them in a route list.
Show ccm-manager
SERVICE PARAMETER> Automated Alternate Routing Enable > True
Higher
36. Describe the relationship between route patterns and end devices.
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.
#voice service voip| #no supplementary-service h225-notify cid-update
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)
37. What are two caveats to using the <i>ccm-manager config server</ip> command?
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)
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
Show ccm-manager
38. How can you confirm the MGCP GW is registered to CUCM - in IOS?
24kb/s
Sh ccm-manager
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
39. A <i>show isdn status</i> shows TEI_ASSIGNED. What do you do to get MULTIPLE_FRAME_ESTABLISHED?
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
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.
SIP Dial Rules
40. When setting up SIP URI where do you configure the CUCM's domain name?'
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Call forwarding between voip to voip (when CUBE is in play)
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.
Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</i>
41. How do you support multiple codecs on a dialpeer?
Dial-peers
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
(config)#voice register dialplan
Apply <i>voice-class codec</i>
42. When a CUCM device dials a number - what happens?
Sh ip rsvp reservation||sh sccp connections
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
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 .)
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
43. How do you ensure an H323 GW always uses the Sub as primary and Pub as a backup?
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.
Automatically configures the MGCP GW for you.
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
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 .)
44. In CUCME what SIP config command is equivalent to ephone and ephone-dn?
#voice service voip| #no supplementary-service h225-notify cid-update
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
<i>voice register pool</i>|and|<i>voice register dn</i>
45. Maximum Wait Time for Desk Pickup?
Sends the Calling Name.
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 ....
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
Master/Slave relationship. CUCM controls it.
46. What is the bit rate for a G.729 call excluding layer 2?
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
8kb/s
<i>SIP Route Pattern</i> over a SIP Trunk.
Allows you to transfer by only pressing the Transfer button once.
47. How do you configure CUBE?
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48. In a SIP Dial Rule how do you represent a dialed * instead of it being used as a wildcard?
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49. How do you configure AAR?
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
Apply it to the ephone.
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
50. What are the steps to integrate CUPS with CUCM?
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
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)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
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