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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. When using <i>drop-through-option</i> What is the max number of huntgroups?
#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
1 ... this is not optional!
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
Call forwarding between voip to voip (when CUBE is in play)
2. In Gatekeeper CAC how do you restrict a specific endpoint?
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
#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
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
Matches any length dialed number and truncates it to 4 digits.
3. How do you configure SRST?
...
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
Call-manager-fallback</i> or <i>telephony-service mode SRST</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)
4. An MGCP gateway serving as an SRST router requires what?
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
IP Voice Media Streaming App
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
5. If you're using Called/Calling Party Transformations at the Route Pattern - what happens if you do Digit manipulation at the route list?
SERVICE PARAMETER> Automated Alternate Routing Enable > True
The Called/Calling Transformations are superceded.
Sh ccm-manager
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 configure a gateway to register with gatekeeper?
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
#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
Default is SRST||<i>mode cme</i> changes to CME|<i>no mode cme</i> changes back to SRST
#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
7. What commands are needed to configure the voice register pool in CME?
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
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
IP Voice Media Streaming App
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
8. How do you set the Call Park Reversion Timer?
Put them in a route list.
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
<i>Auto Call Pickup Enabled</i>
9. Where is it best to manipulate digits for inbound calls?
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10. When do you use translate called? Translate calling?
Service Parameters --> <i>Mobile Voice Access Number</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)
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.
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
11. How do you verify where MOH is being served up from?
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12. What are the commands to create vlans on an ESW?
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
At the CLI: <i>utils ntp status</i>
(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
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
13. When a device is roaming to a different site How does it get the <i>Roaming Sensitive Settings</i> from the Device Pool?
...
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
14. 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
Master/Slave relationship. CUCM controls it.
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
15. What dtmf-relay type do you use for an H323 GW?
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
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
Dtmf-relay h245-alpha
Via the <i>voice hunt-group parallel</i> command
16. How do you set up redundancy on outbound dial-peers on an H323 gateway?
At the CLI: <i>utils ntp status</i>
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
When multiple sites use the same route pattern - do your digit manipulation on a route list.
17. How do you allow the Calling Name to be sent to the PSTN on a router?
The Called/Calling Transformations are superceded.
To notify SIP Phones of NTP
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
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 ....
18. When it comes to H323 gateways - what should you always change - unless explicitly told not to?
Master/Slave relationship. CUCM controls it.
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
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.
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
19. What is call-forward pattern used for?
...
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
<i>SIP Route Pattern</i> over a SIP Trunk.
Call forwarding between voip to voip (when CUBE is in play)
20. How much bandwidth does a G.729 call including layer 3 require?
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
Service Parameters --> <i>Mobile Voice Access Number</i>
24kb/s
In CUCM - configure CFUR to point to it's E164 number.
21. Name 2 commands to verify RSVP functionality.
Sh ip rsvp reservation||sh sccp connections
Matches any length dialed number and truncates it to 4 digits.
<i>#after-hours block pattern</i>
Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</i>
22. How would you set the the timer for Auto Answer?
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23. How do you block calls under call-manager-fallback?
Allows you to transfer by only pressing the Transfer button once.
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
<i>#after-hours block pattern</i>
(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
24. How do you configure DHCP in IOS?
1 ... this is not optional!
...
(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 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
25. What are the steps to integrate CUPS with CUCM?
Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</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
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
26. How do you disable KPML?
IP Voice Media Streaming App
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 ....
Assign SIP Dial Rules
(config)#voice register dialplan
27. How do you configure a SIP Trunk? (router side)
When configuring TEHO.
It's best to strip digits at the voice port.
Max-dn|max-ephone|ip source-address
(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
28. On the RDP What is the CSS used for?
Put them in a route list.
Apply it to the ephone.
Max-dn|max-ephone|ip source-address
Mobile Voice Access
29. What should you configure before entering auto qos?
On CUCM it's identical to adding an H323 GW.
Bandwidth
Outbound dial-peers
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
30. When you make a call and it results in a secondary dial-tone - What is this a symptom of?
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
Device Pool Locations
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
<i>#after-hours block pattern</i>
31. How can you tell the difference between Proxy Line Mode and Shared Line Mode by just looking at the phone?
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
Outbound dial-peers
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
32. 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.
Apply it to the ephone.
Phone number followed by domain name.|i.e. 3006@ipxcme.com
#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
33. How do you configure phone ports on an ESW?
Via the <i>voice hunt-group parallel</i> command
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 .)
To enable two-stage dialing.
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
34. What does <i>rule 1 /^.*(....$)/ /1/</i> do?
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
Automatically configures the MGCP GW for you.
SIP Dial Rules
Matches any length dialed number and truncates it to 4 digits.
35. How do you maintain 4-digit dialing to phones that are running in an SRST scenario (WAN outage)?
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36. What would force you to use telephony-service to configure SRST?
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
Dtmf-relay h245-alpha
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
37. What does an H323 GW require that MGCP GWs do not?
Dial-peers
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
Max-dn|max-ephone|ip source-address
Put them in a route list.
38. What is the <i>Enable Enterprise Feature Access</i> Service Parameter used for?
To enable two-stage dialing.
Service Parameters --> <i>Mobile Voice Access Number</i>
Call forwarding between voip to voip (when CUBE is in play)
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
39. How do you enable AAR system wide?
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
#test voice translation rule 1 <input to test>
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
40. How do you enable AAR?
Allows you to transfer by only pressing the Transfer button once.
SERVICE PARAMETER> Automated Alternate Routing Enable > True
At the CLI: <i>utils ntp status</i>
(config)#<i>sh cdp neigh detail
41. How do you verify that NTP is working on the CUCM server?
SERVICE PARAMETER> Automated Alternate Routing Enable > True
At the CLI: <i>utils ntp status</i>
Automatically configures the MGCP GW for you.
Sh ccm-manager
42. What does the <i>Transfer On-hook Enabled</i> Service Parameter do?
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
Bandwidth
Allows you to transfer by only pressing the Transfer button once.
Put them in a route list.
43. On an H323 GW - how do you adjust the timers for redundancy hunting?
(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
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>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
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
44. A <i>show isdn status</i> shows TEI_ASSIGNED. What do you do to get MULTIPLE_FRAME_ESTABLISHED?
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.
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
Via the <i>voice hunt-group parallel</i> command
45. How do you setup AutoRegistration in CUCM?
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
1 ... this is not optional!
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.
Debug isdn q931
46. What are the commands to configure SRST in fallback?
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
Via the <i>voice hunt-group parallel</i> command
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
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
47. 'If you're not getting a DHCP address from CUCM what then?
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
Apply <i>voice-class codec</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
48. Where can you assign the AAR Group?
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
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
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
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.
49. What are the basic ephone-dn and ephone commands?
Default is SRST||<i>mode cme</i> changes to CME|<i>no mode cme</i> changes back to SRST
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
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
...
50. What are the commands to set up the PRI?
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