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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. How do you maintain 4-digit dialing to phones that are running in an SRST scenario (WAN outage)?
2. How do you ensure that G.711 only is used?
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
Phone number followed by domain name.|i.e. 3006@ipxcme.com
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
3. How do you prevent toll fraud on CUCM?
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
Allows you to transfer by only pressing the Transfer button once.
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
4. How do you allow the Calling Name to be sent to the PSTN on a router?
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
<i>SIP Route Pattern</i> over a SIP Trunk.
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
5. How do you create a trunk on the switch side?
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
Sh ip rsvp reservation||sh sccp connections
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
6. What is call-forward pattern used for?
(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
Call forwarding between voip to voip (when CUBE is in play)
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
Allows you to transfer by only pressing the Transfer button once.
7. How do you see multicast packets being sent?
CUCM OS Administration Settings --> NTP Servers
Debug ephone moh
Mobile Connect
1 ... this is not optional!
8. When configuring a Gatekeep-controlled trunk - what do you do with the <i>Wait for Far End H.245 TCS</i> option?
(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
(config)#<i>sh cdp neigh detail
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
9. How do you configure phone ports on an ESW?
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
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
Phone number followed by domain name.|i.e. 3006@ipxcme.com
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
10. When it comes to H323 gateways - what should you always change - unless explicitly told not to?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Max-dn|max-ephone|ip source-address
'Use before the ... so XXXX
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
11. <b>By Nov. 16 - 2010</b>||I am happy confident and thankful that I am now a CCIE Voice Certified engineer! My technical knowledge of the Cisco UC portfolio os truly unsurpassed. I have opened up a wealth of opportunities for myself and my family and
...
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
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 ....
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
12. In CME Where is the Calling Name derived from?
(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
Ccm-manager music-on-hold
The <i>name</i> configuration field in ephone-dn and voice register dn
(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
13. 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?
Reroute when here is WAN congestion.
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
Default is SRST||<i>mode cme</i> changes to CME|<i>no mode cme</i> changes back to SRST
Higher
14. How do you use an ephone template?
Apply it to the ephone.
Sh ccm-manager
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
Ccm-manager music-on-hold
15. How do you prevent H323 caller-id updates to CUCM
#voice service voip| #no supplementary-service h225-notify cid-update
Phone number followed by domain name.|i.e. 3006@ipxcme.com
Show ccm-manager
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
16. How do you ensure an H323 GW always uses the Sub as primary and Pub as a backup?
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa 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
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.
17. What are the commands to configure SRST in fallback?
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
24kb/s
Outbound dial-peers
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
18. What are the 3 mandatory commands within call-manager-fallback?
Precede the # with a > ... so 9011*>#
<i>voice register pool</i>|and|<i>voice register dn</i>
Max-dn|max-ephone|ip source-address
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 .)
19. When a device is roaming to a different site How does it get the <i>Roaming Sensitive Settings</i> from the Device Pool?
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
IP Voice Media Streaming App
TRUE
20. How do you get around relying on DNS for your CUCMs?
Via the <i>voice hunt-group parallel</i> command
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
Allows you to transfer by only pressing the Transfer button once.
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.
21. What are the commands to configure a MGCP Gateway? (router)
<i>#after-hours block pattern</i>
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
22. What is the Service Parameter to allow the phone that is <i>picking ip</i> to auto answer?
<i>Auto Call Pickup Enabled</i>
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 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
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
23. What are the steps to configure Single Number Reach and Mobile Voice Access?
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
TRUE
...
24. How do you create a trunk on the router side?
To notify SIP Phones of NTP
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
Mobile Connect
Default is SRST||<i>mode cme</i> changes to CME|<i>no mode cme</i> changes back to SRST
25. How do you configure phone ports on a 3750?
Matches any length dialed number and truncates it to 4 digits.
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)#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
<b>Access Port Method</b>||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
26. Is the order of the MRGs in an MRGL significant?
8kb/s
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
27. 'If you're not getting a DHCP address from CUCM what then?
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.
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
Top Down means that channel 1 will be the first channel used to place outgoing calls.
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
28. An MGCP gateway serving as an SRST router requires what?
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 ....
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
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
When multiple sites use the same route pattern - do your digit manipulation on a route list.
29. What is the full E164 format?
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
CUCM OS Administration Settings --> NTP Servers
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
30. What commands are needed to configure the voice register pool in CME?
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
#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
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
Bottom up - to prevent glare.
31. 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?
Dial-peers
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.
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
32. Where do you use VIA zone?
To notify SIP Phones of NTP
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
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
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
33. What does Display-IE do?
Sends the Calling Name.
Mobile Voice Access
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
When configuring TEHO.
34. How do you prevent toll fraud on CME?
Configure it on the route group through the route list - then it will be local to the route list.
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 ....
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
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
35. SNR is also known as?
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
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
SIP Dial Rules
Mobile Connect
36. When can you not use a Standard Local Route Group?
Sh ccm-manager
When configuring TEHO.
Configure it on the route group through the route list - then it will be local to the route list.
...
37. 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)
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
Device Pool Locations
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
38. How do you configure CUCM redundancy on an H323 gateway?
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
#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
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
Dtmf-relay h245-alpha
39. 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
Apply <i>voice-class codec</i>
Bandwidth
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
40. 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?
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
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 .)
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
41. How do you block an external call from being transferred back out to the pstn by an internal user?
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
In CUCM - configure CFUR to point to it's E164 number.
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
<b>Access Port Method</b>||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
42. What do you need to do to activate the CME GUI?
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
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
Sends the Calling Name.
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
43. 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
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
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
44. 'How do you inform a SIP phone of NTP information?
Sends the Calling Name.
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
45. When using <i>drop-through-option</i> What is the max number of huntgroups?
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)
Precede the # with a > ... so 9011*>#
1 ... this is not optional!
46. What command is needed on a gateway for it to invoke the DSP to convert multicast MOH stream to TDM?
Apply <i>voice-class codec</i>
SIP Dial Rules
Ccm-manager music-on-hold
Dial-peers
47. What are the commands to manually configure an MGCP gateway?
To notify SIP Phones of NTP
Mobile Voice Access
#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
24kb/s
48. How do you enable AAR?
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
Reroute when there is a WAN Outage.
SERVICE PARAMETER> Automated Alternate Routing Enable > True
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
49. What kind of relationship does an MGCP gateway have with CUCM?
#test voice translation rule 1 <input to test>
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
...
Master/Slave relationship. CUCM controls it.
50. How do you configure AAR?
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
'Service Parameters --> <i>Auto Answer Timer</i>
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.