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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 set up redundancy on outbound dial-peers on an H323 gateway?
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
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
When they are explicitly matched in a destination-pattern in a dial-peer.
2. 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?
At the CLI: <i>utils ntp status</i>
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
Phone number followed by domain name.|i.e. 3006@ipxcme.com
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.
3. '<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 is truly unsurpassed. I have opened up a wealth of opportunities for myself and my family and
...
CUCM OS Administration Settings --> NTP Servers
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
In CUCM - configure CFUR to point to it's E164 number.
4. How do you configure SRST?
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 .)
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
Matches any length dialed number and truncates it to 4 digits.
SIP Dial Rules
5. How do you set the Call Park Reversion Timer?
Max-dn|max-ephone|ip source-address
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
'Service Parameters --> <i>Auto Answer Timer</i>
6. What does Display-IE do?
<i>SIP Route Pattern</i> over a SIP Trunk.
Sends the Calling Name.
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
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)
7. 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?
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
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)
Assign a logout-profile to the ephone.
On CUCM it's identical to adding an H323 GW.
8. How do you change modes in <i>voice register global</i>?
Default is SRST||<i>mode cme</i> changes to CME|<i>no mode cme</i> changes back to SRST
(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
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
9. How would you set the the timer for Auto Answer?
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10. What are the commands to configure a T1/E1 PRI?
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
(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
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
Assign SIP Dial Rules
11. An MGCP gateway serving as an SRST router requires what?
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.
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
#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
12. How do you block an external call from being transferred back out to the pstn by an internal user?
Debug isdn q931
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
13. What are the commands to configure a SIP phone in CUCME?
Sh ccm-manager
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
Apply <i>voice-class codec</i>
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
14. SNR is also known as?
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.
...
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
Mobile Connect
15. 'If you're not getting a DHCP address from CUCM what then?
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</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.
Automatically configures the MGCP GW for you.
16. 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.
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
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 ....
It's best to strip digits at the voice port.
17. What does KPML do?
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
Max-dn|max-ephone|ip source-address
Dtmf-relay h245-alpha
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
18. What's the difference between AAR and SRST?
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Allows you to transfer by only pressing the Transfer button once.
Phone number followed by domain name.|i.e. 3006@ipxcme.com
Single Number Reach (Mobile Connect)
19. Name 2 commands to verify RSVP functionality.
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
Configure it on the route group through the route list - then it will be local to the route list.
Sh ip rsvp reservation||sh sccp connections
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
20. How do you ensure that G.711 only is used?
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
Automatically configures the MGCP GW for you.
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
21. How do you create a trunk on the router side?
(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
<i>#after-hours block pattern</i>
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
...
22. How do you see the details of calls coming in and out of the PRI?
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
Put them in a route list.
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.
23. How do you configure CUCM redundancy on an H323 gateway?
#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
Put them in a route list.
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
24. What types of digit manipulation can you perform at the route pattern?
<i>voice register pool</i>|and|<i>voice register dn</i>
Call forwarding between voip to voip (when CUBE is in play)
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
25. Where can you assign the AAR Group?
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
Device Pool Locations
Mobile Connect
(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
26. What command is needed on a gateway for it to invoke the DSP to convert multicast MOH stream to TDM?
Ccm-manager music-on-hold
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
Max-dn|max-ephone|ip source-address
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
27. How do you configure AAR?
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
Mobile Voice Access
On CUCM it's identical to adding an H323 GW.
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
28. What does an H323 GW require that MGCP GWs do not?
Dial-peers
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
29. On an MGCP GW - how could you see the primary and backup CUCM servers?
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
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
Show ccm-manager
#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
30. How do you verify where MOH is being served up from?
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31. What CUCM services should you activate?
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
Via the <i>voice hunt-group parallel</i> command
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
SERVICE PARAMETER> Automated Alternate Routing Enable > True
32. What is call-forward pattern used for?
IP Voice Media Streaming App
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 forwarding between voip to voip (when CUBE is in play)
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. How do you use the # as a string terminator within a SIP Dial Rule?
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
Precede the # with a > ... so 9011*>#
Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</i>
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
34. Does CUCM support RSVP natively?
Ccm-manager music-on-hold
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
Dtmf-relay h245-alpha
35. Where do you use VIA zone?
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
SERVICE PARAMETER> Automated Alternate Routing Enable > True
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>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
36. When setting up SIP URI where do you configure the CUCM's domain name?'
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
TRUE
Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</i>
37. 'How do you inform a SIP phone of NTP information?
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
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
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.
(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
38. When do you do digit manipulation at the route pattern as opposed to the route list?
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
When multiple sites use the same route pattern - do your digit manipulation on a route list.
39. In Gatekeeper CAC how do you restrict a specific endpoint?
Via the <i>voice hunt-group parallel</i> command
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
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
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
40. How is NTP sych setup in CUCM?
CUCM OS Administration Settings --> NTP Servers
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
41. Two useful troubleshooting commands for CUCME?
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
(config)#<i>sh cdp neigh detail
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
8kb/s
42. How do you configure a SIP Trunk? (router side)
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
Put them in a route list.
(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
43. Calls coming from CUCM to PSTN need what?
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
Single Number Reach (Mobile Connect)
Outbound dial-peers
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
44. What are the base telephony-service commands for CME?
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
Matches any length dialed number and truncates it to 4 digits.
TRUE
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
45. How do you use an ephone template?
Apply it to the ephone.
Master/Slave relationship. CUCM controls it.
To enable two-stage dialing.
When multiple sites use the same route pattern - do your digit manipulation on a route list.
46. What are the commands to create vlans on an ESW?
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
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
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
47. What terminology translates to SRST?
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
Reroute when there is a WAN Outage.
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
Dial-peers
48. What must you do for a BACD script to work on a CME router?
TRUE
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
'Service Parameters --> <i>Auto Answer Timer</i>
When they are explicitly matched in a destination-pattern in a dial-peer.
49. How do you prevent toll fraud on CME?
(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
TRUE
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 CUCM - configure CFUR to point to it's E164 number.
50. 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
Call forwarding between voip to voip (when CUBE is in play)
#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
To notify SIP Phones of NTP