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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 call-forward pattern used for?
(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
Call forwarding between voip to voip (when CUBE is in play)
Reroute when there is a WAN Outage.
Service Parameters --> <i>Mobile Voice Access Number</i>
2. What are the steps to configure Single Number Reach and Mobile Voice Access?
Bandwidth
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
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
3. With the gw-priority command - does higher or lower priority take precedence?
Higher
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
To enable two-stage dialing.
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
4. A <i>show isdn status</i> shows TEI_ASSIGNED. What do you do to get MULTIPLE_FRAME_ESTABLISHED?
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.
Outbound dial-peers
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
5. What are the basic ephone-dn and ephone commands?
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
Higher
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 ....
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
6. What are the commands to configure a T1/E1 PRI?
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
(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
7. How do you verify that NTP is working on the CUCM server?
<i>SIP Route Pattern</i> over a SIP Trunk.
At the CLI: <i>utils ntp status</i>
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
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 ....
8. On an H323 GW - how do you adjust the timers for redundancy hunting?
To notify SIP Phones of NTP
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
9. <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
Default is SRST||<i>mode cme</i> changes to CME|<i>no mode cme</i> changes back to 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
...
Single Number Reach (Mobile Connect)
10. How do you get around relying on DNS for your CUCMs?
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
The <i>name</i> configuration field in ephone-dn and voice register dn
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.
Reroute when there is a WAN Outage.
11. What are the commands to configure a SIP phone in CUCME?
Configure it on the route group through the route list - then it will be local to the route list.
SERVICE PARAMETER> Automated Alternate Routing Enable > True
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
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
12. What are the commands to configure NTP in IOS?
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
Apply it to the ephone.
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
13. How can you confirm the MGCP GW is registered to CUCM - in IOS?
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
8kb/s
Sh ccm-manager
Top Down means that channel 1 will be the first channel used to place outgoing calls.
14. How much bandwidth does a G.729 call including layer 3 require?
24kb/s
(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
Ccm-manager music-on-hold
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
15. How do you create a trunk on the router side?
IP Voice Media Streaming App
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
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
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
16. How do you prevent toll fraud on CME?
TRUE
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
Matches any length dialed number and truncates it to 4 digits.
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 ....
17. What are the commands to configure a MGCP Gateway? (router)
Via the <i>voice hunt-group parallel</i> command
<i>#after-hours block pattern</i>
Call Simulator. You can use to validate path from router to the PSTN.
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
18. How do you ensure that G.711 only is used?
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
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
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
19. How do you place SCCP and SIP phones into a single huntgroup?
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
Via the <i>voice hunt-group parallel</i> command
#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
20. SNR is also known as?
24kb/s
Mobile Connect
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
21. What are the 3 mandatory commands within call-manager-fallback?
(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
It's best to strip digits at the voice port.
Max-dn|max-ephone|ip source-address
To notify SIP Phones of NTP
22. How do you enable AAR?
Device Pool Locations
SERVICE PARAMETER> Automated Alternate Routing Enable > True
Matches any length dialed number and truncates it to 4 digits.
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
23. How do do configure TEHO?
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
1 ... this is not optional!
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
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
24. Where can you assign the AAR Group?
When configuring TEHO.
TRUE
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
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
25. '<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
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
...
<i>voice register pool</i>|and|<i>voice register dn</i>
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 .)
26. What should you configure before entering auto qos?
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
Bandwidth
Assign a logout-profile to the ephone.
Phone number followed by domain name.|i.e. 3006@ipxcme.com
27. On a gatekeeper trunk - how do you ensure the Sub is used as primary and Pub as backup?
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
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.
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
24kb/s
28. In CME Where is the Calling Name derived from?
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
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.
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
The <i>name</i> configuration field in ephone-dn and voice register dn
29. What do you need to do to activate the CME GUI?
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
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
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
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
30. How do you see the details of calls coming in and out of the PRI?
Debug isdn q931
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.
Sh ccm-manager
It's best to strip digits at the voice port.
31. Is the order of the MRGs in an MRGL significant?
Assign a logout-profile to the ephone.
(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
...
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
32. 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
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
'Service Parameters --> <i>Auto Answer Timer</i>
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
33. When configuring a Gatekeep-controlled trunk - what do you do with the <i>Wait for Far End H.245 TCS</i> option?
Bottom up - to prevent glare.
(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
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
Put them in a route list.
34. What does an H323 GW require that MGCP GWs do not?
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Dial-peers
(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
TRUE
35. How do you use an ephone template?
Assign SIP Dial Rules
When they are explicitly matched in a destination-pattern in a dial-peer.
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
Apply it to the ephone.
36. How can you tell the difference between Proxy Line Mode and Shared Line Mode by just looking at the phone?
SIP Dial Rules
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
Debug isdn q931
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
37. Where is it best to manipulate digits for inbound calls?
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38. What is SIP URI?
Phone number followed by domain name.|i.e. 3006@ipxcme.com
#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
It's best to strip digits at the voice port.
IP Voice Media Streaming App
39. 'If you want to make changes to any softkeys where do you do it?
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
Sh ip rsvp reservation||sh sccp connections
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
40. 'How do you inform a SIP phone of NTP information?
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
(config)#<i>sh cdp neigh detail
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
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
41. 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.
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
(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
The <i>name</i> configuration field in ephone-dn and voice register dn
42. Describe how you configure SIP URI functionality.
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
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
43. What CUCM services should you activate?
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
Sh ip rsvp reservation||sh sccp connections
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
1 ... this is not optional!
44. If the PSTN is presenting on channel 1 - should you configure CUCM to search 'top down' or 'bottom up'?
To notify SIP Phones of NTP
Bottom up - to prevent glare.
Via the <i>voice hunt-group parallel</i> command
(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
45. How do you maintain 4-digit dialing to phones that are running in an SRST scenario (WAN outage)?
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46. What terminology translates to AAR?
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
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
Reroute when here is WAN congestion.
47. What are the commands to create the L3 routing interface for VLANS (SVI)?
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
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
Apply <i>voice-class codec</i>
Dial-peers
48. What CM Service needs to be start in Serviceability for MOH to work?
IP Voice Media Streaming App
Apply it to the ephone.
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
Dial-peers
49. 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
In CUCM - configure CFUR to point to it's E164 number.
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
Call forwarding between voip to voip (when CUBE is in play)
50. When it comes to H323 gateways - what should you always change - unless explicitly told not to?
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
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
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Top Down means that channel 1 will be the first channel used to place outgoing calls.