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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's the best way to do digit manipulation on an IOS gateway?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Bandwidth
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
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
2. How would you set the the timer for Auto Answer?
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3. How do you ensure that G.711 only is used?
When they are explicitly matched in a destination-pattern in a dial-peer.
IP Voice Media Streaming App
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
4. What commands are needed to configure the voice register pool in CME?
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 .)
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 service voip| #no supplementary-service h225-notify cid-update
Mobile Connect
5. How do you block an external call from being transferred back out to the pstn by an internal user?
Assign a logout-profile to the ephone.
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
Sh ccm-manager
6. How do you configure SIP phones to be able to call other SIP phones via SIP URI?
TRUE
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
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
<i>SIP Route Pattern</i> over a SIP Trunk.
7. What command is needed on a gateway for it to invoke the DSP to convert multicast MOH stream to TDM?
(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
8kb/s
Ccm-manager music-on-hold
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
8. What terminology translates to SRST?
Reroute when there is a WAN Outage.
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
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)#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
9. How do you use the # as a string terminator within a SIP Dial Rule?
Bandwidth
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 Called/Calling Transformations are superceded.
Precede the # with a > ... so 9011*>#
10. How do you support multiple codecs on a dialpeer?
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
Allows you to transfer by only pressing the Transfer button once.
SIP Dial Rules
Apply <i>voice-class codec</i>
11. Aside from the <i>Media Resources --> Mobile Voice Access</i> where else do you need to specify the the Mobile Voice Access DN?
'Use before the ... so XXXX
Put them in a route list.
Service Parameters --> <i>Mobile Voice Access Number</i>
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
12. What is a best practice for digit manipulation - in regards to H323 GWs?
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
'Use before the ... so XXXX
Higher
13. How do you configure a gateway to register with gatekeeper?
#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
Show ccm-manager
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
14. 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
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
<i>Auto Call Pickup Enabled</i>
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
15. What is CSIM?
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.
Call Simulator. You can use to validate path from router to the PSTN.
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
Dial-peers
16. Where is it best to manipulate digits for inbound calls?
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17. When using <i>drop-through-option</i> What is the max number of huntgroups?
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
1 ... this is not optional!
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.
#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
18. 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?
Device Pool Locations
Dtmf-relay h245-alpha
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)
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
19. How do you test a Voice Translation Rule?
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
When multiple sites use the same route pattern - do your digit manipulation on a route list.
#test voice translation rule 1 <input to test>
Sends the Calling Name.
20. What are the base telephony-service commands for CME?
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
Sends the Calling Name.
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
21. When you make a call and it results in a secondary dial-tone - What is this a symptom of?
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 ....
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.
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
'Service Parameters --> <i>Auto Answer Timer</i>
22. If you need terminate and reoriginate all calls through the CUBE - what do you need to do.
Outbound dial-peers
Top Down means that channel 1 will be the first channel used to place outgoing calls.
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
23. What is the <i>Enable Enterprise Feature Access</i> Service Parameter used for?
To enable two-stage dialing.
<i>#after-hours block pattern</i>
Matches any length dialed number and truncates it to 4 digits.
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
24. What are the commands to configure an H323 GW?
#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
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
IP Voice Media Streaming App
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
25. When using GK redundancy - what must you do to ensure both SUB and PUB use port 1720 for call-signalling.
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26. How do you configure an MGCP GW? (router side)
Apply it to the ephone.
Automatically configures the MGCP GW for you.
CUCM OS Administration Settings --> NTP Servers
(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
27. If the PSTN is presenting on channel 1 - should you configure CUCM to search 'top down' or 'bottom up'?
#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
Bottom up - to prevent glare.
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
28. How do you enable AAR system wide?
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
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
Precede the # with a > ... so 9011*>#
(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
29. How do you enable AAR?
Apply <i>voice-class codec</i>
SERVICE PARAMETER> Automated Alternate Routing Enable > True
To enable two-stage dialing.
(config)#voice register dialplan
30. What is SIP URI?
Precede the # with a > ... so 9011*>#
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
Phone number followed by domain name.|i.e. 3006@ipxcme.com
31. What are the commands to ensure an H323 GW always uses the Sub as primary and Pub as a backup?
#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
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
32. What does KPML do?
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
<i>voice register pool</i>|and|<i>voice register dn</i>
Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</i>
On CUCM it's identical to adding an H323 GW.
33. How do you prioritize route groups?
Put them in a route list.
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
Automatically configures the MGCP GW for you.
34. How do you configure phone ports on an ESW?
Device Pool Locations
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
TRUE
Assign a logout-profile to the ephone.
35. 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
To enable two-stage dialing.
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
36. How do you allow the Calling Name to be sent to the PSTN on a router?
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
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.
...
Sh ccm-manager
37. What should you configure before entering auto qos?
Master/Slave relationship. CUCM controls it.
...
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
Bandwidth
38. How do you convert a full PRI 9config'd by ccm-manager config server) back to a Fractional PRI?
Max-dn|max-ephone|ip source-address
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.
39. What are the steps to configure Single Number Reach and Mobile Voice Access?
Device Pool Locations
To enable two-stage dialing.
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
40. With the gw-priority command - does higher or lower priority take precedence?
CUCM OS Administration Settings --> NTP Servers
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
When multiple sites use the same route pattern - do your digit manipulation on a route list.
Higher
41. How do you configure SRST?
<i>SIP Route Pattern</i> over a SIP Trunk.
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
#test voice translation rule 1 <input to test>
42. What is call-forward pattern used for?
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
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
Matches any length dialed number and truncates it to 4 digits.
Call forwarding between voip to voip (when CUBE is in play)
43. 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
Mobile Connect
Configure it on the route group through the route list - then it will be local to the route list.
On CUCM it's identical to adding an H323 GW.
44. Does CUCM support RSVP natively?
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
45. 'How do you inform a SIP phone of NTP information?
(config)#<i>sh cdp neigh detail
(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
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
Outbound dial-peers
46. What are the steps to integrate CUPS with CUCM?
Device Pool Locations
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
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
...
47. What's the difference between AAR and SRST?
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
SERVICE PARAMETER> Automated Alternate Routing Enable > True
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
Assign SIP Dial Rules
48. What are the commands to create vlans on an ESW?
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
Assign a logout-profile to the ephone.
49. Which takes precedence Device Locations or Device Pool Locations?
Single Number Reach (Mobile Connect)
Phone number followed by domain name.|i.e. 3006@ipxcme.com
Device Pool Locations
50. What CUCM services should you activate?
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.