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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 command is needed on a gateway for it to invoke the DSP to convert multicast MOH stream to TDM?
Ccm-manager music-on-hold
Mobile Voice Access
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
2. How do you configure DHCP in IOS?
Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</i>
Put them in a route list.
Phone number followed by domain name.|i.e. 3006@ipxcme.com
(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
3. When you make a call and it results in a secondary dial-tone - What is this a symptom of?
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
Assign SIP Dial Rules
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
#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
4. <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
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
...
'Use before the ... so XXXX
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
5. 'If you're not getting a DHCP address from CUCM what then?
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
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
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
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
6. How do you ensure that G.711 only is used?
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
When configuring TEHO.
Service Parameters --> <i>Mobile Voice Access Number</i>
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
7. Which takes precedence Device Locations or Device Pool Locations?
Device Pool Locations
Bandwidth
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
Apply it to the ephone.
8. How do you configure SIP phones to be able to call other SIP phones via SIP URI?
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>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.
<i>SIP Route Pattern</i> over a SIP Trunk.
9. 'If you want to make changes to any softkeys where do you do it?
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
To notify SIP Phones of NTP
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
10. What are the commands to configure a SIP phone in CUCME?
<i>#after-hours block pattern</i>
Call forwarding between voip to voip (when CUBE is in play)
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</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
11. How do you enable Extension Mobility for a device in CME?
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
Assign a logout-profile to the ephone.
It's best to strip digits at the voice port.
Outbound dial-peers
12. How do do configure TEHO?
Bandwidth
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
Phone number followed by domain name.|i.e. 3006@ipxcme.com
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
13. When setting up SIP URI where do you configure the CUCM's domain name?'
Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</i>
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
Mobile Connect
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
14. How do you block calls under call-manager-fallback?
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
<i>#after-hours block pattern</i>
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
15. What is call-forward pattern used for?
Call forwarding between voip to voip (when CUBE is in play)
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-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
16. Name 2 commands to verify RSVP functionality.
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
Reroute when there is a WAN Outage.
Sh ip rsvp reservation||sh sccp connections
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
17. On a gatekeeper trunk - how do you ensure the Sub is used as primary and Pub as backup?
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
Debug ephone moh
18. How do you create a trunk on the switch side?
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
19. How do you prioritize route groups?
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
Put them in a route list.
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
Assign a logout-profile to the ephone.
20. What are the commands to configure SIP phones in CME?
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
'Service Parameters --> <i>Auto Answer Timer</i>
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</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
21. What are the commands to manually configure an MGCP gateway?
#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
(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
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
Ccm-manager music-on-hold
22. An MGCP gateway serving as an SRST router requires what?
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
<i>SIP Route Pattern</i> over a SIP Trunk.
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
23. On the RDP What is the Rerouting CSS used for?
At the CLI: <i>utils ntp status</i>
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
Single Number Reach (Mobile Connect)
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
24. On an MGCP GW - if you select <i>Top Down</i> under Channel Selection Order - What does that mean?
Sh ip rsvp reservation||sh sccp connections
Top Down means that channel 1 will be the first channel used to place outgoing calls.
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
Apply <i>voice-class codec</i>
25. What are the commands to create vlans on an ESW?
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
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
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
<i>#after-hours block pattern</i>
26. How do you configure class of service (CoS) in CUCM? CME?
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
Higher
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
27. 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
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
<i>Auto Call Pickup Enabled</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
28. When can you not use a Standard Local Route Group?
Bandwidth
When configuring TEHO.
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.
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
29. How do you prevent H323 caller-id updates to CUCM
#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 service voip| #no supplementary-service h225-notify cid-update
Higher
Mobile Connect
30. What do you need to do to activate the CME GUI?
#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
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
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
31. What are the steps to integrate CUPS with CUCM?
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
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
Mobile Voice Access
To enable two-stage dialing.
32. How do you allow the Calling Name to be sent to the PSTN on a router?
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
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
33. What should you configure before entering auto qos?
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
The <i>name</i> configuration field in ephone-dn and voice register dn
Allows you to transfer by only pressing the Transfer button once.
Bandwidth
34. 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?
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
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
1 ... this is not optional!
Apply it to the ephone.
35. How do you allow H323 calls to be preserved should the primary H323 GW fail?
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Mobile Voice Access
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)
36. Maximum Wait Time for Desk Pickup?
Precede the # with a > ... so 9011*>#
Mobile Connect
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
#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
37. 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?
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
To enable two-stage dialing.
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
38. When do you do digit manipulation at the route pattern as opposed to the route list?
Call Simulator. You can use to validate path from router to the PSTN.
<i>SIP Route Pattern</i> over a SIP Trunk.
Put them in a route list.
When multiple sites use the same route pattern - do your digit manipulation on a route list.
39. If the PSTN is presenting on channel 1 - should you configure CUCM to search 'top down' or 'bottom up'?
Bottom up - to prevent glare.
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
Automatically configures the MGCP GW for you.
(config)#<i>sh cdp neigh detail
40. Where can you assign the AAR Group?
(config)#<i>sh cdp neigh detail
Call forwarding between voip to voip (when CUBE is in play)
Mobile Connect
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
41. What is the Service Parameter to allow the phone that is <i>picking ip</i> to auto answer?
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
In CUCM - configure CFUR to point to it's E164 number.
<i>Auto Call Pickup Enabled</i>
...
42. How do you see the details of calls coming in and out of the PRI?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Bandwidth
...
Debug isdn q931
43. How do you prevent toll fraud on CUCM?
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
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>
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
44. How do you configure SRST?
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
#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
Sh ip rsvp reservation||sh sccp connections
45. How do you convert a full PRI 9config'd by ccm-manager config server) back to a Fractional PRI?
Dial-peers
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
46. What is Phone NTP Reference used for?
To notify SIP Phones of NTP
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
Matches any length dialed number and truncates it to 4 digits.
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
47. In a SIP Dial Rule how do you represent a dialed * instead of it being used as a wildcard?
Warning
: Invalid argument supplied for foreach() in
/var/www/html/basicversity.com/show_quiz.php
on line
183
48. When it comes to H323 gateways - what should you always change - unless explicitly told not to?
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
Mobile Voice Access
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
49. 'How do you inform a SIP phone of NTP information?
Configure it on the route group through the route list - then it will be local to the route list.
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
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
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
50. In CME Where is the Calling Name derived from?
The <i>name</i> configuration field in ephone-dn and voice register dn
To enable two-stage dialing.
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
Apply it to the ephone.