SUBJECTS
|
BROWSE
|
CAREER CENTER
|
POPULAR
|
JOIN
|
LOGIN
Business Skills
|
Soft Skills
|
Basic Literacy
|
Certifications
About
|
Help
|
Privacy
|
Terms
|
Email
Search
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. When it comes to H323 gateways - what should you always change - unless explicitly told not to?
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
<i>SIP Route Pattern</i> over a SIP Trunk.
(config)#voice register dialplan
2. On the RDP What is the Rerouting CSS used for?
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
Single Number Reach (Mobile Connect)
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
3. How do you configure phone ports on a 3750?
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
<b>Access Port Method</b>||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
4. How do you enable AAR system wide?
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
'Service Parameters --> <i>Auto Answer Timer</i>
5. How do you configure SIP phones to be able to call other SIP phones via SIP URI?
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
<i>SIP Route Pattern</i> over a SIP Trunk.
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
(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
6. 'How do you inform a SIP phone of NTP information?
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
<i>SIP Route Pattern</i> over a SIP Trunk.
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
Via the <i>voice hunt-group parallel</i> command
7. Which takes precedence Device Locations or Device Pool Locations?
When multiple sites use the same route pattern - do your digit manipulation on a route list.
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
Device Pool Locations
8. On an MGCP GW - how could you see the primary and backup CUCM servers?
Show ccm-manager
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
When they are explicitly matched in a destination-pattern in a dial-peer.
Service Parameters --> <i>Mobile Voice Access Number</i>
9. What are the 3 mandatory commands within call-manager-fallback?
...
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
Max-dn|max-ephone|ip source-address
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
10. What is Phone NTP Reference used for?
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
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
To notify SIP Phones of NTP
11. What would force you to use telephony-service to configure SRST?
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
Call forwarding between voip to voip (when CUBE is in play)
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
12. Where is it best to manipulate digits for inbound calls?
Warning
: Invalid argument supplied for foreach() in
/var/www/html/basicversity.com/show_quiz.php
on line
183
13. In CME Where is the Calling Name derived from?
The <i>name</i> configuration field in ephone-dn and voice register dn
Dial-peers
To enable two-stage dialing.
#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
14. How do you set the inter-digit timeout for SIP phones in CME?
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
(config)#voice register dialplan
...
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
15. How do you verify where MOH is being served up from?
Warning
: Invalid argument supplied for foreach() in
/var/www/html/basicversity.com/show_quiz.php
on line
183
16. What do you need to do to activate the CME GUI?
(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
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
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
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.
17. What is Mobile Voice Access?
<b>Access Port Method</b>||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</i>
18. How do you prevent toll fraud on CME?
#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
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
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 ....
Apply <i>voice-class codec</i>
19. <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
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
Apply it to the ephone.
...
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
20. What's the best way to do digit manipulation on an IOS gateway?
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
SIP Dial Rules
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
21. What is CSIM?
Precede the # with a > ... so 9011*>#
Call Simulator. You can use to validate path from router to the PSTN.
'Use before the ... so XXXX
22. What is SIP URI?
The Called/Calling Transformations are superceded.
Phone number followed by domain name.|i.e. 3006@ipxcme.com
Single Number Reach (Mobile Connect)
When they are explicitly matched in a destination-pattern in a dial-peer.
23. In CUCME what SIP config command is equivalent to ephone and ephone-dn?
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
<i>voice register pool</i>|and|<i>voice register dn</i>
24. What are the commands to ensure an H323 GW always uses the Sub as primary and Pub as a backup?
Assign a logout-profile to the ephone.
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 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
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
25. Name 4 useful show commands for active calls.
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
<i>#after-hours block pattern</i>
26. How do you see the details of calls coming in and out of the PRI?
'Use before the ... so XXXX
Single Number Reach (Mobile Connect)
Debug isdn q931
(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
27. When can you not use a Standard Local Route Group?
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
When configuring TEHO.
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
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
28. How do you allow the Calling Name to be sent to the PSTN on a router?
#voice service voip| #no supplementary-service h225-notify cid-update
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
Sh ccm-manager
29. 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
When multiple sites use the same route pattern - do your digit manipulation on a route list.
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
CUCM OS Administration Settings --> NTP Servers
30. When are digits stripped in a gateway?
When they are explicitly matched in a destination-pattern in a dial-peer.
#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
CUCM OS Administration Settings --> NTP Servers
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
31. What does <i>ccm-manager config server [IP]</ip> do?
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</i>
Device Pool Locations
Automatically configures the MGCP GW for you.
32. Two useful troubleshooting commands for CUCME?
The <i>name</i> configuration field in ephone-dn and voice register dn
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.
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
33. Is the + character supported on a VOIP dial-peer?
Warning
: Invalid argument supplied for foreach() in
/var/www/html/basicversity.com/show_quiz.php
on line
183
34. On an MGCP GW - if you select <i>Top Down</i> under Channel Selection Order - What does that mean?
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
Master/Slave relationship. CUCM controls it.
In CUCM - configure CFUR to point to it's E164 number.
Top Down means that channel 1 will be the first channel used to place outgoing calls.
35. How would you verify that DHCP is working in IOS?
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
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.
(config)#<i>sh cdp neigh detail
36. A <i>show isdn status</i> shows TEI_ASSIGNED. What do you do to get MULTIPLE_FRAME_ESTABLISHED?
Precede the # with a > ... so 9011*>#
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
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.
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
37. How do you convert a full PRI 9config'd by ccm-manager config server) back to a Fractional PRI?
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
Apply <i>voice-class codec</i>
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
38. Privacy is enabled system-wide in CUCM by default. (T or F)
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
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
Dial-peers
TRUE
39. 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?
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
<i>Auto Call Pickup Enabled</i>
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
40. When using <i>drop-through-option</i> What is the max number of huntgroups?
#voice service voip| #no supplementary-service h225-notify cid-update
Dtmf-relay h245-alpha
Sends the Calling Name.
1 ... this is not optional!
41. When a device is roaming to a different site How does it get the <i>Roaming Sensitive Settings</i> from the Device Pool?
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
8kb/s
Mobile Voice Access
42. What's the difference between AAR and SRST?
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
Apply it to the ephone.
Reroute when here is WAN congestion.
43. 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
44. What are the steps to integrate CUPS with 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
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
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
45. 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?
(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
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
Allows you to transfer by only pressing the Transfer button once.
46. How do you configure a gateway to register with gatekeeper?
Configure it on the route group through the route list - then it will be local to the route list.
#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
Assign a logout-profile to the ephone.
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
47. What does KPML do?
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
At the CLI: <i>utils ntp status</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
48. How do you create a trunk on the switch side?
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
<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)
49. What is the <i>Enable Enterprise Feature Access</i> Service Parameter used for?
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
To enable two-stage dialing.
Reroute when there is a WAN Outage.
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
50. How do you test a Voice Translation Rule?
#test voice translation rule 1 <input to test>
Apply it to the ephone.
SIP Dial Rules
Apply <i>voice-class codec</i>