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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 create a trunk on the switch side?
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 .)
SERVICE PARAMETER> Automated Alternate Routing Enable > True
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
<i>voice register pool</i>|and|<i>voice register dn</i>
2. What commands are needed to configure the voice register pool in CME?
Allows you to transfer by only pressing the Transfer button once.
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
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
CUCM OS Administration Settings --> NTP Servers
3. When using <i>drop-through-option</i> What is the max number of huntgroups?
1 ... this is not optional!
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
When configuring TEHO.
4. What are the steps to integrate CUPS with CUCM?
#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
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
8kb/s
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
5. 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>
It's best to strip digits at the voice port.
CUCM OS Administration Settings --> NTP Servers
SERVICE PARAMETER> Automated Alternate Routing Enable > True
6. How would you verify that DHCP is working in IOS?
(config)#<i>sh cdp neigh detail
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
Bottom up - to prevent glare.
Device Pool Locations
7. When do you do digit manipulation at the route pattern as opposed to the route list?
When multiple sites use the same route pattern - do your digit manipulation on a route list.
#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
#test voice translation rule 1 <input to test>
<i>voice register pool</i>|and|<i>voice register dn</i>
8. What should you configure before entering auto qos?
At the CLI: <i>utils ntp status</i>
Bandwidth
1 ... this is not optional!
It's best to strip digits at the voice port.
9. What is the full E164 format?
To enable two-stage dialing.
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
Mobile Voice Access
10. How do you configure a gateway to register with gatekeeper?
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
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
SIP Dial Rules
#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
11. How do you allow the Calling Name to be sent to the PSTN on a router?
(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
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
Bottom up - to prevent glare.
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
12. How do you disable KPML?
Matches any length dialed number and truncates it to 4 digits.
Assign SIP Dial Rules
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
Mobile Voice Access
13. Is the + character supported on a VOIP dial-peer?
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14. Name 4 useful show commands for active calls.
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
(config)#<i>sh cdp neigh detail
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
15. Maximum Wait Time for Desk Pickup?
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
16. What are the commands to configure a SIP phone in CUCME?
Debug isdn q931
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
Default is SRST||<i>mode cme</i> changes to CME|<i>no mode cme</i> changes back to SRST
Phone number followed by domain name.|i.e. 3006@ipxcme.com
17. What must you do for a BACD script to work on a CME router?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
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
18. How do you configure phone ports on an ESW?
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
Sends the Calling Name.
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
Show ccm-manager
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
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
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
...
(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
20. What does Display-IE do?
Precede the # with a > ... so 9011*>#
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
Sends the Calling Name.
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
21. 'If you're not getting a DHCP address from CUCM what then?
Assign a logout-profile to the ephone.
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
22. Describe how you configure SIP URI functionality.
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
(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
23. What are the basic SCCP Commands fro telephony-service in CUCME?
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
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
(config)#voice register dialplan
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.
24. How do you configure a SIP Trunk? (router side)
(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
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
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
25. What is the bit rate for a G.729 call excluding layer 2?
8kb/s
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
CUCM OS Administration Settings --> NTP Servers
26. What do you need to do to activate the CME GUI?
Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</i>
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
Put them in a route list.
Debug isdn q931
27. Privacy is enabled system-wide in CUCM by default. (T or F)
<b>Access Port Method</b>||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
TRUE
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
28. What is Phone NTP Reference used for?
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
#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
'Service Parameters --> <i>Auto Answer Timer</i>
To notify SIP Phones of NTP
29. How do you configure CUCM redundancy on an H323 gateway?
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 ....
#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
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
30. What CUCM services should you activate?
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
31. How do you test a Voice Translation Rule?
#test voice translation rule 1 <input to test>
Reroute when there is a WAN Outage.
Default is SRST||<i>mode cme</i> changes to CME|<i>no mode cme</i> changes back to SRST
32. Need to Know||Anytime you have an external call - it needs to match on a route pattern.
(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
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
33. How do you prioritize route groups?
Put them in a route list.
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
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
34. How can you tell the difference between Proxy Line Mode and Shared Line Mode by just looking at the phone?
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 .)
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
On CUCM it's identical to adding an H323 GW.
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
35. What are the commands to configure NTP in IOS?
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
Bottom up - to prevent glare.
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
36. How would you set the the timer for Auto Answer?
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37. How do you see multicast packets being sent?
Debug ephone moh
Max-dn|max-ephone|ip source-address
Service Parameters --> <i>Mobile Voice Access Number</i>
Debug isdn q931
38. Which takes precedence Device Locations or Device Pool Locations?
Bottom up - to prevent glare.
Device Pool Locations
'Use before the ... so XXXX
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
39. What is Mobile Voice Access?
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
(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
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
40. How do you enable AAR system wide?
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
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)
41. How do you get around relying on DNS for your CUCMs?
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
<i>#after-hours block pattern</i>
Higher
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
42. What are the steps to configure Single Number Reach and Mobile Voice Access?
Top Down means that channel 1 will be the first channel used to place outgoing calls.
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
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.
43. What does <i>rule 1 /^.*(....$)/ /1/</i> do?
Matches any length dialed number and truncates it to 4 digits.
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.
'Service Parameters --> <i>Auto Answer Timer</i>
44. When do you use translate called? Translate calling?
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
Reroute when here is WAN congestion.
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
45. When you make a call and it results in a secondary dial-tone - What is this a symptom of?
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
Top Down means that channel 1 will be the first channel used to place outgoing calls.
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
Master/Slave relationship. CUCM controls it.
46. How do you ensure that G.711 only is used?
Higher
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
47. How do you maintain 4-digit dialing to phones that are running in an SRST scenario (WAN outage)?
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48. What is CSIM?
#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
Call Simulator. You can use to validate path from router to the PSTN.
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
49. In CME Where is the Calling Name derived from?
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
The <i>name</i> configuration field in ephone-dn and voice register dn
Matches any length dialed number and truncates it to 4 digits.
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
50. On an H323 GW - how do you adjust the timers for redundancy hunting?
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
<i>Auto Call Pickup Enabled</i>
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
Top Down means that channel 1 will be the first channel used to place outgoing calls.