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. An MGCP gateway serving as an SRST router requires what?
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
Master/Slave relationship. CUCM controls it.
Precede the # with a > ... so 9011*>#
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
2. Which takes precedence Device Locations or Device Pool Locations?
24kb/s
Device Pool Locations
In CUCM - configure CFUR to point to it's E164 number.
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
3. How do you enable AAR system wide?
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
Configure it on the route group through the route list - then it will be local to the route list.
4. How do do configure TEHO?
Automatically configures the MGCP GW for you.
Call Simulator. You can use to validate path from router to the PSTN.
#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
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
5. What dtmf-relay type do you use for an H323 GW?
#voice service voip| #no supplementary-service h225-notify cid-update
It's best to strip digits at the voice port.
Dtmf-relay h245-alpha
6. What are the commands to configure an H323 GW?
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
(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
#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
...
7. What should you configure before entering auto qos?
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
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
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
Bandwidth
8. On an H323 GW - how do you adjust the timers for redundancy hunting?
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
(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
#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
9. What is the <i>Enable Enterprise Feature Access</i> Service Parameter used for?
To enable two-stage dialing.
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
Apply it to the ephone.
10. How do you configure CUBE?
11. When do you use translate called? Translate calling?
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
Reroute when here is WAN congestion.
Reroute when there is a WAN Outage.
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
12. Name 2 commands to verify RSVP functionality.
Sh ip rsvp reservation||sh sccp connections
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
(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
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
13. In CUCME what SIP config command is equivalent to ephone and ephone-dn?
<i>voice register pool</i>|and|<i>voice register dn</i>
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
When multiple sites use the same route pattern - do your digit manipulation on a route list.
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
14. How would you verify that DHCP is working in IOS?
...
Precede the # with a > ... so 9011*>#
(config)#<i>sh cdp neigh detail
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
15. On an MGCP GW - if you select <i>Top Down</i> under Channel Selection Order - What does that mean?
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
Reroute when here is WAN congestion.
Top Down means that channel 1 will be the first channel used to place outgoing calls.
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
16. What is a best practice for digit manipulation - in regards to H323 GWs?
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
Mobile Voice Access
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
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
17. How do you verify that NTP is working on the CUCM server?
At the CLI: <i>utils ntp status</i>
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
Precede the # with a > ... so 9011*>#
18. On the RDP What is the CSS used for?
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
Sends the Calling Name.
...
Mobile Voice Access
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
Single Number Reach (Mobile Connect)
Bandwidth
...
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)
20. What command is needed on a gateway for it to invoke the DSP to convert multicast MOH stream to TDM?
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.
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
On CUCM it's identical to adding an H323 GW.
Ccm-manager music-on-hold
21. When do you do digit manipulation at the route pattern as opposed to the route list?
...
(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
When multiple sites use the same route pattern - do your digit manipulation on a route list.
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.
22. What is CSIM?
Put them in a route list.
#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 Simulator. You can use to validate path from router to the PSTN.
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 .)
23. What are the commands to configure SIP phones in CME?
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
In CUCM - configure CFUR to point to it's E164 number.
<b>Access Port Method</b>||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
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
24. How do you enable Extension Mobility for a device in CME?
Assign a logout-profile to the ephone.
Outbound dial-peers
Mobile Connect
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
25. What CM Service needs to be start in Serviceability for MOH to work?
Bandwidth
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
IP Voice Media Streaming App
26. How do you prevent toll fraud on CME?
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.
Show ccm-manager
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 ....
1 ... this is not optional!
27. <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
...
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
Sends the Calling Name.
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.
28. What is SIP URI?
#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
Call forwarding between voip to voip (when CUBE is in play)
Matches any length dialed number and truncates it to 4 digits.
Phone number followed by domain name.|i.e. 3006@ipxcme.com
29. Privacy is enabled system-wide in CUCM by default. (T or F)
TRUE
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
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.
30. How do you change modes in <i>voice register global</i>?
Default is SRST||<i>mode cme</i> changes to CME|<i>no mode cme</i> changes back to SRST
Configure it on the route group through the route list - then it will be local to the route list.
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
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
31. When are digits stripped in a gateway?
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
When they are explicitly matched in a destination-pattern in a dial-peer.
Sh ccm-manager
SIP Dial Rules
32. If you need terminate and reoriginate all calls through the CUBE - what do you need to do.
Call Simulator. You can use to validate path from router to the PSTN.
Assign SIP Dial Rules
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
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
33. When using GK redundancy - what must you do to ensure both SUB and PUB use port 1720 for call-signalling.
34. How do you configure DHCP in IOS?
(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
To enable two-stage dialing.
<i>voice register pool</i>|and|<i>voice register dn</i>
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
35. When configuring a Gatekeep-controlled trunk - what do you do with the <i>Wait for Far End H.245 TCS</i> option?
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
SERVICE PARAMETER> Automated Alternate Routing Enable > True
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
36. When using <i>drop-through-option</i> What is the max number of huntgroups?
1 ... this is not optional!
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</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
37. Aside from the <i>Media Resources --> Mobile Voice Access</i> where else do you need to specify the the Mobile Voice Access DN?
Service Parameters --> <i>Mobile Voice Access Number</i>
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
Default is SRST||<i>mode cme</i> changes to CME|<i>no mode cme</i> changes back to SRST
38. How do you configure SIP phones to be able to call other SIP phones via SIP URI?
(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
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.
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
39. Where do you use VIA zone?
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
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
SERVICE PARAMETER> Automated Alternate Routing Enable > True
40. How is NTP sych setup in CUCM?
CUCM OS Administration Settings --> NTP Servers
1 ... this is not optional!
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 forwarding between voip to voip (when CUBE is in play)
41. What does <i>rule 1 /^.*(....$)/ /1/</i> do?
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
Matches any length dialed number and truncates it to 4 digits.
1 ... this is not optional!
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
42. 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
(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
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
43. With the gw-priority command - does higher or lower priority take precedence?
Higher
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
The Called/Calling Transformations are superceded.
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 ....
44. What do you need to do to activate the CME GUI?
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
Master/Slave relationship. CUCM controls it.
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</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
45. '<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
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
...
#test voice translation rule 1 <input to test>
46. How do you use the # as a string terminator within a SIP Dial Rule?
Top Down means that channel 1 will be the first channel used to place outgoing calls.
...
Precede the # with a > ... so 9011*>#
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
47. How do you configure AAR?
Call forwarding between voip to voip (when CUBE is in play)
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
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
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
48. How do you see multicast packets being sent?
Bottom up - to prevent glare.
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
Debug ephone moh
<i>Auto Call Pickup Enabled</i>
49. What CUCM services should you activate?
#test voice translation rule 1 <input to test>
...
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
50. What is Mobile Voice Access?
'Use before the ... so XXXX
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)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify