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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. 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>
On CUCM it's identical to adding an H323 GW.
'Use before the ... so XXXX
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
2. What are the commands to ensure an H323 GW always uses the Sub as primary and Pub as a backup?
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
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
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.
3. What does <i>rule 1 /^.*(....$)/ /1/</i> do?
Matches any length dialed number and truncates it to 4 digits.
1 ... this is not optional!
On CUCM it's identical to adding an H323 GW.
...
4. Where can you find the command to restore any of the default IP Phone Services if they happen to have been deleted?
CUCM OS Administration Settings --> NTP Servers
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
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
5. What does an H323 GW require that MGCP GWs do not?
#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
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.
Dial-peers
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.
6. Name 2 commands to verify RSVP functionality.
Sh ip rsvp reservation||sh sccp connections
On CUCM it's identical to adding an H323 GW.
SERVICE PARAMETER> Automated Alternate Routing Enable > True
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 .)
7. What is a best practice for digit manipulation - in regards to H323 GWs?
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
8. What is the <i>Enable Enterprise Feature Access</i> Service Parameter used for?
To enable two-stage dialing.
Via the <i>voice hunt-group parallel</i> command
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
9. How would you verify that DHCP is working in IOS?
(config)#<i>sh cdp neigh detail
SERVICE PARAMETER> Automated Alternate Routing Enable > True
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
Higher
10. Which takes precedence Device Locations or Device Pool Locations?
Master/Slave relationship. CUCM controls it.
Device Pool Locations
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
At the CLI: <i>utils ntp status</i>
11. <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
...
(config)#<i>sh cdp neigh detail
1 ... this is not optional!
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
12. Is the order of the MRGs in an MRGL significant?
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
Mobile Voice Access
(config)#<i>sh cdp neigh detail
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
13. How much bandwidth does a G.729 call including layer 3 require?
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
24kb/s
Master/Slave relationship. CUCM controls it.
SIP Dial Rules
14. What commands are needed to configure the voice register pool in CME?
#voice service voip| #no supplementary-service h225-notify cid-update
On CUCM it's identical to adding an H323 GW.
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
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
15. Aside from the <i>Media Resources --> Mobile Voice Access</i> where else do you need to specify the the Mobile Voice Access DN?
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
Service Parameters --> <i>Mobile Voice Access Number</i>
Default is SRST||<i>mode cme</i> changes to CME|<i>no mode cme</i> changes back to SRST
Allows you to transfer by only pressing the Transfer button once.
16. How do you use an ephone template?
At the CLI: <i>utils ntp status</i>
Apply it to the ephone.
Sends the Calling Name.
8kb/s
17. How do you create a trunk on the router side?
Single Number Reach (Mobile Connect)
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
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
Service Parameters --> <i>Mobile Voice Access Number</i>
18. What terminology translates to SRST?
At the CLI: <i>utils ntp status</i>
Reroute when there is a WAN Outage.
Outbound dial-peers
19. What are the commands to configure an H323 GW?
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.
#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
Allows you to transfer by only pressing the Transfer button once.
#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
20. How do you configure a gateway to register with gatekeeper?
Sh ccm-manager
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
#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
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
21. How do you ensure that G.711 only is used?
Default is SRST||<i>mode cme</i> changes to CME|<i>no mode cme</i> changes back to SRST
When configuring TEHO.
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
22. 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.
<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
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
23. How do you disable KPML?
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
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
Assign SIP Dial Rules
24. What is the Service Parameter to allow the phone that is <i>picking ip</i> to auto answer?
#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
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
Sends the Calling Name.
<i>Auto Call Pickup Enabled</i>
25. How do you enable AAR system wide?
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
The Called/Calling Transformations are superceded.
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
26. Two useful troubleshooting commands for CUCME?
#voice service voip| #no supplementary-service h225-notify cid-update
It's best to strip digits at the voice port.
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
27. How do you test a Voice Translation Rule?
Outbound dial-peers
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
#test voice translation rule 1 <input to test>
CUCM OS Administration Settings --> NTP Servers
28. How do you verify that NTP is working on the CUCM server?
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
At the CLI: <i>utils ntp status</i>
IP Voice Media Streaming App
29. What are the basic SCCP Commands fro telephony-service in CUCME?
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
In CUCM - configure CFUR to point to it's E164 number.
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
30. How do you convert a full PRI 9config'd by ccm-manager config server) back to a Fractional PRI?
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.
(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
31. How do you configure AAR?
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
32. When a CUCM device dials a number - what happens?
Sh ccm-manager
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 .)
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
SIP Dial Rules
33. What are two caveats to using the <i>ccm-manager config server</ip> command?
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.
Put them in a route list.
Reroute when there is a WAN Outage.
34. When configuring a Gatekeep-controlled trunk - what do you do with the <i>Wait for Far End H.245 TCS</i> option?
At the CLI: <i>utils ntp status</i>
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
<i>SIP Route Pattern</i> over a SIP Trunk.
35. What are the commands to configure NTP in IOS?
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
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
Ccm-manager music-on-hold
36. How do you change modes in <i>voice register global</i>?
24kb/s
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
Default is SRST||<i>mode cme</i> changes to CME|<i>no mode cme</i> changes back to SRST
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
37. How do do configure TEHO?
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
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
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
TRUE
38. Does CUCM support RSVP natively?
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
When they are explicitly matched in a destination-pattern in a dial-peer.
#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 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
39. On an MGCP GW - how could you see the primary and backup CUCM servers?
Show ccm-manager
(config)#voice register dialplan
...
Dtmf-relay h245-alpha
40. Describe how you configure SIP URI functionality.
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
'Use before the ... so XXXX
(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
41. How do you maintain 4-digit dialing to phones that are running in an SRST scenario (WAN outage)?
42. What are the steps to configure Single Number Reach and Mobile Voice Access?
(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
Service Parameters --> <i>Mobile Voice Access Number</i>
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
43. How do you configure class of service (CoS) in CUCM? CME?
24kb/s
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
44. Where is it best to manipulate digits for inbound calls?
45. How do you ensure no inter-digit timeout fro SIP phones within CUCM?
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Device Pool Locations
Assign a logout-profile to the ephone.
SIP Dial Rules
46. What must you do for a BACD script to work on a CME router?
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
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
47. What's the best way to do digit manipulation on an IOS gateway?
#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
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
#test voice translation rule 1 <input to test>
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
48. How do you allow H323 calls to be preserved should the primary H323 GW fail?
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
1 ... this is not optional!
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
49. How do you create a trunk on the switch side?
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
Top Down means that channel 1 will be the first channel used to place outgoing calls.
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
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)
50. How do you support multiple codecs on a dialpeer?
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
Apply <i>voice-class codec</i>
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 ....