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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 are two caveats to using the <i>ccm-manager config server</ip> command?
Via the <i>voice hunt-group parallel</i> command
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
When multiple sites use the same route pattern - do your digit manipulation on a route list.
2. What is the Service Parameter to allow the phone that is <i>picking ip</i> to auto answer?
<i>Auto Call Pickup Enabled</i>
Assign a logout-profile to the ephone.
SIP Dial Rules
(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
3. How do you verify where MOH is being served up from?
4. How do you configure SRST?
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
To enable two-stage dialing.
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.
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
5. What is SIP URI?
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
IP Voice Media Streaming App
Phone number followed by domain name.|i.e. 3006@ipxcme.com
Call Simulator. You can use to validate path from router to the PSTN.
6. How do you allow H323 calls to be preserved should the primary H323 GW fail?
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
7. What are the 3 mandatory commands within call-manager-fallback?
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 the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
Max-dn|max-ephone|ip source-address
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 ....
8. Where can you find the command to restore any of the default IP Phone Services if they happen to have been deleted?
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
'Use before the ... so XXXX
Dtmf-relay h245-alpha
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
9. How do you configure class of service (CoS) in CUCM? CME?
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
Sh ip rsvp reservation||sh sccp connections
Top Down means that channel 1 will be the first channel used to place outgoing calls.
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
10. When using GK redundancy - what must you do to ensure both SUB and PUB use port 1720 for call-signalling.
11. On an MGCP GW - if you select <i>Top Down</i> under Channel Selection Order - What does that mean?
(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
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
Top Down means that channel 1 will be the first channel used to place outgoing calls.
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
12. In a SIP Dial Rule how do you represent a dialed * instead of it being used as a wildcard?
13. When configuring a Gatekeep-controlled trunk - what do you do with the <i>Wait for Far End H.245 TCS</i> option?
The Called/Calling Transformations are superceded.
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
Dial-peers
14. Where is it best to manipulate digits for inbound calls?
15. How do you get around relying on DNS for your CUCMs?
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
16. Calls coming from CUCM to PSTN need what?
Service Parameters --> <i>Mobile Voice Access Number</i>
Outbound dial-peers
On CUCM it's identical to adding an H323 GW.
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
17. What is call-forward pattern used for?
Call forwarding between voip to voip (when CUBE is in play)
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
Assign SIP Dial Rules
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
18. What are the commands to create vlans on an ESW?
To enable two-stage dialing.
#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
24kb/s
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
19. 'How do you inform a SIP phone of NTP information?
Bandwidth
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
20. What dtmf-relay type do you use for an H323 GW?
(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
Dtmf-relay h245-alpha
Sends the Calling Name.
21. How do you block calls under call-manager-fallback?
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
...
<i>#after-hours block pattern</i>
Max-dn|max-ephone|ip source-address
22. SNR is also known as?
(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
Put them in a route list.
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
Mobile Connect
23. When a device is roaming to a different site How does it get the <i>Roaming Sensitive Settings</i> from the Device Pool?
When multiple sites use the same route pattern - do your digit manipulation on a route list.
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
In CUCM - configure CFUR to point to it's E164 number.
24. 'If you're not getting a DHCP address from CUCM what then?
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
25. What are the commands to configure a T1/E1 PRI?
<b>Access Port Method</b>||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
(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
Allows you to transfer by only pressing the Transfer button once.
Assign SIP Dial Rules
26. Need to Know||Anytime you have an external call - it needs to match on a route pattern.
Configure it on the route group through the route list - then it will be local to the route list.
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
Debug ephone moh
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
27. How much bandwidth does a G.729 call including layer 3 require?
24kb/s
On CUCM it's identical to adding an H323 GW.
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)
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
28. If you have an H323 Gateway at a siet where the phones are controlled by CUCM - what do you need from a dial-peer perspective?
Apply <i>voice-class codec</i>
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.
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
#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
29. How do you maintain 4-digit dialing to phones that are running in an SRST scenario (WAN outage)?
30. What are the commands to ensure an H323 GW always uses the Sub as primary and Pub as a backup?
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.
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
Master/Slave relationship. CUCM controls it.
#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
31. When do you do digit manipulation at the route pattern as opposed to the route list?
1 ... this is not optional!
<i>Auto Call Pickup Enabled</i>
Debug isdn q931
When multiple sites use the same route pattern - do your digit manipulation on a route list.
32. How would you verify that DHCP is working in IOS?
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)#<i>sh cdp neigh detail
Dial-peers
33. What are the commands to create the L3 routing interface for VLANS (SVI)?
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
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 .)
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
34. On an H323 GW - how do you adjust the timers for redundancy hunting?
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
To enable two-stage dialing.
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
Apply it to the ephone.
35. How do you test a Voice Translation Rule?
#test voice translation rule 1 <input to test>
<i>SIP Route Pattern</i> over a SIP Trunk.
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
36. What are the commands to configure SRST in fallback?
Reroute when there is a WAN Outage.
Reroute when here is WAN congestion.
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
37. Name 2 commands to verify RSVP functionality.
Sh ip rsvp reservation||sh sccp connections
(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
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
38. What types of digit manipulation can you perform at the route pattern?
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
Mobile Voice Access
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
39. How would you enable security on a GK?
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
Reroute when there is a WAN Outage.
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
40. What is a best practice for digit manipulation - in regards to H323 GWs?
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
...
41. 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?
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
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
Reroute when here is WAN congestion.
Ccm-manager music-on-hold
42. A <i>show isdn status</i> shows TEI_ASSIGNED. What do you do to get MULTIPLE_FRAME_ESTABLISHED?
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
#test voice translation rule 1 <input to test>
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
When they are explicitly matched in a destination-pattern in a dial-peer.
43. What are the commands to configure a MGCP Gateway? (router)
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
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
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
44. How can you confirm the MGCP GW is registered to CUCM - in IOS?
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 ipv4:10.1.5.3 (sub)| #dtmf-relay h245-alpha||#dial-peer voice 30001 voip| #destination-pattern 3...| #pre
Sh ccm-manager
When configuring TEHO.
45. When setting up SIP URI where do you configure the CUCM's domain name?'
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
Debug isdn q931
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
Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</i>
46. What is the full E164 format?
...
Apply <i>voice-class codec</i>
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
47. What terminology translates to SRST?
Debug ephone moh
Reroute when there is a WAN Outage.
When multiple sites use the same route pattern - do your digit manipulation on a route list.
Mobile Connect
48. 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-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
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.
Allows you to transfer by only pressing the Transfer button once.
49. How do you convert a full PRI 9config'd by ccm-manager config server) back to a Fractional PRI?
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Mobile Connect
50. When a CUCM device dials a number - what happens?
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
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 > 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