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CCIE Voice Test
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Study First
Subjects
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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 CUCM services should you activate?
Via the <i>voice hunt-group parallel</i> command
'Service Parameters --> <i>Auto Answer Timer</i>
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
2. In CME Where is the Calling Name derived from?
When they are explicitly matched in a destination-pattern in a dial-peer.
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
The <i>name</i> configuration field in ephone-dn and voice register dn
3. How do you prevent toll fraud on CUCM?
(config)#<i>sh cdp neigh detail
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
4. What are two caveats to using the <i>ccm-manager config server</ip> command?
Sh ip rsvp reservation||sh sccp connections
#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
'Use before the ... so XXXX
5. How do you set up redundancy on outbound dial-peers on an H323 gateway?
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
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.
Put them in a route list.
In CUCM - configure CFUR to point to it's E164 number.
6. How do you enable Extension Mobility for a device in CME?
Assign a logout-profile to the ephone.
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
To enable two-stage dialing.
CUCM OS Administration Settings --> NTP Servers
7. What are the commands to set up the PRI?
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8. How do you ensure no inter-digit timeout fro SIP phones within CUCM?
Debug ephone moh
SIP Dial Rules
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
9. What are the commands to configure SIP phones in CME?
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
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 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
Single Number Reach (Mobile Connect)
10. What should you configure before entering auto qos?
Bandwidth
Default is SRST||<i>mode cme</i> changes to CME|<i>no mode cme</i> changes back to SRST
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
#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
11. How do you configure a gateway to register with gatekeeper?
Dtmf-relay h245-alpha
#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:(SUB IP)| #dtmf-relay h245-alpha||#dial-peer voice 30001 voip| #destination-pattern 3...| #voice-clas
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
12. Privacy is enabled system-wide in CUCM by default. (T or F)
(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
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)
TRUE
When they are explicitly matched in a destination-pattern in a dial-peer.
13. How do you create a trunk on the switch side?
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
Assign a logout-profile to the ephone.
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.
Mobile Voice Access
14. What are the commands to configure a MGCP Gateway? (router)
In CUCM - configure CFUR to point to it's E164 number.
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
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
15. What does <i>ccm-manager config server [IP]</ip> do?
Automatically configures the MGCP GW for you.
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
16. What terminology translates to SRST?
(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
Max-dn|max-ephone|ip source-address
Reroute when there is a WAN Outage.
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
17. What CM Service needs to be start in Serviceability for MOH to work?
Sh ip rsvp reservation||sh sccp connections
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 ....
IP Voice Media Streaming App
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
18. What is a best practice for digit manipulation - in regards to H323 GWs?
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
Call Simulator. You can use to validate path from router to the PSTN.
19. In Gatekeeper CAC how do you restrict a specific endpoint?
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
Allows you to transfer by only pressing the Transfer button once.
The <i>name</i> configuration field in ephone-dn and voice register dn
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
20. What are the basic ephone-dn and ephone commands?
(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
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
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
Via the <i>voice hunt-group parallel</i> command
21. What are the commands to configure SRST in fallback?
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
The <i>name</i> configuration field in ephone-dn and voice register dn
(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
(config)#voice register dialplan
22. How is NTP sych setup in CUCM?
CUCM OS Administration Settings --> NTP Servers
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Max-dn|max-ephone|ip source-address
Bandwidth
23. When using <i>drop-through-option</i> What is the max number of huntgroups?
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)#voice register dialplan
1 ... this is not optional!
24. 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)
#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
#test voice translation rule 1 <input to test>
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
25. What's the difference between AAR and SRST?
CUCM OS Administration Settings --> NTP Servers
...
Call Simulator. You can use to validate path from router to the PSTN.
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
26. If you're using Called/Calling Party Transformations at the Route Pattern - what happens if you do Digit manipulation at the route list?
Service Parameters --> <i>Mobile Voice Access Number</i>
The Called/Calling Transformations are superceded.
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
27. How do you verify where MOH is being served up from?
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28. Where can you assign the AAR Group?
<b>Access Port Method</b>||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
Put them in a route list.
29. How do you ensure that G.711 only is used?
Ccm-manager music-on-hold
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
30. How do you place SCCP and SIP phones into a single huntgroup?
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
Via the <i>voice hunt-group parallel</i> command
Outbound dial-peers
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
31. What are the commands to create vlans on an ESW?
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Dial-peers
32. How do you configure phone ports on an ESW?
(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
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
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
At the CLI: <i>utils ntp status</i>
33. Maximum Wait Time for Desk Pickup?
'Service Parameters --> <i>Auto Answer Timer</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
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
1 ... this is not optional!
34. What must you do for a BACD script to work on a CME router?
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
Ccm-manager music-on-hold
Bottom up - to prevent glare.
35. When you make a call and it results in a secondary dial-tone - What is this a symptom of?
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
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
36. How do you prevent toll fraud on CME?
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
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 ....
SERVICE PARAMETER> Automated Alternate Routing Enable > True
IP Voice Media Streaming App
37. On the RDP What is the Rerouting CSS used for?
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
Ccm-manager music-on-hold
SIP Dial Rules
Single Number Reach (Mobile Connect)
38. How do you disable KPML?
In CUCM - configure CFUR to point to it's E164 number.
Assign SIP Dial Rules
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
#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
39. What are the steps to configure Single Number Reach and Mobile Voice Access?
#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
Show ccm-manager
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.
40. On an H323 GW - how do you adjust the timers for redundancy hunting?
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
#voice service voip| #no supplementary-service h225-notify cid-update
41. How do you allow H323 calls to be preserved should the primary H323 GW fail?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
To enable two-stage dialing.
Dtmf-relay h245-alpha
Call Simulator. You can use to validate path from router to the PSTN.
42. Calls coming from CUCM to PSTN need what?
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
<i>Auto Call Pickup Enabled</i>
Outbound dial-peers
43. What's the best way to do digit manipulation on an IOS gateway?
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
44. How do you get around relying on DNS for your CUCMs?
Debug isdn q931
...
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
Call Simulator. You can use to validate path from router to the PSTN.
45. How would you enable security on a GK?
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
Phone number followed by domain name.|i.e. 3006@ipxcme.com
46. How do you configure phone ports on a 3750?
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
<b>Access Port Method</b>||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
Allows you to transfer by only pressing the Transfer button once.
47. <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 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
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
#voice service voip| #no supplementary-service h225-notify cid-update
...
48. 'How do you inform a SIP phone of NTP information?
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)
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
49. How do you configure an MGCP GW? (router side)
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
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
(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
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.
50. Where is it best to manipulate digits for inbound calls?
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