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Test your basic knowledge |
CCIE Voice Test
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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 set the Call Park Reversion Timer?
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
2. Name 2 commands to verify RSVP functionality.
Dtmf-relay h245-alpha
Reroute when there is a WAN Outage.
Sh ip rsvp reservation||sh sccp connections
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
3. On the RDP What is the Rerouting CSS used for?
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Mobile Voice Access
Single Number Reach (Mobile Connect)
4. <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
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
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
...
5. How do you configure class of service (CoS) in CUCM? CME?
'Service Parameters --> <i>Auto Answer Timer</i>
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
Reroute when here is WAN congestion.
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
6. When it comes to H323 gateways - what should you always change - unless explicitly told not to?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
#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>
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
7. What are the commands to configure SIP phones in CME?
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
#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 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
#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
8. When can you not use a Standard Local Route Group?
At the CLI: <i>utils ntp status</i>
When configuring TEHO.
Mobile Connect
24kb/s
9. How do you create a trunk on the router side?
1 ... this is not optional!
When configuring TEHO.
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
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
10. How do you setup AutoRegistration 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
Sh ip rsvp reservation||sh sccp connections
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.
Debug ephone moh
11. How do you maintain 4-digit dialing to phones that are running in an SRST scenario (WAN outage)?
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12. On an MGCP GW - if you select <i>Top Down</i> under Channel Selection Order - What does that mean?
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
Top Down means that channel 1 will be the first channel used to place outgoing calls.
(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 > 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
13. What is the bit rate for a G.729 call excluding layer 2?
Call Simulator. You can use to validate path from router to the PSTN.
To notify SIP Phones of NTP
8kb/s
#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
14. How do you enable AAR?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
<b>Access Port Method</b>||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
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
SERVICE PARAMETER> Automated Alternate Routing Enable > True
15. How do you set up redundancy on outbound dial-peers on an H323 gateway?
SERVICE PARAMETER> Automated Alternate Routing Enable > True
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
Max-dn|max-ephone|ip source-address
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
16. How do you see multicast packets being sent?
Apply <i>voice-class codec</i>
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
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
Debug ephone moh
17. How do you use the # as a string terminator within a SIP Dial Rule?
Precede the # with a > ... so 9011*>#
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
18. In Gatekeeper CAC how do you restrict a specific endpoint?
#test voice translation rule 1 <input to test>
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
19. What should you configure before entering auto qos?
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
'Service Parameters --> <i>Auto Answer Timer</i>
Bandwidth
Device Pool Locations
20. Describe how you configure SIP URI functionality.
Single Number Reach (Mobile Connect)
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
...
21. How do you prevent toll fraud on CME?
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</i>
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
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 ....
22. What are the steps to integrate CUPS with CUCM?
...
Service Parameters --> <i>Mobile Voice Access Number</i>
#voice service voip| #no supplementary-service h225-notify cid-update
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
23. When using <i>drop-through-option</i> What is the max number of huntgroups?
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
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
1 ... this is not optional!
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
24. What kind of relationship does an MGCP gateway have with CUCM?
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
Phone number followed by domain name.|i.e. 3006@ipxcme.com
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
Master/Slave relationship. CUCM controls it.
25. What commands are needed to configure the voice register pool in CME?
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
It's best to strip digits at the voice port.
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
The Called/Calling Transformations are superceded.
26. How do you place SCCP and SIP phones into a single huntgroup?
Via the <i>voice hunt-group parallel</i> command
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
27. Aside from the <i>Media Resources --> Mobile Voice Access</i> where else do you need to specify the the Mobile Voice Access DN?
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
Service Parameters --> <i>Mobile Voice Access Number</i>
Call Simulator. You can use to validate path from router to the PSTN.
Ccm-manager music-on-hold
28. What are the commands to configure an H323 GW?
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
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 ....
The <i>name</i> configuration field in ephone-dn and voice register dn
#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
29. What is the full E164 format?
<b>Access Port Method</b>||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
Matches any length dialed number and truncates it to 4 digits.
Allows you to transfer by only pressing the Transfer button once.
30. 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
...
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
(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
31. How do you configure SRST?
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)
Sh ccm-manager
#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-manager-fallback</i> or <i>telephony-service mode SRST</i>
32. With the gw-priority command - does higher or lower priority take precedence?
Higher
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
Configure it on the route group through the route list - then it will be local to the route list.
33. How do you verify where MOH is being served up from?
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34. What terminology translates to AAR?
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
#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
Reroute when here is WAN congestion.
35. What are the commands to configure a SIP phone in CUCME?
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.
Apply <i>voice-class codec</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
Dial-peers
36. How do you prioritize route groups?
Put them in a route list.
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
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
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
37. What does the <i>Transfer On-hook Enabled</i> Service Parameter do?
Allows you to transfer by only pressing the Transfer button once.
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
Service Parameters --> <i>Mobile Voice Access Number</i>
38. What types of digit manipulation can you perform at the route pattern?
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
Assign SIP Dial Rules
39. How do you enable AAR system wide?
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
(config)#<i>sh cdp neigh detail
Allows you to transfer by only pressing the Transfer button once.
<i>#after-hours block pattern</i>
40. '<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
When they are explicitly matched in a destination-pattern in a dial-peer.
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
...
Debug ephone moh
41. What's the best way to do digit manipulation on an IOS gateway?
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
On CUCM it's identical to adding an H323 GW.
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
42. How do you block an external call from being transferred back out to the pstn by an internal user?
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
Debug isdn q931
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
43. Is the + character supported on a VOIP dial-peer?
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44. When you make a call and it results in a secondary dial-tone - What is this a symptom of?
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
Phone number followed by domain name.|i.e. 3006@ipxcme.com
Device Pool Locations
(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
45. What command is needed on a gateway for it to invoke the DSP to convert multicast MOH stream to TDM?
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
The <i>name</i> configuration field in ephone-dn and voice register dn
Ccm-manager music-on-hold
Sh ccm-manager
46. How do you create a trunk on the switch side?
TRUE
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
(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
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
47. Need to Know||Anytime you have an external call - it needs to match on a route pattern.
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
Call Simulator. You can use to validate path from router to the PSTN.
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 ....
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
48. On a gatekeeper trunk - how do you ensure the Sub is used as primary and Pub as backup?
(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
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
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.
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 ....
49. What are the steps to configure Single Number Reach and Mobile Voice Access?
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</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
#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
50. How can you tell the difference between Proxy Line Mode and Shared Line Mode by just looking at the phone?
Max-dn|max-ephone|ip source-address
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
To enable two-stage dialing.
Can you answer 50 questions in 15 minutes?
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