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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 types of digit manipulation can you perform at the route pattern?
It's best to strip digits at the voice port.
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
Reroute when there is a WAN Outage.
Mobile Voice Access
2. How do you ensure an H323 GW always uses the Sub as primary and Pub as a backup?
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.
...
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
3. What are the basic SCCP Commands fro telephony-service in CUCME?
IP Voice Media Streaming App
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
On CUCM it's identical to adding an H323 GW.
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
4. What are the commands to configure SIP phones in CME?
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
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
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
5. How do you create a trunk on the switch side?
IP Voice Media Streaming App
The <i>name</i> configuration field in ephone-dn and voice register dn
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
#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
6. What command is needed on a gateway for it to invoke the DSP to convert multicast MOH stream to TDM?
CUCM OS Administration Settings --> NTP Servers
Ccm-manager music-on-hold
Via the <i>voice hunt-group parallel</i> command
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
7. What kind of relationship does an MGCP gateway have with CUCM?
Master/Slave relationship. CUCM controls it.
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
Matches any length dialed number and truncates it to 4 digits.
Ccm-manager music-on-hold
8. How do you set up redundancy on outbound dial-peers on an H323 gateway?
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
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
9. How do you enable AAR system wide?
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)#<i>sh cdp neigh detail
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
Bandwidth
10. How do you disable KPML?
Assign SIP Dial Rules
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
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.
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
11. What are the base telephony-service commands for CME?
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
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
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
In CUCM - configure CFUR to point to it's E164 number.
12. How do you setup AutoRegistration in CUCM?
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
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
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
13. Maximum Wait Time for Desk Pickup?
SERVICE PARAMETER> Automated Alternate Routing Enable > True
Reroute when here is WAN congestion.
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
14. How do you set the inter-digit timeout for SIP phones in CME?
(config)#voice register dialplan
Bottom up - to prevent glare.
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
15. How do you place SCCP and SIP phones into a single huntgroup?
Max-dn|max-ephone|ip source-address
(config)#<i>sh cdp neigh detail
Via the <i>voice hunt-group parallel</i> command
(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
16. On an MGCP GW - how could you see the primary and backup CUCM servers?
Show ccm-manager
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
The Called/Calling Transformations are superceded.
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
17. What terminology translates to AAR?
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
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
Reroute when here is WAN congestion.
Call forwarding between voip to voip (when CUBE is in play)
18. Where can you find the command to restore any of the default IP Phone Services if they happen to have been deleted?
The <i>name</i> configuration field in ephone-dn and voice register dn
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
Debug isdn q931
19. What are the commands to manually configure an MGCP gateway?
#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
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.
<i>SIP Route Pattern</i> over a SIP Trunk.
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
20. When can you not use a Standard Local Route Group?
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
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.
When configuring TEHO.
Automatically configures the MGCP GW for you.
21. When a CUCM device dials a number - what happens?
(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
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 .)
#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
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
22. How do you prevent toll fraud on CUCM?
Device Pool Locations
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
When they are explicitly matched in a destination-pattern in a dial-peer.
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. 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
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
24. 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?
Assign a logout-profile to the ephone.
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
Dial-peers
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
25. What does <i>rule 1 /^.*(....$)/ /1/</i> do?
Matches any length dialed number and truncates it to 4 digits.
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
TRUE
Automatically configures the MGCP GW for you.
26. What are the commands to configure a MGCP Gateway? (router)
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)
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
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
27. How do you ensure that G.711 only is used?
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
Sh ip rsvp reservation||sh sccp connections
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
28. What are the commands to create vlans on an ESW?
Apply it to the ephone.
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
Reroute when here is WAN congestion.
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
29. What is the <i>Enable Enterprise Feature Access</i> Service Parameter used for?
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
It's best to strip digits at the voice port.
To enable two-stage dialing.
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
30. What does Display-IE do?
Sh ccm-manager
Sends the Calling Name.
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
31. 'If you want to make changes to any softkeys where do you do it?
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.
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
32. How do you use the # as a string terminator within a SIP Dial Rule?
To enable two-stage dialing.
Precede the # with a > ... so 9011*>#
<i>#after-hours block pattern</i>
33. How do you test a Voice Translation Rule?
TRUE
#test voice translation rule 1 <input to test>
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 .)
#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
34. 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?
#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
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
Single Number Reach (Mobile Connect)
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.
35. 'How do you inform a SIP phone of NTP information?
8kb/s
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
(config)#<i>sh cdp neigh detail
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
36. How do you verify that NTP is working on the CUCM server?
At the CLI: <i>utils ntp status</i>
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
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 .)
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
37. How do you prevent toll fraud on CME?
#test voice translation rule 1 <input to test>
Automatically configures the MGCP GW for you.
Bottom up - to prevent glare.
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 ....
38. 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>
(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
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
39. What is the bit rate for a G.729 call excluding layer 2?
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
8kb/s
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
40. When it comes to H323 gateways - what should you always change - unless explicitly told not to?
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
(config)#voice register dialplan
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
41. How do you configure DHCP in IOS?
...
<i>Auto Call Pickup Enabled</i>
(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
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
42. What's the best way to perform digit manipulation on a route group?
8kb/s
Configure it on the route group through the route list - then it will be local to the route list.
#voice service voip| #no supplementary-service h225-notify cid-update
Precede the # with a > ... so 9011*>#
43. 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>
Max-dn|max-ephone|ip source-address
#test voice translation rule 1 <input to test>
Matches any length dialed number and truncates it to 4 digits.
44. How do you get around relying on DNS for your CUCMs?
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
SIP Dial Rules
24kb/s
45. What are the steps to configure Single Number Reach and Mobile Voice Access?
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
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
When multiple sites use the same route pattern - do your digit manipulation on a route list.
46. Is the + character supported on a VOIP dial-peer?
47. On the RDP What is the Rerouting CSS used for?
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
Single Number Reach (Mobile Connect)
To notify SIP Phones of NTP
24kb/s
48. What CUCM services should you activate?
The Called/Calling Transformations are superceded.
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
Via the <i>voice hunt-group parallel</i> command
Top Down means that channel 1 will be the first channel used to place outgoing calls.
49. What are the commands to configure SRST in fallback?
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
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.
Sends the Calling Name.
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
50. How do you configure class of service (CoS) in CUCM? CME?
<i>voice register pool</i>|and|<i>voice register dn</i>
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
...
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow