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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. How do you use an ephone template?
Max-dn|max-ephone|ip source-address
Apply it to the ephone.
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
8kb/s
2. Two useful troubleshooting commands for CUCME?
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 .)
<i>#after-hours block pattern</i>
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
3. What are the commands to configure SRST in fallback?
'Use before the ... so XXXX
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
<b>Access Port Method</b>||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</i>
4. What are the commands to set up the PRI?
5. What would force you to use telephony-service to configure SRST?
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.
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
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
6. 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
1 ... this is not optional!
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
On CUCM it's identical to adding an H323 GW.
7. What are the commands to create the L3 routing interface for VLANS (SVI)?
Allows you to transfer by only pressing the Transfer button once.
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
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
8. Does CUCM support RSVP natively?
CUCM OS Administration Settings --> NTP Servers
To enable two-stage dialing.
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
9. How do you support multiple codecs on a dialpeer?
Apply <i>voice-class codec</i>
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
10. What is Phone NTP Reference used for?
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
#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
It's best to strip digits at the voice port.
To notify SIP Phones of NTP
11. How do you configure AAR?
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
SERVICE PARAMETER> Automated Alternate Routing Enable > True
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
12. When can you not use a Standard Local Route Group?
<b>Access Port Method</b>||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
<i>Auto Call Pickup Enabled</i>
It's best to strip digits at the voice port.
When configuring TEHO.
13. What should you configure before entering auto qos?
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
Bandwidth
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
14. What does the <i>Transfer On-hook Enabled</i> Service Parameter do?
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
To notify SIP Phones of NTP
Allows you to transfer by only pressing the Transfer button once.
15. What is the <i>Enable Enterprise Feature Access</i> Service Parameter used for?
Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</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
To enable two-stage dialing.
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
16. Where can you assign the AAR Group?
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
<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.
#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
17. 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 --> <i>True</i>
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
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)
'Use before the ... so XXXX
18. How can you confirm the MGCP GW is registered to CUCM - in IOS?
#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
Sh ccm-manager
19. How do you configure CUBE?
20. What CM Service needs to be start in Serviceability for MOH to work?
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
IP Voice Media Streaming App
'Service Parameters --> <i>Auto Answer Timer</i>
21. How do you convert a full PRI 9config'd by ccm-manager config server) back to a Fractional PRI?
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
Dtmf-relay h245-alpha
Bandwidth
22. In a SIP Dial Rule how do you represent a dialed * instead of it being used as a wildcard?
23. How do you configure a SIP Trunk? (router side)
(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
Allows you to transfer by only pressing the Transfer button once.
When configuring TEHO.
(config)#voice register dialplan
24. What are the basic SCCP Commands fro telephony-service in CUCME?
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
Automatically configures the MGCP GW for you.
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
25. When do you use translate called? Translate calling?
To enable two-stage dialing.
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
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
26. What are the commands to configure NTP in IOS?
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
27. How can you tell the difference between Proxy Line Mode and Shared Line Mode by just looking at the phone?
Top Down means that channel 1 will be the first channel used to place outgoing calls.
#test voice translation rule 1 <input to test>
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
#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
28. What CUCM services should you activate?
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
To notify SIP Phones of NTP
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
Bottom up - to prevent glare.
29. An MGCP gateway serving as an SRST router requires what?
Show ccm-manager
Bottom up - to prevent glare.
#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
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
30. 'If you're not getting a DHCP address from CUCM what then?
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
...
Via the <i>voice hunt-group parallel</i> command
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
31. How do you change modes in <i>voice register global</i>?
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
Default is SRST||<i>mode cme</i> changes to CME|<i>no mode cme</i> changes back to SRST
(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
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
32. What is SIP URI?
IP Voice Media Streaming App
Phone number followed by domain name.|i.e. 3006@ipxcme.com
(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
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
33. How do you setup AutoRegistration in CUCM?
'Use before the ... so XXXX
TRUE
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
Service Parameters --> <i>Mobile Voice Access Number</i>
34. Describe how you configure SIP URI functionality.
Reroute when there is a WAN Outage.
24kb/s
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
35. When using GK redundancy - what must you do to ensure both SUB and PUB use port 1720 for call-signalling.
36. In CUCME what SIP config command is equivalent to ephone and ephone-dn?
Dial-peers
Matches any length dialed number and truncates it to 4 digits.
<i>voice register pool</i>|and|<i>voice register dn</i>
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
37. How do you configure an MGCP GW? (router side)
(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
<i>SIP Route Pattern</i> over a SIP Trunk.
Configure it on the route group through the route list - then it will be local to the route list.
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
38. How do you allow the Calling Name to be sent to the PSTN on a router?
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
Debug ephone moh
Device Pool Locations
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
39. How do you set the Call Park Reversion Timer?
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
TRUE
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
Mobile Voice Access
40. If you're using Called/Calling Party Transformations at the Route Pattern - what happens if you do Digit manipulation at the route list?
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
The Called/Calling Transformations are superceded.
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
...
41. How do you allow H323 calls to be preserved should the primary H323 GW fail?
#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
Configure it on the route group through the route list - then it will be local to the route list.
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
42. How do you configure a gateway to register with gatekeeper?
(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
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
#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
...
43. How do you ensure no inter-digit timeout fro SIP phones within CUCM?
<i>voice register pool</i>|and|<i>voice register dn</i>
Mobile Voice Access
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
SIP Dial Rules
44. When a device is roaming to a different site How does it get the <i>Roaming Sensitive Settings</i> from the Device Pool?
Via the <i>voice hunt-group parallel</i> command
Sh ip rsvp reservation||sh sccp connections
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
45. Aside from the <i>Media Resources --> Mobile Voice Access</i> where else do you need to specify the the Mobile Voice Access DN?
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
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
Service Parameters --> <i>Mobile Voice Access Number</i>
The <i>name</i> configuration field in ephone-dn and voice register dn
46. What are two caveats to using the <i>ccm-manager config server</ip> command?
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
1 ... this is not optional!
(config)#<i>sh cdp neigh detail
47. How do you maintain 4-digit dialing to phones that are running in an SRST scenario (WAN outage)?
48. How do you configure CUCM redundancy on an H323 gateway?
<i>#after-hours block pattern</i>
#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
#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
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
49. '<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
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
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
50. What command is needed on a gateway for it to invoke the DSP to convert multicast MOH stream to TDM?
Ccm-manager music-on-hold
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.
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
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