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3CX ACADEMY ADVANCED CERTIFICATION EXAM ACTUAL EXAM
COMPLETE 150 QUESTIONS WITH DETAILED VERIFIED ANSWERS
ALREADY GRADED A+
The 3CX SIP port should be filtered by firewall ACL rules to
maximize security and allow only trusted IPs to reach it
(VoIP providers or STUN extensions if any). - CORRECT
ANSWER >>>>True
SRTP will secure calls so that a middle-man can't see the SIP
traffic in plain-text. - CORRECT ANSWER >>>>False
CID and DID "Inbound Rules" can both be configured to
route calls differently depending on the Office Hours. -
CORRECT ANSWER >>>>True
"Prepend" will add digits to the end of the dialed number
before sending hte call to the destination defined in the
"Route" - CORRECT ANSWER >>>>False
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Your Local IP Phone loses the Registration to 3CX and you
want to troubleshoot the issue. You should start a
Wireshark Capture on the 3CX Server, reboot the phone,
and then apply the Filter sip.Cseq.Method==SUBSCRIBE in
order to see if registrations are reaching 3CX. - CORRECT
ANSWER >>>>False
The "Server Activity Log" will provide information for: Phone
Registrations, Gateway and SIP Trunk Interactions, All
Related Calls - CORRECT ANSWER >>>>True
Each extension gets a default password which is always the
same and should be changed for more safety. - CORRECT
ANSWER >>>>False
You are debugging an audio issue using wireshark by
analysing the RTP streams. While using the RTP Stream
analysis tools you see that the MAX Delta of packages from
the Provider to 3CX is 8-- MS while the Max Delta for the
reverse direction is 20 Ms. The issue is located at the
provider end and not 3CX end. - CORRECT ANSWER
>>>>True
If you want to check 3CX routing of a call then only a
Wireshark capture is needed, - CORRECT ANSWER >>>>False
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Knowing only the FQDN/port of 3CX and the MAC address of
a phone I could guess its provisioning URL. - CORRECT
ANSWER >>>>False
An outbound call is routed through the VoIP Provider. This
"Route" of the "Outbound Rule" has "Strip Digits" set as 0
and has no values in the "Prepend" field. A user dials
004433272020. 3CX will send the call to the provider exactly
as dialed - CORRECT ANSWER >>>>True
You can use the same VoIP Provider in all of the Outgoing
Routes of a rule - CORRECT ANSWER >>>>True
If you want to decode audio in wireshark without any
plugins, your calls will need to be using either G711A or
G711U - CORRECT ANSWER >>>>True
If you want to check the internal PBX routing of a call, the
most reliable way is through the use of the Binary Log
Viewer, or which you must know some information about
the call you are investigating, like the Caller ID, the time of
the call, etc. - CORRECT ANSWER >>>>True
Assume the following scenario: 3CX is connected via a site-
to-site VPN with two different buildings where remote
phones have been provisioned. 3CX is on network
192.168.9.xxx, Phones on location A are on network
3CX ACADEMY ADVANCED CERTIFICATION EXAM ACTUAL EXAM
COMPLETE 150 QUESTIONS WITH DETAILED VERIFIED ANSWERS
ALREADY GRADED A+
The 3CX SIP port should be filtered by firewall ACL rules to
maximize security and allow only trusted IPs to reach it
(VoIP providers or STUN extensions if any). - CORRECT
ANSWER >>>>True
SRTP will secure calls so that a middle-man can't see the SIP
traffic in plain-text. - CORRECT ANSWER >>>>False
CID and DID "Inbound Rules" can both be configured to
route calls differently depending on the Office Hours. -
CORRECT ANSWER >>>>True
"Prepend" will add digits to the end of the dialed number
before sending hte call to the destination defined in the
"Route" - CORRECT ANSWER >>>>False
,2|Page
Your Local IP Phone loses the Registration to 3CX and you
want to troubleshoot the issue. You should start a
Wireshark Capture on the 3CX Server, reboot the phone,
and then apply the Filter sip.Cseq.Method==SUBSCRIBE in
order to see if registrations are reaching 3CX. - CORRECT
ANSWER >>>>False
The "Server Activity Log" will provide information for: Phone
Registrations, Gateway and SIP Trunk Interactions, All
Related Calls - CORRECT ANSWER >>>>True
Each extension gets a default password which is always the
same and should be changed for more safety. - CORRECT
ANSWER >>>>False
You are debugging an audio issue using wireshark by
analysing the RTP streams. While using the RTP Stream
analysis tools you see that the MAX Delta of packages from
the Provider to 3CX is 8-- MS while the Max Delta for the
reverse direction is 20 Ms. The issue is located at the
provider end and not 3CX end. - CORRECT ANSWER
>>>>True
If you want to check 3CX routing of a call then only a
Wireshark capture is needed, - CORRECT ANSWER >>>>False
, 3|Page
Knowing only the FQDN/port of 3CX and the MAC address of
a phone I could guess its provisioning URL. - CORRECT
ANSWER >>>>False
An outbound call is routed through the VoIP Provider. This
"Route" of the "Outbound Rule" has "Strip Digits" set as 0
and has no values in the "Prepend" field. A user dials
004433272020. 3CX will send the call to the provider exactly
as dialed - CORRECT ANSWER >>>>True
You can use the same VoIP Provider in all of the Outgoing
Routes of a rule - CORRECT ANSWER >>>>True
If you want to decode audio in wireshark without any
plugins, your calls will need to be using either G711A or
G711U - CORRECT ANSWER >>>>True
If you want to check the internal PBX routing of a call, the
most reliable way is through the use of the Binary Log
Viewer, or which you must know some information about
the call you are investigating, like the Caller ID, the time of
the call, etc. - CORRECT ANSWER >>>>True
Assume the following scenario: 3CX is connected via a site-
to-site VPN with two different buildings where remote
phones have been provisioned. 3CX is on network
192.168.9.xxx, Phones on location A are on network
