3CX ACADEMY CERTIFICATION EXAM (ACTUAL 2025/2026)
QUESTIONS WITH verified DETAILED /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). - <<<Answers....True
SRTP will secure calls so that a middle-man can't see the SIP
traffic in plain-text. - <<<Answers....False
If a caller enters the PIN of the voicemail of an extension
incorrectly 3 times, the specific voice mail account gets
blocked for 2 minutes. - <<<Answers....False
The default Blacklist time interval is of 1800 minutes. -
<<<Answers....True
When an "Outbound Rule" has an extension group defined,
outgoing calls from the conference extension are able to be
made from 3CX - <<<Answers....False
You have a Mater Bridge with 4-digit extensions 1xxx and a
Slave Bridge with 4-digit extensions 2xxx. In the "Outbound
Rules" you use to route calls across the bridge, using a prefix
is mandatory. - <<<Answers....False
, "Inbound CID Reformatting" can be used to change teh Caller
ID name of an incoming call based on a set of rules. -
<<<Answers....False
The "Server Activity Log" will provide information for: Phone
Registrations, Gateway and SIP Trunk Interactions, All Related
Calls - <<<Answers....True
If you are having audio issues with calls between internal local
extensions and your firewall checker fails the first thing that
you should do is to make sure that your firewall checker test
passes - <<<Answers....False
When viewing in wireshark a Remote Extension via STUN
registering to your 3CX, you should see in the SIP section of
the Registration message in the "Contact" field the Local IP
address of the remote phone - <<<Answers....False
If a site-to-site VPN is already in place between the PBX
location and a remote location, phones should be provisioned
with Direct/STUN provisioning method. - <<<Answers....False
The default Voicemail PIN of an extension consist of 4 random
alphanumerical characters. - <<<Answers....False
In order to check which side (Caller or Calee) terminated the
call first, you can check using wireshark to see who sent the
first BYE message. - <<<Answers....True
QUESTIONS WITH verified DETAILED /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). - <<<Answers....True
SRTP will secure calls so that a middle-man can't see the SIP
traffic in plain-text. - <<<Answers....False
If a caller enters the PIN of the voicemail of an extension
incorrectly 3 times, the specific voice mail account gets
blocked for 2 minutes. - <<<Answers....False
The default Blacklist time interval is of 1800 minutes. -
<<<Answers....True
When an "Outbound Rule" has an extension group defined,
outgoing calls from the conference extension are able to be
made from 3CX - <<<Answers....False
You have a Mater Bridge with 4-digit extensions 1xxx and a
Slave Bridge with 4-digit extensions 2xxx. In the "Outbound
Rules" you use to route calls across the bridge, using a prefix
is mandatory. - <<<Answers....False
, "Inbound CID Reformatting" can be used to change teh Caller
ID name of an incoming call based on a set of rules. -
<<<Answers....False
The "Server Activity Log" will provide information for: Phone
Registrations, Gateway and SIP Trunk Interactions, All Related
Calls - <<<Answers....True
If you are having audio issues with calls between internal local
extensions and your firewall checker fails the first thing that
you should do is to make sure that your firewall checker test
passes - <<<Answers....False
When viewing in wireshark a Remote Extension via STUN
registering to your 3CX, you should see in the SIP section of
the Registration message in the "Contact" field the Local IP
address of the remote phone - <<<Answers....False
If a site-to-site VPN is already in place between the PBX
location and a remote location, phones should be provisioned
with Direct/STUN provisioning method. - <<<Answers....False
The default Voicemail PIN of an extension consist of 4 random
alphanumerical characters. - <<<Answers....False
In order to check which side (Caller or Calee) terminated the
call first, you can check using wireshark to see who sent the
first BYE message. - <<<Answers....True
