3CX ACADEMY ADVANCED CERTIFICATION
UPDATED CORE MAIN EXAMINATION QUESTIONS
AND ANSWERS SURE A+
✔✔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. - ✔✔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 192.168.3.xxx, Phones on location
B are on network 192.168.8.xxx. When a call is established, between an extension on
remote location A to an extension to remote location B, by default 3CX will proxy the
SIP and audio traffic and send it to the extensions - ✔✔True
✔✔Assume the following scenario: 2 Phone Systems bridged over a site-to-site VPN
connection. Phone system A on network 192.168.9.xxx, Phone System B on network
192.168.3.xxx. The options "Supports Re-Invite" and "Supports Re-places" are enabled
on the bridge settings. When a call is established between an extension from PBX A to
an extension behind PBX B, the audio will be exchanged directly between the 2
extensions. - ✔✔False
, ✔✔When an established call from your local extension to an external number is
terminated from your IP Phone you should see a BYE being send in from the Provider /
PSTN gateway, towards your PBX IP address in wireshark - ✔✔False
✔✔When 3CX has been installed without an FQDN from 3CX and in split DNS mode,
the DNS server must not be installed on the same machine as the phone system -
✔✔True
✔✔A TLS certificate and key will have to be imported into 3CX so taht SRTP can be
used. - ✔✔False
✔✔You cannot create multiple CID "Inbound Rules" and associate them with different
SIP Trunks you have in 3CX. - ✔✔False
✔✔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). - ✔✔True
✔✔SRTP will secure calls so that a middle-man can't see the SIP traffic in plain-text. -
✔✔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. - ✔✔False
✔✔The default Blacklist time interval is of 1800 minutes. - ✔✔True
✔✔When an "Outbound Rule" has an extension group defined, outgoing calls from the
conference extension are able to be made from 3CX - ✔✔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. - ✔✔False
✔✔"Inbound CID Reformatting" can be used to change teh Caller ID name of an
incoming call based on a set of rules. - ✔✔False
✔✔The "Server Activity Log" will provide information for: Phone Registrations, Gateway
and SIP Trunk Interactions, All Related Calls - ✔✔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 - ✔✔False
UPDATED CORE MAIN EXAMINATION QUESTIONS
AND ANSWERS SURE A+
✔✔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. - ✔✔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 192.168.3.xxx, Phones on location
B are on network 192.168.8.xxx. When a call is established, between an extension on
remote location A to an extension to remote location B, by default 3CX will proxy the
SIP and audio traffic and send it to the extensions - ✔✔True
✔✔Assume the following scenario: 2 Phone Systems bridged over a site-to-site VPN
connection. Phone system A on network 192.168.9.xxx, Phone System B on network
192.168.3.xxx. The options "Supports Re-Invite" and "Supports Re-places" are enabled
on the bridge settings. When a call is established between an extension from PBX A to
an extension behind PBX B, the audio will be exchanged directly between the 2
extensions. - ✔✔False
, ✔✔When an established call from your local extension to an external number is
terminated from your IP Phone you should see a BYE being send in from the Provider /
PSTN gateway, towards your PBX IP address in wireshark - ✔✔False
✔✔When 3CX has been installed without an FQDN from 3CX and in split DNS mode,
the DNS server must not be installed on the same machine as the phone system -
✔✔True
✔✔A TLS certificate and key will have to be imported into 3CX so taht SRTP can be
used. - ✔✔False
✔✔You cannot create multiple CID "Inbound Rules" and associate them with different
SIP Trunks you have in 3CX. - ✔✔False
✔✔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). - ✔✔True
✔✔SRTP will secure calls so that a middle-man can't see the SIP traffic in plain-text. -
✔✔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. - ✔✔False
✔✔The default Blacklist time interval is of 1800 minutes. - ✔✔True
✔✔When an "Outbound Rule" has an extension group defined, outgoing calls from the
conference extension are able to be made from 3CX - ✔✔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. - ✔✔False
✔✔"Inbound CID Reformatting" can be used to change teh Caller ID name of an
incoming call based on a set of rules. - ✔✔False
✔✔The "Server Activity Log" will provide information for: Phone Registrations, Gateway
and SIP Trunk Interactions, All Related Calls - ✔✔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 - ✔✔False
