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3CX Academy, Advanced Certification Exam Graded A+
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3CX Academy, Advanced Certification Exam Graded A+
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3CX Academy, Advanced CertificationExam Graded A+
"Inbound CID Reformatting" can be used to change teh Caller ID name of an incoming
call based on a set of rules. - ANSWER-False
"Prepend" will add digits to the end of the dialed number before sending hte call to the
destination defined in the "Route" - ANSWER-False
A TLS certificate and key will have to be imported into 3CX so taht SRTP can be used. -
ANSWER-False
An extension will be allowed up to 25 attempts (default) for authenticating successfully,
after what it will be blacklisted for the default blacklisting interval of 1800 minutes. -
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 - ANSWER-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. - ANSWER-False
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 - ANSWER-True
CID and DID "Inbound Rules" can both be configured to route calls differently
depending on the Office Hours. - ANSWER-True
Each extension gets a default password which is always the same and should be
changed for more safety. - ANSWER-False
For extensions that are registered over multiple site-to-site VPNs, by default 3CX
delivers the audio between phones if they are on different subnets. - ANSWER-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. - ANSWER-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. - ANSWER-False
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 - ANSWER-False
If you want to check 3CX routing of a call then only a Wireshark capture is needed, -
ANSWER-False
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. - ANSWER-True
If you want to decode audio in wireshark without any plugins, your calls will need to be
using either G711A or G711U - ANSWER-True
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. - ANSWER-True
In the criteria of the "Outbound Rules", "Calls from Extension(s)" having comma
separated values will allow multiple extension ranges to be defined - ANSWER-True
Knowing only the FQDN/port of 3CX and the MAC address of a phone I could guess its
provisioning URL. - ANSWER-False
On outbound calls to external numbers, 3CX will process "Outbound Rules" in a "Best
Matching" way, depending on how many of the criteria match. - ANSWER-False
Once Secure SIP has been configured in 3CX, Secure SIP certificates will need to be
deployed manually in phones and softphones OS so that they can communicate. -
ANSWER-False
Remote extensions may be provisioned using the HTTP or HTTPS URLs - ANSWER-
False
Remote presence of a bridged system is available in both IP Phones (via BLF) and the
3CX Phone for Windows - ANSWER-False
Setting a Name when creating a "CID Inbound Rule" is mandatory - ANSWER-False
SRTP will secure calls so that a middle-man can't see the SIP traffic in plain-text. -
ANSWER-False
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