3CX Academy, Advanced Certification Latest Update
Graded A+
In the criteria of the "Outbound Rules", "Calls from Extension(s)" having comma
separated values will allow multiple extension ranges to be defined
True
The Virtual Extension of the slave must match the master side virtual extension
number
True
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.
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.
False
Wireshark can - without any additional Plugins Decode all Codecs including
G711A, G711U, GSM, G729, G726
False
STUN extensions can be configured to connect to the tunnel port of 3CX instead
of the SIP port for more security.
False
The log files of 3CX are never cleaned even when you restart the 3CX Services
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.
False
Remote extensions may be provisioned using the HTTP or HTTPS URLs
False
The default Voicemail PIN of an extension consist of 4 random alphanumerical
characters.
False
Your 3CX has only one SIP Trunk and receives a call from number 8135791691. If
you have a "Inbound CID Reformatting" rule on the Trunk with "Source Pattern"
813(...)(.*) and "Replace Pattern" \1\2, the extension that receives the call will see
"5791691" as the caller ID on its display.
True
When 3CX has been installed without a FQDN from 3CX and in split DNS mode,
the DNS server must not be installed on the same machine as the phone system.
True
Your Local IP Phone loses the registration to 3CX and you want do troubleshoot
the issue. You should start a Wireshark Capture on the 3CX Server, reboot the
phone, and then apply the Filter sip.Cseq.Method==REGISTER in order to see if
registrations are reaching 3CX
True
, Your 3CX has only one SIP Trunk and receives a call from number 422033272020
and you want it to be presented on the extension display as +44272020. You can
do this with a "Inbound CID Reformatting" rule on the Trunk with "Source
Pattern" 44(..)(..)(.*) and "Replace Pattern" +44\3.
True
When selecting the option "I need a 3CX FQDN" an internal DNS is not mandatory
True
The order of "Inbound Rules" is not important when you have DID and CID
"Inbound Rules", CIDs always have higher priority.
False
You have a Master Bridge with 3-digit extensions 1xxx and a Slave Bridge with 4-
digit extensions 2xxx. In the "Outbound Rules" you use to rote calls across the
bridge, using a prefix is mandatory.
False
You have run the 3CX "Firewall Checker" and comes up as Green, but you still
have audio issues and calls dropping on on outbound / inbound calls. Can SIP
ALG be the culprit?
True
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.
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
CID and DID "Inbound Rules" can both be configured to route calls differently
depending on the Office Hours.
True
"Prepend" will add digits to the end of the dialed number before sending hte call
to the destination defined in the "Route"
False
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.
False
The "Server Activity Log" will provide information for: Phone Registrations,
Gateway and SIP Trunk Interactions, All Related Calls
True
Each extension gets a default password which is always the same and should be
changed for more safety.
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
Graded A+
In the criteria of the "Outbound Rules", "Calls from Extension(s)" having comma
separated values will allow multiple extension ranges to be defined
True
The Virtual Extension of the slave must match the master side virtual extension
number
True
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.
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.
False
Wireshark can - without any additional Plugins Decode all Codecs including
G711A, G711U, GSM, G729, G726
False
STUN extensions can be configured to connect to the tunnel port of 3CX instead
of the SIP port for more security.
False
The log files of 3CX are never cleaned even when you restart the 3CX Services
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.
False
Remote extensions may be provisioned using the HTTP or HTTPS URLs
False
The default Voicemail PIN of an extension consist of 4 random alphanumerical
characters.
False
Your 3CX has only one SIP Trunk and receives a call from number 8135791691. If
you have a "Inbound CID Reformatting" rule on the Trunk with "Source Pattern"
813(...)(.*) and "Replace Pattern" \1\2, the extension that receives the call will see
"5791691" as the caller ID on its display.
True
When 3CX has been installed without a FQDN from 3CX and in split DNS mode,
the DNS server must not be installed on the same machine as the phone system.
True
Your Local IP Phone loses the registration to 3CX and you want do troubleshoot
the issue. You should start a Wireshark Capture on the 3CX Server, reboot the
phone, and then apply the Filter sip.Cseq.Method==REGISTER in order to see if
registrations are reaching 3CX
True
, Your 3CX has only one SIP Trunk and receives a call from number 422033272020
and you want it to be presented on the extension display as +44272020. You can
do this with a "Inbound CID Reformatting" rule on the Trunk with "Source
Pattern" 44(..)(..)(.*) and "Replace Pattern" +44\3.
True
When selecting the option "I need a 3CX FQDN" an internal DNS is not mandatory
True
The order of "Inbound Rules" is not important when you have DID and CID
"Inbound Rules", CIDs always have higher priority.
False
You have a Master Bridge with 3-digit extensions 1xxx and a Slave Bridge with 4-
digit extensions 2xxx. In the "Outbound Rules" you use to rote calls across the
bridge, using a prefix is mandatory.
False
You have run the 3CX "Firewall Checker" and comes up as Green, but you still
have audio issues and calls dropping on on outbound / inbound calls. Can SIP
ALG be the culprit?
True
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.
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
CID and DID "Inbound Rules" can both be configured to route calls differently
depending on the Office Hours.
True
"Prepend" will add digits to the end of the dialed number before sending hte call
to the destination defined in the "Route"
False
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.
False
The "Server Activity Log" will provide information for: Phone Registrations,
Gateway and SIP Trunk Interactions, All Related Calls
True
Each extension gets a default password which is always the same and should be
changed for more safety.
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
