In the criteria of the "Outbound Rules", "Calls from Extension(s)" having comma separated
values will allow multiple extension ranges to be defined - ANSTrue
The Virtual Extension of the slave must match the master side virtual extension number -
ANSTrue
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. - ANSFalse
On outbound calls to external numbers, 3CX will process "Outbound Rules" in a "Best Matching"
way, depending on how many of the criteria match. - ANSFalse
Wireshark can - without any additional Plugins Decode all Codecs including G711A, G711U,
GSM, G729, G726 - ANSFalse
STUN extensions can be configured to connect to the tunnel port of 3CX instead of the SIP port
for more security. - ANSFalse
The log files of 3CX are never cleaned even when you restart the 3CX Services - ANSFalse
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. - ANSFalse
Remote extensions may be provisioned using the HTTP or HTTPS URLs - ANSFalse
The default Voicemail PIN of an extension consist of 4 random alphanumerical characters. -
ANSFalse
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. - ANSTrue
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. - ANSTrue
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 - ANSTrue
, 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. -
ANSTrue
When selecting the option "I need a 3CX FQDN" an internal DNS is not mandatory - ANSTrue
The order of "Inbound Rules" is not important when you have DID and CID "Inbound Rules",
CIDs always have higher priority. - ANSFalse
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. - ANSFalse
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? -
ANSTrue
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. - ANSFalse
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). - ANSTrue
SRTP will secure calls so that a middle-man can't see the SIP traffic in plain-text. - ANSFalse
CID and DID "Inbound Rules" can both be configured to route calls differently depending on the
Office Hours. - ANSTrue
"Prepend" will add digits to the end of the dialed number before sending hte call to the
destination defined in the "Route" - ANSFalse
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. - ANSFalse
The "Server Activity Log" will provide information for: Phone Registrations, Gateway and SIP
Trunk Interactions, All Related Calls - ANSTrue
Each extension gets a default password which is always the same and should be changed for
more safety. - ANSFalse
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