3CX ADVANCED CERTIFICATION EXAM 2023-2024 ACTUAL
EXAM 100 QUESTIONS AND CORRECT DETAILED
ANSWERS|AGRADE||NEWEST UPDATE
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. - ANSWER- 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. - ANSWER- 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 - ANSWER- 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. - ANSWER- True
When selecting the option "I need a 3CX FQDN" an internal DNS is not
mandatory - ANSWER- True
The order of "Inbound Rules" is not important when you have DID and
CID "Inbound Rules", CIDs always have higher priority. - ANSWERFalse
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. - ANSWERFalse
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? - ANSWER- 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. - ANSWER- 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). - ANSWER- True
SRTP will secure calls so that a middle-man can't see the SIP traffic in
plain-text. - ANSWER- False
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