3CX Academy Advanced Certification | 62 Questions with 100% Correct Answers | Verified | L, Exams of Computer Science

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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 The Virtual Extension of the slave must match the master side virtual extension number -ANSWER 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. -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 Wireshark can - without any additional Plugins Decode all Codecs including G711A, G711U, GSM, G729, G726 -ANSWER False STUN extensions can be configured to connect to the tunnel port of 3CX instead of the SIP port for more security. -ANSWER False The log files of 3CX are never cleaned even when you restart the 3CX Services - 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 The default Voicemail PIN of an extension consist of 4 random alphanumerical characters. -ANSWER 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. -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

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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. -ANSWER 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. -ANSWER 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? -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 CID and DID "Inbound Rules" can both be configured to route calls differently depending on the Office Hours. -ANSWER True "Prepend" will add digits to the end of the dialed number before sending hte call to the destination defined in the "Route" -ANSWER 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. -ANSWER False The "Server Activity Log" will provide information for: Phone Registrations, Gateway and SIP Trunk Interactions, All Related Calls -ANSWER True Each extension gets a default password which is always the same and should be changed for more safety. -ANSWER False

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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 3CX is 8-- MS while the Max Delta for the reverse direction is 20 Ms. The issue is located at the provider end and not 3CX end. -ANSWER True If you want to check 3CX routing of a call then only a Wireshark capture is needed, - ANSWER False Knowing only the FQDN/port of 3CX and the MAC address of a phone I could guess its provisioning URL. -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

4433272020. 3CX will send the call to the provider exactly as dialed -ANSWER True You can use the same VoIP Provider in all of the Outgoing Routes of a rule -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 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 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 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 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 -ANSWER False

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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 -ANSWER True A TLS certificate and key will have to be imported into 3CX so taht SRTP can be used. - ANSWER False You cannot create multiple CID "Inbound Rules" and associate them with different SIP Trunks you have in 3CX. -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 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 The default Blacklist time interval is of 1800 minutes. -ANSWER True When an "Outbound Rule" has an extension group defined, outgoing calls from the conference extension are able to be made from 3CX -ANSWER 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. -ANSWER False "Inbound CID Reformatting" can be used to change teh Caller ID name of an incoming call based on a set of rules. -ANSWER False The "Server Activity Log" will provide information for: Phone Registrations, Gateway and SIP Trunk Interactions, All Related Calls -ANSWER 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 -ANSWER False When viewing in wireshark a Remote Extension via STUN registering to your 3CX, you should see in the SIP section of the Registration message in the "Contact" field the Local IP address of the remote phone -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

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The default Voicemail PIN of an extension consist of 4 random alphanumerical characters. -ANSWER False 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 You are having an issue with incoming call routing being sent to the wrong internal destination once it is received by 3CX. The first step to troubleshoot the issue would be to set your System to Verbose Logging, start wireshark, replicate the issue and then Restart all 3CX Services, generate the support files and send them to 3CX Support for troubleshooting -ANSWER False Setting a Name when creating a "CID Inbound Rule" is mandatory -ANSWER False When the option "Remote PBX uses SBC/Tunnel Connection" is enabled on the master bridge, the Slabe bridge must be registered behind the SBC. -ANSWER False You can take a phone that has already been provisioned with RPS as remote STUN Extension and configure it as a Local LAN extension, as long as 2 weeks have passed since the phone was provisioned as a STUN extension -ANSWER 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? -ANSWER True Remote presence of a bridged system is available in both IP Phones (via BLF) and the 3CX Phone for Windows -ANSWER False The numbering plan of bridged systems must be different in order for outgoing calls to the bridge to work -ANSWER False ZRTP and SRTP are both supported to secure the audio stream of calls within the 3CX Phone System. -ANSWER False The 3CX Instance Manager is available for 3CX v16+ installs on what Operating System? -ANSWER Linux x86 with no failover or hosting mode enabled The 3CX Instance Manager allows you to perform batch updates -ANSWER True