3CX Academy, Advanced Certification Study Guide Questions With Solutions

3CX Academy, Advanced Certification Study Guide Questions With Solutions 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 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 . 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 , Phones on location A are on network , Phones on location B are on network . 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 , Phone System B on network . 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 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