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Quizzes > Quizzes for Business > Technology

PBX Systems Knowledge Test Challenge

Assess Your VoIP and Telephony Skills

Difficulty: Moderate
Questions: 20
Learning OutcomesStudy Material
Colorful paper art representing a quiz on PBX Systems Knowledge Test

Welcome to this PBX Systems Knowledge Test where telecom professionals and students can engage with real-world VoIP and telephony scenarios. Whether you're preparing for certification or sharpening your call management skills, this PBX quiz offers in-depth multiple-choice questions tailored to improve your technical expertise. Inspired by Joanna Weib's teaching style, it's easy to customise the format in our editor and adapt questions for any training need. Explore similar assessments like the IT Systems Technician Knowledge Test or dive into the Operating Systems Knowledge Test. Don't forget to browse all our quizzes for more practice and insights.

What is the primary function of a PBX system?
To convert analog voice signals to digital data for storage
To route internal calls and manage external connectivity
To provide web-based conferencing services exclusively
To act as a firewall between LAN and WAN
A PBX system routes calls between internal users and external networks, centralizing telephony management. It does not primarily store voice or act as a firewall, nor does it exclusively handle web conferencing.
Which component in a traditional PBX connects the internal telephone network to the public switched telephone network (PSTN)?
Extension port
SIP Trunk
PSTN trunk
VoIP gateway
A PSTN trunk provides the interface between the PBX and the public telephone network for external calls. Extension ports serve internal handsets, SIP trunks are for VoIP, and gateways translate between protocols.
Which protocol is most commonly used for signaling in VoIP call setups?
RTP
SIP
FTP
HTTP
SIP (Session Initiation Protocol) is the predominant protocol for initiating, managing, and terminating VoIP sessions. RTP handles media transport, while FTP and HTTP are unrelated to call signaling.
Which performance metric indicates perceived voice quality on a scale from 1 to 5?
MOS (Mean Opinion Score)
Jitter
Packet Loss
Latency
The Mean Opinion Score (MOS) reflects user-rated voice quality on a 1 - 5 scale. Jitter, packet loss, and latency are network performance metrics, not direct quality ratings.
What is the default UDP port number used by SIP for signaling?
443
5060
80
23
SIP typically uses UDP port 5060 for unencrypted signaling traffic. Ports 443 and 80 are for HTTPS/HTTP, and 23 is for Telnet.
In a VoIP environment, which component enforces security policies and mediates SIP signaling at the network boundary?
Session Border Controller
VoIP Softphone
PBX Extension
Codec Translator
A Session Border Controller (SBC) enforces security, controls signaling, and manages media at the network edge. Softphones and extensions do not handle boundary security, and translators focus on media format conversions.
Which layer 2 technology is commonly used to separate voice traffic from data traffic in a converged network?
Firewall
MPLS
VLAN
DNS
VLANs (Virtual LANs) logically segment voice and data traffic on the same switch infrastructure. MPLS is a layer 3/4 technology, and firewalls/DNS are not used for layer 2 segmentation.
What does RTP stand for, and what is its primary function in VoIP?
Real-Time Protocol; encrypting voice traffic
Reliable Transmission Protocol; ensuring packet delivery
Real-Time Transport Protocol; carrying media streams
Rapid Transfer Protocol; optimizing bandwidth
RTP stands for Real-Time Transport Protocol and is used for transporting media streams like audio and video. It does not inherently provide encryption or reliability guarantees.
When evaluating PBX performance, what does BHCA measure?
Busy Hour Call Attempts; peak call handling capacity
Bandwidth Hours Consumed Average; data usage
Base Handset Call Activity; handset usage
Broadcast Host Call Allocation; system broadcast capability
BHCA (Busy Hour Call Attempts) quantifies the number of call attempts during the busiest hour, reflecting a PBX's peak handling capacity. Other options are not standard telecom metrics.
In SIP signaling, which message confirms that a call has been accepted by the recipient?
200 OK
INVITE
100 Trying
180 Ringing
The SIP 200 OK response indicates that the call has been accepted by the called party. Other responses acknowledge receipt or indicate ringing.
A user reports one-way audio on VoIP calls. Which is the most likely cause?
Incorrect SIP registrars
Blocked RTP traffic due to NAT/firewall rules
Poor handset battery level
Faulty voicemail system
One-way audio typically indicates that RTP media streams are blocked in one direction by NAT or firewall settings. Registration or battery issues would prevent calls entirely.
Which codec is known for low bandwidth usage while maintaining acceptable voice quality?
G.711
G.729
G.722
PCM
G.729 uses around 8 kbps and provides acceptable audio quality, making it efficient for bandwidth-constrained networks. G.711/PCM use much higher bitrates, and G.722 is high-definition.
Which troubleshooting step should you perform first when diagnosing PBX call connectivity issues?
Upgrade the PBX firmware
Check network connectivity and ping endpoints
Reset all user passwords
Replace all handsets
Verifying basic network connectivity by pinging endpoints is the foundational first step. Firmware upgrades or hardware changes come after confirming the network path is functional.
What dial plan pattern would strip the prefix "9" from a 7-digit number before routing an outside call?
^(\d{7})$ replaced by 9$1
^9(\d{7})$ replaced by $1
^9(\d{7})$ replaced by 9$1
^(\d{7})$ replaced by $1
The regex ^9(\d{7})$ matches numbers starting with 9 followed by 7 digits and replaces them with $1, effectively removing the 9. This prepares the correct number for external routing.
When configuring QoS for VoIP, which DSCP value is typically used for expedited forwarding of voice packets?
0 (Best Effort)
46 (EF)
10 (AF11)
26 (AF31)
DSCP value 46 designates Expedited Forwarding (EF), which is commonly used to prioritize voice packets for low latency and jitter. Other values represent different service classes.
In a standard SIP call setup between two endpoints, which sequence of messages occurs?
INVITE, 180 Ringing, 100 Trying, 200 OK, ACK
INVITE, 100 Trying, 180 Ringing, 200 OK, ACK
INVITE, 200 OK, 100 Trying, 180 Ringing, ACK
INVITE, 180 Ringing, 200 OK, CANCEL, ACK
The correct order is INVITE, 100 Trying, 180 Ringing, 200 OK when answered, and finally ACK to confirm session parameters. Other sequences misplace responses or include invalid messages.
A PBX deployed over a WAN shows MOS scores dropping below acceptable levels during peak usage. Which action will most effectively improve voice quality?
Increase RTP port range
Implement a larger jitter buffer and adjust QoS settings
Disable VLAN tagging
Switch from SIP to H.323 signaling
Adding a jitter buffer smooths out packet arrival variances and, with proper QoS, helps maintain voice quality under congestion. Changing protocols or removing VLANs will not address packet delay variation.
During troubleshooting, you notice that DTMF tones are not being recognized by the receiving end. What is the most likely cause?
Mismatched DTMF mode between RFC2833 and inband signaling
Incorrect RTP port assignment
Failure to register the handset to the PBX
Insufficient MOS scores
DTMF failures often result from mismatched modes (RFC2833 vs inband). Ensuring both ends use the same DTMF signaling method resolves digit transmission issues.
Which protocol should be used to encrypt VoIP media streams to prevent eavesdropping?
TLS (for media)
SRTP
IPSec
SSH
SRTP (Secure Real-Time Transport Protocol) encrypts and authenticates RTP media streams specifically. TLS secures signaling but not media, and IPSec/SSH are not standard for VoIP media encryption.
In a multi-tenant PBX setup requiring strict call segregation, which feature ensures that one tenant's dial plan cannot affect another's?
Codec negotiation
Session Border Controller
Dial plan context or call context isolation
RTP proxy
Dial plan context isolation segments tenants within the PBX, ensuring that each tenant's rules and extension patterns remain independent. Other features handle media or security but not dial plan segregation.
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Learning Outcomes

  1. Identify core PBX system components and their functions
  2. Analyse VoIP call routing and switching scenarios
  3. Evaluate performance metrics for PBX deployments
  4. Demonstrate knowledge of telephony protocols and standards
  5. Apply troubleshooting techniques to common PBX issues
  6. Master configuration best practices for efficient PBX setups

Cheat Sheet

  1. Understand Core PBX Components - Dive into trunk lines, call processing units, and station sets to see how your PBX juggles both internal chatter and external calls like a telecom maestro. Getting to know each component helps you troubleshoot faster and design smarter phone networks. Learn more on Wikipedia
  2. Master VoIP Call Routing - Explore how VoIP systems use SIP and H.323 protocols to shuttle voice packets around the globe at lightning speed - and why they're the secret sauce behind crystal-clear calls. Grasping these routing rules lets you optimize paths and avoid those dreaded "can you hear me now?" moments. Wheelhouse's PBX Interface Guide
  3. Evaluate PBX Performance Metrics - Track call volume, average call duration, and uptime to measure your PBX's health like a seasoned data detective. These KPIs reveal usage trends, potential bottlenecks, and areas ripe for improvement - so you can keep callers smiling. TelecomWorld101 Maintenance Basics
  4. Grasp Telephony Protocols and Standards - Get cozy with ISDN, QSIG, and other essential protocols that ensure different phone systems play nicely together. Understanding these standards helps you bridge legacy gear with modern VoIP setups without skipping a beat. Read about Telephony Standards on Wikipedia
  5. Implement Effective Troubleshooting Techniques - Arm yourself with strategies to squash jitter, latency, and packet loss one step at a time. From network sniffing to call-quality reports, you'll learn to isolate issues faster than you can say "hold, please." Advanced Troubleshooting Tips
  6. Apply Configuration Best Practices - Secure your PBX fortress by enabling firewalls, enforcing strong passwords, and keeping software up to date. A locked-down system isn't just safe - it's also leaner, faster, and more reliable. Configuration Best Practices
  7. Monitor Call Quality Factors - Jitter, latency, and packet loss can turn a smooth conversation into a digital echo chamber. Learn to measure these metrics and apply fixes like QoS settings to keep voices crisp and conversations flowing. Nextiva's Call Quality Guide
  8. Conduct Regular System Maintenance - Treat your PBX to routine software updates, backups, and hardware health checks to avoid nasty surprises. Consistent upkeep means fewer outages, happier users, and a system that's always ready for action. Maintenance Essentials
  9. Optimize Network Bandwidth for VoIP - Allocate at least 100 kbps per call and set up QoS policies so voice traffic always gets VIP treatment. Proper bandwidth planning prevents call drops and keeps your team sounding like pros. VoIP Network Best Practices
  10. Stay Informed on PBX Security Measures - Keep an eye on the ever-evolving threat landscape by enabling encryption, monitoring logs, and patching vulnerabilities pronto. A proactive security stance shields your telephony systems from hackers and eavesdroppers. Security & Configuration Tips
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