Active Calls
0
Channels currently in speech
PBX Telephony Mode
EXTENSION
2 Virtual Port(s) Configured
TTS Voice Synthesis
CARTESIA
Sub-100ms Cloud Engine
System Resources
-- MB
Rocky Linux 10 Host
Active Inbound Telephony Sessions Real-time PBX Call Session Table
Call ID Caller Number DID / Target Duration State Routed Extension
No active calls. Waiting for inbound PBX traffic...
Real-time Audio & SIP Signaling Ladder Trace
00:00:00 [SYSTEM] Appliance online (HTTPS & WSS). PBX telephony and AI pipelines ready.
PBX Connection Topology & Registration Virtual Extension Mode
Virtual Extension Registration Details
Virtual SIP Ports & Extension Registrations

Configure individual credentials for each virtual line appearance. Each line registers separately to the PBX with its own extension and password.

Port # Extension Number SIP Auth Username SIP Secret / Password Registration Status
Firewall Whitelist & Media DSP Transcoding Pools

The appliance firewall drops all SIP and RTP packets from unauthorized IPs. If your PBX uses dedicated DSP cards or separate media gateways for audio/transcoding, specify their IPs or subnets below.

These IPs will be automatically whitelisted in the Rocky Linux firewall for RTP audio ports (UDP 10000–20000).
3. SMTP Voicemail-to-Email Delivery

Configure outbound mail settings to deliver caller voicemail recordings (.wav) and Faster-Whisper AI transcripts directly to staff email inboxes when they are busy or decline calls.

Voice AI Pipeline & Architecture Selection FREE LOCAL LAB MODE ($0.00 EXPENSE)
Unified Voice AI Suite Presets
Select between 100% Free On-Premise models or an All-in-One Cloud Suite with a single subscription / single API key covering STT, LLM, and TTS.
1. Speech-to-Text (STT) Engine
2. Conversational LLM Engine
3. Text-to-Speech (TTS) Voice Engine
4. Initial Spoken Greeting & Call Recording Disclosure

This greeting is spoken immediately when an inbound call connects. It is pre-cached in memory for instantaneous (0ms) response.

Listen to the greeting audio with pronunciation rules applied in your browser.
5. Receptionist Persona & Instructions
6. Custom Pronunciation Lexicon & Phonics Rules

Define phonetic replacements for brand names, technical acronyms, or specialized surnames. The VR substitutes matching written words before synthesizing speech.

Phonetic Formatting Guidelines
  • Syllable Sound-Alikes: Spell words with familiar English syllables (e.g. Apex → Ay-pex, Polycom → Polly-com).
  • Letter-by-Letter Acronyms: Capitalize and hyphenate each letter (e.g. SLA → S-L-A, PBX → P-B-X, IT → I-T).
  • Syllable Separation: Use hyphens to keep the neural voice from blending adjacent vowels (e.g. Coop → Co-op).
Written Word / Term Spoken Phonics Test Audio Action
Rules are applied dynamically across greetings, FAQ responses, and consultative announcements.
Directory & Department Routing Table
Used by the AI for intent, phonetic name & skill resolution
Filter by Branch Site:
Type / Source Name Extension Transfer Mode Email / Role Spoken Aliases Routing Skills / Keywords Actions
Spoken Intent & Skills Resolution Simulator

Test how caller utterances (phonetic names, nicknames, and skill requests) resolve to extensions in real time.

Click "Simulate Spoken Query" to test matching algorithm...
Branch Sites & Locations 0 Sites

Configure multiple physical sites, clinics, or branch offices within this company. Each site can have dedicated virtual ports, incoming DIDs, local reception extensions, and personalized spoken greetings.

Site ID Site Name Virtual Ports / DIDs Reception Ext Ext Range Custom Greeting Override Actions
Loading sites...
Appliance Login Accounts

Manage administrator and operator credentials for accessing the AI Virtual Receptionist Appliance.

Username Role Company Scope 2FA Security Created Date Actions
Loading user accounts...
Voice AI Knowledge Base (FAQ Engine) 0 FAQs

Directly answer routine caller questions (return policies, trading hours, location, payment methods) with zero perceived latency. The AI speaks these pre-approved answers instantly without waiting for Ollama reasoning or handing over to a human receptionist.

Interactive Knowledge Base Tester
Category Question / Triggers Spoken Answer & Follow-up Action Actions
Loading knowledge base...
Appliance Documentation & Operator User Guide
Comprehensive architectural overview, configuration instructions, and operational reference organized by module.
Wirepulse VR v1.4.9
QUICK JUMP: 1. Live Call Monitor 2. PBX & Virtual Ports 3. Voice AI & Knowledge Base 4. Directory & Skills Routing 5. System Administration
1. Live Call Monitor & Operational Diagnostics

The Live Call Monitor provides real-time visibility into active telephone calls, speech recognition decoding, conversational reasoning, and SIP signaling handoffs.

Active Calls Table
  • Call ID: Unique SIP Call-ID handle tracking the call session across PBX dialogs.
  • Caller Number: Inbound caller ID (internal PBX extension or external E.164 number).
  • Port: Indicates which virtual extension accepted the call (e.g. 500-io).
  • Target / Routed: Real-time destination resolved by the AI intent engine.
  • Duration: Elapsed connection timer streaming live via WebSocket.
Real-Time Event Trace Stream
  • [SIP]: Telephony signaling events (INVITE, 200 OK, SDP negotiation, BYE).
  • [STT]: Caller speech transcription with millisecond latency.
  • [LLM]: AI dialogue reasoning, entity extraction, and sentiment.
  • [TRANSFER]: Consultative handoff, consent inquiry, and bridge status.
  • [KNOWLEDGE]: FAQ keyword matches and sub-50ms instant response delivery.
Operator Tip: The Trace Log automatically scrolls to display the newest events as they happen in real time. Use the Clear Trace button in the top-right corner of the trace panel to flush the log buffer between testing scenarios or diagnostic calls.
2. PBX Integration, Virtual Ports & Firewall Whitelist

The appliance connects to your PBX (e.g. Mitel MiVoice 400, Cisco CUCM, Asterisk/FreePBX) via SIP over standard UDP/TCP/TLS.

Virtual Extension & Line Port Architecture

The appliance functions as an intelligent SIP client registering virtual extension appearances directly to your PBX:

Single-Port Operation

Standard deployments operate efficiently using a single registered virtual extension (e.g. Ext 500). In single-port mode, all inbound calls, hold music playback, interactive voice responses, and transfers are handled reliably on that registered line appearance.

Multi-Port Scalability

You can scale from 1 up to 16 virtual line ports using the Virtual Ports Count dropdown. Additional virtual ports enable concurrent call handling, dedicated hunt group appearances, or multi-line consultation according to your PBX capacity.

PBX SIP Trunk Mode: Alternatively, you can select Mode B (SIP Trunk) for IP-to-IP peering without extension registrations, using DID pattern matching and CID headers.
Media DSP & Firewall Synchronization

Whenever telephony settings are saved, the appliance automatically updates kernel firewall-cmd rich rules to whitelist:

  • SIP Signaling: PBX Host IP on Port 5060/UDP.
  • Media RTP Streams: PBX media processors and DSP subnets (e.g. 192.168.2.0/24) on Ports 10000-20000/UDP.
Note: Always list your PBX Media DSP subnet in "Media Processor (DSP) Subnet Whitelist" to prevent one-way audio!
Voicemail Dispatch & Transcription

When a call is unanswered, busy, or declined, the automated voicemail pipeline engages:

  • The VR emits an audible 1000Hz tone and records the caller's message.
  • The recorded audio is automatically transcribed to text using the configured speech recognition engine (local models or cloud STT APIs).
  • An email is dispatched via SMTP containing the full text transcript in the message body and the original audio attached as a .wav file.
3. Voice AI Pipeline, Pronunciation Lexicon & Knowledge Base

The Voice AI pipeline features a universal, modular architecture engineered to support both privacy-first local models and high-capacity cloud APIs.

The Universal 3-Stage Voice AI Architecture
Stage 1: Speech-to-Text (STT)
Local Models or Cloud Speech

Transcribes caller speech into text with low latency. Supports local privacy models (Faster-Whisper with int8 CPU quantization) as well as cloud speech APIs (OpenAI Whisper, Deepgram, Google Cloud Speech).

Stage 2: Intelligence & Dialogue (LLM)
Local Ollama or Enterprise Cloud LLMs

Interprets caller intent, resolves staff names and skills, and reasons conversationally. Supports local private models (Llama 3.2, Mistral, Qwen) or enterprise cloud LLMs (OpenAI GPT-4o, Anthropic Claude, Google Gemini).

Stage 3: Speech Synthesis (TTS)
Neural Voices & Local Fallback

Synthesizes natural, human-like voice responses with Australian English neural models (Edge-TTS, ElevenLabs, OpenAI TTS, Amazon Polly) and automatic local Piper fallback.

Note: The appliance ships pre-configured in local lab mode ($0.00 expense), but seamlessly scales to cloud enterprise tiers without changing telephony architecture.
Initial Spoken Greeting & In-Browser Audio Preview

The greeting text in Section 4 is spoken immediately when an inbound call connects.

  • Zero-Latency Pre-Caching: On appliance startup and configuration save, the greeting is pre-synthesized into linear PCM8 memory cache, providing an instantaneous (0ms) response on call answer.
  • Play Greeting Audio: Click the blue Play Greeting Audio button in Section 4 to listen to how the VR speaks the greeting directly in your browser. The button toggles to a red Stop Playback control during playback.
Custom Pronunciation Lexicon: All 9 Acoustic Rules

Neural voice models cannot be fine-tuned directly on custom vocabularies. Use the Custom Pronunciation Lexicon table in Section 6 to define phonetic replacements before speech is synthesized. Here are the 9 core acoustic rules:

Rule Type How It Works Example Mapping
1. Syllable Sound-Alikes Spell unfamiliar words with common English syllables that rhyme with words the model knows. Apex → Ay-pex
Polycom → Polly-com
2. Letter-by-Letter Acronyms Capitalize and hyphenate each letter to force individual letter spelling. SLA → S-L-A
PBX → P-B-X
IT → I-T
3. Syllable Separation Insert hyphens between syllables to prevent the engine from merging adjacent vowels. Coop → Co-op
Subnet → Sub-net
4. Punctuation & Cadence Commas add ~250ms pauses; periods add ~500ms pauses with falling pitch; ellipses (...) add deliberate hesitation. Welcome, to Support
Hold on... checking now
5. Syllable Stress & Pitch Capitalize stressed syllables in phonetic spellings to guide neural pitch inflection. pro-CESS vs PRO-cess
6. Silent Letter Elimination Remove unpronounced consonants from irregular English or foreign loanwords. Knight → Nite
Pharaoh → Fair-oh
Chopin → Show-pan
7. Homophone Substitution Replace awkward names or vowel clusters with common everyday words that sound identical. Wright → Right
Bough → Bow
Nguyen → Win
8. Symbols & Units Normalization Spell out symbols, fractions, and units to prevent inconsistent pronunciation. & → and
% → percent
1st → First
$ → dollars
9. Compound Word Splitting Insert hyphens between fused technical terms for crisp, clear articulation. Voicemail → Voice-mail
Firewall → Fire-wall
Inline Audio Testing: Click the ▶ Test button on any row in the lexicon table to audition how your phonetic spelling sounds directly in the browser before saving!
Voice AI Knowledge Base (FAQ Engine)

The Knowledge Base allows the appliance to answer frequent caller inquiries (operating hours, location, return policies, payment terms) with sub-50ms instant response latency:

  • Zero-Latency Audio Cache: Spoken answers are pre-cached in memory on appliance startup, bypassing LLM delays.
  • Follow-Up Offers: After answering, the VR can offer assistance (e.g. "Would you like me to transfer you to customer support?").
  • Clean Closure: If the caller says "No thank you, that was all", the VR politely closes the call and sends a clean SIP BYE.
4. Directory & Department Routing Table: Granular Field Reference

The Directory & Skills table maps inbound caller intent and phonetic names to staff extensions, hunt groups, and skill-based queues.

Directory Entry Modal: Granular Field-by-Field Reference
Field Name Operational Purpose Behavior & Detailed Configuration
Select Entry Type Defines entry category and transfer eligibility. • 👤 Staff Member: Individual user eligible for transfers, voice consultation, and voicemail.
• 👥 Department / Group: PBX ring group or hunt queue (typically Blind Transfer).
• 📋 Customer Contact: External client/VIP used exclusively to speak caller identity during consultation; 100% excluded from outbound transfers and phonetic search.
Full Name Spoken display name. The formal name spoken by the VR during caller announcements and transfers. If synchronized from PBX LDAP, it is marked with an amber lock icon and protected against manual or CSV overwrites.
Extension Number Target PBX extension or external phone. Internal PBX extension number (e.g. 210) for staff and groups. For Customer contacts, this holds their external telephone or mobile number for caller ID matching.
Department Key Organizational classification. Text identifier (e.g. sales, support, accounts, general) linking individual staff members to broader departmental queues and hunt groups.
Job Title / Role Functional job description. Title or role function (e.g. Technical Support, Manager, Sales, Customer). Used by the AI to assist intent matching when callers ask by role.
Email Address Voicemail dispatch destination. Destination email where caller audio recordings (.wav) and Faster-Whisper text transcripts are dispatched if the contact is busy or unavailable.
Call Transfer Mode Handoff negotiation style. • Attended Transfer: VR calls the staff member, announces caller identity, and requires verbal ("yes"/"no") or DTMF (press 1) consent before bridging.
• Blind Transfer: Immediate SIP REFER transfer to the PBX extension without consultation (ideal for queues and ring groups).
Caller Announcement Consultation caller announcement. • Enabled: The VR speaks who is calling (recognized Customer name or spoken phone digits) before asking for consent.
• Disabled: Standard generic prompt only ("Call for <Staff>").
Fallback Action Handling when busy or declined. • Voicemail: Record caller message and dispatch email audio/transcript.
• Queue: Divert call to designated department hunt queue.
• Operator: Return caller back to receptionist operator.
Phonetic Aliases Nicknames & speech variations. Comma-separated alternate names, first names, or phonetic variations callers might speak (e.g. Chuck, Chuck Norris, Mr Norris).
Routing Skills & Tiers Skill matching & priority dispatch. Comma-separated skill keywords with optional priority tiers (e.g. Support:1, Billing:2). Tier 1 agents are called first; Tier 2 agents are called if Tier 1 is busy. Equal tiers rotate via longest-idle round-robin.
Anti-Overwrite Protection LDAP synchronization security. Locally created Groups and Customers are never overwritten or deleted by PBX LDAP updates. Custom transfer modes, emails, and aliases assigned to synced staff are strictly preserved across syncs.
Strict 5-Tier Routing Precedence

When a caller speaks, the resolver evaluates candidate targets in strict descending order of certainty:

  1. Exact Extension Match: e.g. "extension two zero one".
  2. Phonetic Full Name: Levenshtein & Soundex fuzzy matching.
  3. Staff Nicknames & Aliases: Configured alternate names.
  4. Department & PBX Groups: Ring groups and hunt queues.
  5. Tiered Skills Hierarchy: Semantic skill matches.
Round-Robin & Hold Audio Continuity

When multiple agents qualify for a skill or department:

  • The agent with the longest idle time is called first to balance workload.
  • If an agent declines or does not answer, the VR automatically progresses to the next agent in the tier.
  • Throughout multi-agent consultation cascades, the caller hears unbroken hold music with zero audio clipping.
5. System Administration, Idle-Gated Updates & Security

Administrative tools for maintaining appliance uptime, configuring role-based access, and applying updates safely.

Idle-Gated Update Coordinator

To guarantee zero dropped calls in production, configuration changes are managed by the UpdateCoordinator:

  • If the appliance is idle (0 active calls), changes are applied immediately.
  • If a call is in progress, changes are queued safely and applied the millisecond the active call completes.
Role-Based Access Control (RBAC)
  • Administrator: Full permissions to modify PBX settings, edit the directory, configure AI models, and manage user accounts.
  • Operator: Read-only monitoring permissions. Can view live call status, trace logs, and directory entries, but cannot alter telephony or system configurations.
Call Analytics & Intelligence Reporting
Operational call volume, recognition confidence in 10% brackets, Voice AI latency telemetry, and historical CDR audit.
From: To:
Total Inbound Calls
--
Analyzed sessions
FAQ Deflection Rate
--%
Self-served inquiries
Transfer Success
--%
Staff transfers
Average Handling Time
--s
Caller interaction time
Voicemails Captured
--
Emailed with transcripts
Model Matching Confidence (10% Brackets) Avg: --%

Distribution of directory & intent matching confidence scores across caller utterances.

Loading confidence distribution...
Call Outcome Distribution

Proportion of call resolution methods across all inbound sessions.

Loading outcomes...
24-Hour Traffic Heatmap

Call volume distribution by hour of day (00:00 to 23:00) identifying peak rush windows.

00:00 (Midnight) 08:00 12:00 (Noon) 16:00 23:00
Top Requested Staff & Departments

Most frequent transfer destinations requested by inbound callers.

Loading department demand...
Voice AI Pipeline Latency Benchmarks

Average sub-second processing timers across speech transcription, intent reasoning, and audio generation.

STT Speech Recognition
-- ms
Faster-Whisper int8 CPU
LLM Dialogue Reasoning
-- ms
Ollama Llama 3.2 (3B)
TTS Speech Synthesis
-- ms
Edge-TTS / 0ms Memory Cache
Total Turnaround Latency
-- ms
Caller silence to spoken audio
Historical Call Detail Records (CDR Log)
Auditable call activity with confidence brackets and delivery outcomes.
Date / Time Caller ID Port Destination Target Outcome Confidence Bracket Duration Voicemail
Loading call records...
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Appliance Network & SSL Management
Manage local appliance IP addressing, hostname, edge reverse proxy (NPMplus), and SSL certificates.
🔒 Master Admin Only
Network Interface
ens18
--:--:--:--:--:--
Current IPv4 Address
192.168.1.154/24
DHCP Active
Default Gateway
192.168.1.1
Active DNS Resolvers
192.168.1.254
OS Resolver
🌐 Appliance Identity & Network Settings

Configure network addressing mode and local appliance hostname.

Static IP Parameters
⚠️ Note: Changing the IP address will alter the URL used to access this web portal and may require updating PBX SIP trunk endpoints.
🔀 Master Reverse Proxy Setup (NPMplus) Edge Architecture

When sharing a single public IP address across multiple internal services, your public SSL certificate lives on your frontend NPMplus / Nginx Proxy Manager server. The proxy terminates HTTPS and forwards traffic to this internal appliance.

💡 Where does the SSL certificate go?
Install the public certificate (Let's Encrypt / Wildcard) on NPMplus at the network edge. This appliance communicates via HTTP over the private LAN, and NPMplus handles public HTTPS, HTTP/2, and WebSocket proxying.
Enter the public hostname you have pointed to your master public IP in DNS.
Provides 1-click step-by-step instructions for NPMplus Web UI and copyable WebSocket configuration.
🔒 Appliance SSL Certificate (Local HTTPS) Valid

Active certificate installed at /opt/virtual-receptionist/config/certs/cert.pem used when accessing this appliance directly via LAN IP.

Common Name (CN)
192.168.1.154
Issuer
192.168.1.154 (Self-Signed)
Validity Period
Oct 21, 2026 – Oct 21, 2027
Days Remaining
358 Days
SHA-256 Fingerprint: Loading...