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Documentation Index

Fetch the complete documentation index at: https://docs.getsmartalex.com/llms.txt

Use this file to discover all available pages before exploring further.

Not a deep glossary. Short definitions with cross-links to wherever the term actually matters.

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Protocol and signalling

The signalling protocol for setting up, modifying, and tearing down voice calls over IP networks. Defined in RFC 3261. Think of it as “the HTTP of phone calls”. SmartAlex communicates with your PBX using SIP.
Carried inside SIP messages. Describes the media session: which codecs, which RTP ports, what capabilities. Negotiation happens in the INVITE and its 200 OK response.
Carries the actual audio. Uses UDP. Once a SIP session is established, audio flows over RTP directly between endpoints (or through a media proxy like SmartAlex).
Encrypted RTP. Same protocol, TLS-style encryption for the media layer.
The SIP method that starts a call. The caller’s side sends an INVITE; the target side responds with 100 Trying, 180 Ringing, and finally 200 OK.
Ends a call. Either side can send it.
The SIP method for “please transfer this call to somewhere else”. SmartAlex sends a REFER to your PBX when the AI wants to hand the caller over to a human. Key mechanic covered in Call Routing & Transfers.
An INVITE sent on an already-active call. Used to change media parameters (e.g., put the caller on hold, or change codec). Different from REFER.
A SIP mechanism where a new dialog replaces an existing one. Used in call-parking scenarios. SmartAlex handles it transparently; not a user-facing feature.
A SIP header (RFC 3325) that asserts the identity of the original caller when a call is transferred through trusted intermediaries. SmartAlex sets this header on transfers so your PBX shows the real caller’s number on the target extension.

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Infrastructure

Your company phone system. Handles extensions, ring groups, queues, voicemail, call recording. SmartAlex integrates with your PBX, it doesn’t replace it. Common examples: 3CX, Yeastar, FreePBX, Avaya, Mitel.
A SIP connection between two phone systems. In SmartAlex, a trunk is the configured connection between your PBX and our voice platform. One trunk can carry many concurrent calls.
A phone number. “DID” is the American term, “DDI” is the European term, same thing. E.g., +27 87 250 0100 is a DID.
The international standard format for phone numbers: + followed by country code, area code, and subscriber number, with no spaces or punctuation. +27872500100 is E.164. SmartAlex uses E.164 internally everywhere.
A specialised firewall/proxy for SIP traffic. Used in enterprise deployments to bridge between networks, enforce security policies, and interoperate with systems that don’t speak standard SIP directly (e.g., Microsoft Teams Direct Routing).
The traditional telephone network. When a call leaves the IP world and goes to a “real” phone (mobile, landline), it traverses the PSTN.
Cloud-based platform that bundles voice, video, messaging, and collaboration. Examples: RingCentral, 8x8, Microsoft Teams Phone.

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Call handling

An internal identifier for a person or group on a PBX. Usually 2 to 6 digits. E.g., 101 might be Sales, 200 might be a queue.
A holding line for callers waiting for the next available agent in a group. Typically plays hold music and announces position.
A group of extensions that ring simultaneously (or in sequence) when a call comes in. First person to answer takes the call.
Automated menu system. “Press 1 for sales, 2 for support.” SmartAlex replaces these with natural conversation. See Replacing Your IVR.
A log entry capturing everything about a call: numbers, duration, direction, outcome, trunk used. Your PBX generates CDRs. SmartAlex generates its own call records, which complement but don’t replace your PBX CDRs.
CLI (Calling Line Identification) is the caller’s phone number. CNAM (Calling Name) is the caller’s display name, looked up via a subscriber database (US-specific feature).
American equivalent of CLI. Same concept: the caller’s number.

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Auth and security

Username and password auth for SIP. When your PBX registers or places a call, it proves identity with these credentials.
A list of IP addresses permitted to send SIP traffic to a trunk. Alternative to digest auth. Requires the PBX to have a static public IP.
A firewall feature that tries to “help” SIP traffic traverse NAT by rewriting headers. It invariably breaks something. Always disable it. Covered in Network Requirements.
Your firewall or router rewriting internal IPs to a single external IP. Generally fine for SIP as long as SIP ALG is off.

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Call quality

A 1-to-5 rating of voice quality. 4.0 and above is considered good. Affected by codec, packet loss, jitter, delay.
Variation in packet arrival time. High jitter causes choppy audio. Buffered by the jitter buffer in each endpoint.
Percentage of RTP packets dropped in transit. Above 1% is audible. Above 3% is poor quality.
Time from one person speaking to the other hearing. Under 150 ms is excellent, 150 to 250 ms is acceptable, above 300 ms starts to feel awkward.
Delay introduced by batching audio into packets. G.711 uses 20 ms packets; G.729 is similar. Negligible compared to network latency.

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SmartAlex-specific

An AI voice persona configured in SmartAlex. Has a first message, system prompt, voice, tools, and optional knowledge base. One agent per AI use case.
A SmartAlex customer account. All agents, trunks, call records, and configurations are scoped to one workspace, and data never crosses workspace boundaries.
An action the AI can invoke during a call. Examples: check calendar availability, look up a customer in CRM, transfer to an extension, send SMS.
Documents uploaded to an agent that the AI can search during calls. Used to answer company-specific questions.
SmartAlex deployment model where the customer brings their own carrier account. Common with existing enterprise customers who want to keep their carrier contracts.

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Next steps

Architecture

See these terms in context.

Overview

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