Technical Support FAQ

Overview

This FAQ answers common troubleshooting questions grouped by topic: voicemail detection, call forwarding and transfers, idle messages, SIP networking, and how to report an issue to support with the right information attached. For the marketing/general-onboarding FAQ, see the existing Frequently Asked Questions page in the Support category — this page is scoped specifically to technical, in-product troubleshooting questions.

Which detection provider should I choose?

The Sulus provider is strongly recommended for most use cases — it's fast, accurate, and gracefully handles interruptions. Google is a solid second choice with slightly longer detection time. OpenAI offers high accuracy and flexible phrasing at a higher cost. Legacy carrier beep detection is fast but prone to false positives, so reserve it for special cases. See Voicemail Detection for the full provider comparison table.

What's the difference between audio and transcript detection?

The type parameter controls this: audio (the default) analyzes the audio signal directly and is the only option available for the Sulus provider; transcript is ASR-based and is the only type supported by the Google and OpenAI providers.

How do I fix false positives?

False positives usually come down to timing or provider choice. Adjust backoffPlan.startAtSeconds and backoffPlan.frequencySeconds to change how soon and how often detection checks run, or switch providers — the Sulus provider's hybrid approach (model-based detection plus real-time call monitoring) is built specifically to catch a human picking up unexpectedly and cancel a false machine-detection.

How do I fix missed voicemails?

A missed voicemail is usually a timing issue on the other side: beepMaxAwaitSeconds (default 30 seconds) controls how long the assistant waits for a beep before speaking. Setting it too low, under 15–20 seconds, can cause your message to be cut off or missed since most voicemail greetings run 10–20 seconds before the beep. Raise it toward the conservative 25–30 second range if voicemails are being missed.

Can I use a pre-recorded audio file as the voicemail message?

Yes — rather than relying on TTS to speak voicemailMessage, you can configure a pre-recorded audio file to be played once voicemail is detected. See Pre-Recorded Voicemail Messages for setup details.

Can I disable voicemail detection entirely?

Yes, voicemail detection can be turned off entirely on the assistant if you'd rather every outbound call simply proceed as if a live person answered.

Why does a call drop immediately after transfer?

Start by checking the endedReason field on the call — it tells you whether the drop was a transfer-specific failure (see Ended Reason: Complete Reference for the full transfer-error list) or something unrelated. The next most common cause is attempting an unsupported from/to combination:

See Call Forwarding & Transfer Modes for the full SIP REFER signaling explanation behind this table.

Why aren't my idle messages triggering?

Idle messages are configured via the customer.speech.timeout hook, and they're automatically disabled during tool calls and warm transfers — if the assistant is mid-tool-call or mid-transfer when the silence occurs, the hook won't fire. Also confirm triggerMaxCount hasn't already been reached for that call; once it has, a log entry notes the hook was skipped for that reason.

How many times do they trigger, and how do I reset the count?

triggerMaxCount caps how many times the hook fires per call, from 1 to 10 (default 3). Set triggerResetMode to onUserSpeech if you want the count to reset whenever the caller speaks, rather than counting down permanently for the whole call (the never default).

What timeout should I use?

See Idle Messages for the full hook configuration reference, including how to escalate through multiple timeout tiers up to ending the call.

Before a BYO SIP trunk will pass calls, your carrier needs to allowlist the platform's signaling IPs for your region:

Note that US RTP media IPs are dynamic while the EU region uses a single static RTP IP — if you're pre-allowlisting media IPs on a strict firewall, the EU static IP can be allowlisted directly, but the US side needs the full UDP port range allowed rather than a fixed IP. See SIP Trunk Troubleshooting for the complete gateway field reference and SIP response code lookup table.

The information that speeds up a support ticket depends on the type of issue:

Your Call ID is visible on each call's detail view under Observe > Logs > Calls in the dashboard. Your Organization ID is available from your account/organization settings page. Never include SIP passwords or API keys in anything you share with support.