transcribe

name	transcribe
description	Start a phone line that transcribes incoming call audio to text. Use when the user asks to transcribe a phone call, get a live transcript of a call, capture speech-to-text from a phone number, or convert call audio to text. Also triggers on "transcribe the call", "live transcript of incoming calls", "speech-to-text for phone".
when_to_use	The user wants to receive calls and transcribe the caller's speech to text. Runs the VoIP CLI transcribe command, which waits for an inbound call (or dials out with --dial) and prints each transcribed utterance.
argument-hint	[--dial TARGET] [--stt-model MODEL]
allowed-tools	Bash(uvx*) Bash(voip sip transcribe*)

Transcribe call audio to text

Start a phone line that transcribes incoming call audio to text using a local Whisper model. Each transcribed utterance is printed to stdout. Use this skill when the user wants a live speech-to-text feed from a phone number.

When to use this skill

Use this skill when the user wants to transcribe phone call audio. Examples:

• "Transcribe my incoming calls."
• "Get a live transcript of the next call to my SIP number."
• "Dial [phone] and transcribe what they say."

If the user wants a two-way AI conversation, use the call skill instead.

Prerequisites

The user needs a SIP account. The SIP address-of-record (AOR) looks like:

sip:USER:PASSWORD@SIP_SERVER

The VoIP CLI is distributed as the voip[cli] Python extra and runs via uvx. The first run downloads the model and may take longer.

Commands

Wait for an inbound call and transcribe it:

uvx 'voip[cli]' sip sip:USER:PASSWORD@SIP_SERVER transcribe

Dial out and transcribe the remote party:

uvx 'voip[cli]' sip sip:USER:PASSWORD@SIP_SERVER transcribe --dial sip:[e-mail]

Common options:

• --stt-model: Whisper model size (default tiny). Larger models are more accurate but slower. Common values: tiny, base, small, medium.
• --dial TARGET: dial a target instead of waiting for an inbound call.
• --stun-server HOST[:PORT]: STUN server for RTP NAT traversal (default stun.cloudflare.com:3478).
• --no-verify-tls: disable TLS certificate verification (testing only).

Instructions

1. Ask the user for their SIP AOR if it was not already provided. Never invent credentials.
2. Confirm whether the user wants to wait for an inbound call or dial out. If dialing out, get the target number or SIP URI.
3. Run the transcribe CLI command shown above with the user's AOR and any requested options. Stream the transcribed utterances back to the user.
4. Stop the command (Ctrl-C / send a termination signal) when the user is done.