test(lab): add Asterisk lab — a fake PSTN for media validation

An Asterisk instance that answers calls, plays an IVR, holds with music and
connects a "human", so the gateway has something real to dial that is not the
PSTN: no charges, no strangers, no E911 exposure.

Asterisk rather than Kamailio because the unproven risks are media risks.
Kamailio is a proxy — it routes signalling and answers nothing, so it would
forward the INVITE and find nobody home. Asterisk is a B2BUA: it answers,
plays prompts and collects DTMF, which is the hold-slayer scenario itself.
Kamailio remains the better model for trunk registration/digest auth later.

No application changes are needed to use it. SIP_TRUNK_HOST is just an
address, so the production code path runs unmodified — there is no test-only
branch anywhere in the gateway. It also means safety is structural: while the
trunk points at the lab there is no route to the PSTN at all, an absence of
route rather than a policy that could be misconfigured.

Nine scenarios (1001-1008 plus an echo test) cover the baseline call, the
IVR/DTMF path, hold-then-human, long hold, busy, no-answer, remote hangup and
silence.

The image ships no sound files, so sounds/generate.py synthesises three
fixtures from fixed seeds — byte-identical on every run, which is what makes
a classifier regression distinguishable from noise. Verified against
AudioClassifier: music→MUSIC 0.85, speech→LIVE_HUMAN 0.75,
silence→SILENCE 1.00. The speech formants deliberately avoid the DTMF bands;
the first version landed on a valid pair and classified as a keypress.

Anonymous inbound calls are refused, and endpoint matching is by source
address — Asterisk's default matches the From-header domain, which Hold
Slayer populates from its SIP bind address (0.0.0.0 on a wildcard bind).

Generated audio and the rendered per-host configs are gitignored: the former
is reproducible from a fixed seed, the latter carry a host-specific IP and
the lab password.

Known limit, documented in the README: MediaPipeline.create_tap is a stub, so
the classifier receives no audio on a live call. RTP flows and Asterisk plays
audio, but the tap is never fed — the fixture results above were measured by
feeding the classifier directly. This blocks scenarios 1002/1003/1004.

Co-Authored-By: Claude Opus 5 (1M context) <noreply@anthropic.com>

tests/lab/dialplan/pjsip.conf

@@ -0,0 +1,75 @@
+; ---------------------------------------------------------------------------
+; Hold Slayer lab — PJSIP configuration
+; ---------------------------------------------------------------------------
+; SECURITY: this endpoint answers calls. Asterisk's stock examples allow
+; anonymous inbound, which is a well-known toll-fraud target. This config
+; refuses it: every call must authenticate as the `hold-slayer` endpoint.
+;
+; There is no PSTN behind this Asterisk — an unauthorised call reaches only
+; the lab dialplan and costs nothing. The lock-down is defence in depth and
+; so this config is never copied somewhere it would matter.
+; ---------------------------------------------------------------------------
+
+[global]
+type = global
+; Do not fall through to an `anonymous` endpoint for unmatched calls.
+; This is the single most important line in the file.
+unidentified_request_count = 5
+unidentified_request_period = 5
+unidentified_request_prune_interval = 30
+
+[transport-udp]
+type = transport
+protocol = udp
+bind = 0.0.0.0:{{ asterisk_sip_port }}
+; The address Asterisk advertises in SDP. Without this, containers advertise
+; their internal bridge IP and RTP arrives at an unroutable address — the
+; classic "call connects but there is no audio" failure.
+external_media_address = {{ asterisk_external_ip }}
+external_signaling_address = {{ asterisk_external_ip }}
+local_net = {{ asterisk_local_net }}
+
+; ---------------------------------------------------------------------------
+; Hold Slayer endpoint
+; ---------------------------------------------------------------------------
+; Hold Slayer authenticates as this endpoint to place calls into the lab.
+
+; Identify the endpoint by source address. Asterisk's default matching uses
+; the From-header domain, which Hold Slayer populates from its SIP bind
+; address (0.0.0.0 on a wildcard bind) — never a value Asterisk can match.
+; Matching on where the packet actually came from sidesteps that.
+[hold-slayer]
+type = identify
+endpoint = hold-slayer
+match = {{ asterisk_match_host }}
+
+[hold-slayer]
+type = endpoint
+context = hold-slayer-lab
+disallow = all
+; ulaw first: it is what the PSTN uses, so the lab exercises the same codec
+; path a real trunk would. alaw as fallback.
+allow = ulaw
+allow = alaw
+auth = hold-slayer-auth
+aors = hold-slayer
+; RFC 2833 out-of-band DTMF — what send_dtmf must produce. Setting this
+; explicitly (rather than `auto`) means a DTMF failure is a real failure and
+; not a negotiation fallback quietly rescuing it.
+dtmf_mode = rfc4733
+direct_media = no
+force_rport = yes
+rewrite_contact = yes
+rtp_symmetric = yes
+
+[hold-slayer-auth]
+type = auth
+auth_type = userpass
+username = {{ asterisk_sip_username }}
+password = {{ asterisk_sip_password }}
+
+[hold-slayer]
+type = aor
+max_contacts = 2
+remove_existing = yes
+qualify_frequency = 60