Two bugs found via a live test with two real registered browser softphones:
1. retry_escalation_no_answer() re-resolved the client channel via
_resolve_handoff_channel() -> a live AMI CoreShowChannels round-trip that
can take ~10s. The new mvpcc-transfer dialplan wait window (MusicOnHold,
also ~10s, added to give the backend time to redirect before the final
Hangup) was consistently LOST to this exact same duration: the backend's
AMI Redirect fired against a channel the dialplan had already hung up
('Channel does not exist: PJSIP/...', confirmed in escalation timeline).
Fixed by reusing the actively-maintained AsteriskCallLinkRow.channel_name
directly (unchanged for a PJSIP channel across Redirect between contexts
of the same call) instead of re-discovering it, falling back to the slow
path only if that field is empty.
2. VoiceEventIn.event_type is a pydantic Literal restricted to 6 legacy
values (call.started/ivr.completed/...). None of the Phase 2 event
catalog names (AgentReserved/AgentRinging/AgentNoAnswer/AgentConnected/
TransferCompleted/TransferFailed) were ever in it, so every single
_emit_escalation_event() call has been failing with 422 since Phase 2
shipped (swallowed silently by the broad except there) - confirmed by
calling app._emit_voice_event() directly against the running service.
Extended the Literal to include all six.
process_agent_dial_outcome only recognized hangup causes 17/18/19/21/34/38.
When the reserved agent's AOR has zero registered contacts (e.g. the
softphone dropped, or nobody ever registered), Asterisk immediately
hangs up with cause 3 (no route to destination) instead of running a
Dial() long enough to produce a DialEnd/NOANSWER at all - so the retry
listener silently ignored it and the escalation was left dangling in
'ringing' status (the agent itself still got released via the
call-ended fallback path, but no retry to the next agent was ever
attempted and the escalation record never reflects the failure).
Added causes 1 (unallocated number), 3 (no route), 20 (subscriber
absent), 22 (number changed) alongside the existing set.
request_handoff() (called by ai_voice_runtime_service for every real
call handoff, the only handoff path production calls actually use)
reserved an agent from the same routing pool as create_escalation()
but never created an EscalationRow, so the Phase 2 no-answer-retry
listener (DialEnd/Hangup) could never find it. A failed transfer
(no SIP registration, no answer, redirect error) left the agent
stuck in RESERVED forever with no retry to the next agent.
Now creates an EscalationRow (status=ringing) alongside the agent
reservation, releases the agent + marks the escalation failed if the
AMI Redirect itself errors immediately, and lets the existing
DialEnd/Hangup handler drive no-answer retry / release exactly like
the /escalations endpoint already does.
Reproduced live: call handed off to extension 2002 with no SIP
contact registered -> immediate hangup, cause=3, both pool agents
stuck in RESERVED indefinitely (had to release manually via psql).
Phase 2 of the L1->L2 routing engine (Phase 1: MR!4).
- ami_loop() now also captures native AMI DialEnd/Hangup frames (not
only UserEvent), needed to detect that an escalated agent did not
answer. No dialplan change required - Redirect already routes the
client channel into an existing Dial()-based transfer context, so
Asterisk emits these events on its own; the listener just wasn't
reading them before.
- retry_escalation_no_answer(): on NOANSWER/BUSY/CANCEL/CHANUNAVAIL/
CONGESTION, releases the non-answering agent, excludes it, and
reserves+redirects to the next available agent via the routing
engine's existing exclude_agent_ids support. Exhausted pool marks
the escalation failed and leaves the call with the AI instead of
dropping the client (ТЗ §32).
- Agent status now actually moves through
RESERVED -> RINGING -> TALKING -> AFTER_CALL_WORK -> AVAILABLE
instead of staying stuck on RESERVED for the whole call; a new
acw_sweep_loop background thread (same pattern as the existing
failed_retry_loop) times out AFTER_CALL_WORK back to AVAILABLE.
- escalations gains attempt_count/real_agent_id/attempted_agent_ids_json
(migration 0033); fixes a latent bug where assigned_agent_id stored
the SIP extension instead of the real agent_id despite routing-service
already returning it in RoutingAgentReserveOut.
- Every transition now records an interaction timeline entry and
publishes the ТЗ §25 event catalog (AgentReserved/AgentRinging/
AgentNoAnswer/AgentConnected/TransferCompleted/TransferFailed)
through the existing emit_voice_event/EventOutboxRow idempotent path.
Not in this MR (see plan): SLA config, Callback, L3 (needs real
technical agents from the business), metrics.
Found during production smoke test: if the AMI Redirect call in
create_escalation() raises (channel gone, AMI hiccup), the agent stays
RESERVED forever with no owning call — orphaned out of the pool until
someone fixes it by hand. Now releases the agent and marks the
escalation failed before re-raising as a 502.
Replaces the hardcoded single-extension redirect for AI->human call
escalation with a real Agent Pool + Routing Engine:
- agents/escalations/routing_rules tables (migration 0031), asterisk_call_links
gains tenant_id/current_level/required_skills_json/priority.
- services/routing_service/engine.py: level/tenant/skill filtered agent
selection with atomic (CAS) reservation, no double-booking.
- routing-service: /agents CRUD + /internal/routing/reserve-agent and
/internal/routing/release-agent.
- asterisk-bridge-service: voice_ai.request_handoff now uses the Routing
Engine automatically for any queue_code configured in
ASTERISK_QUEUE_LEVEL_MAP_JSON (all other queue_codes keep the existing
static ASTERISK_TRANSFER_TARGET_MAP_JSON behavior unchanged); new
POST /asterisk/live-calls/{call_id}/escalations entrypoint; agent is
released back to AVAILABLE and the escalation closed when the call ends.
Targets the Tele2 Kazgaz DID +77476456048 (from-tele2-kazgaz context) as the
first queue wired to real L2 routing instead of AI-only.
Known gap (documented in docs/architecture/l1-l2-routing-engine.md):
automatic no-answer retry-to-next-agent needs a small, separately reviewed
dialplan change and is left for a follow-up MR rather than guessed at blind.
Tests: services/routing_service/engine.py covered by
tests/test_routing_engine.py (selection filtering, atomic reservation,
release); existing test_asterisk_bridge_service.py and
test_routing_service_pg_counter.py suites still pass unmodified.
arys