tests/lab/README.md

@@ -63,6 +63,72 @@ GATEWAY_SIP_PORT=21062     # must differ from the Asterisk port
 > engine is assigned — `main.py`'s lifespan calls `build_sip_engine()` after
 > construction. A harness that skips that step silently tests the mock.
 
+## The softphone (transfer target)
+
+**Asterisk is the registrar for devices, not Hold Slayer.** The gateway reaches
+a desk phone by dialling extension `2001`, which Asterisk routes to whatever
+has registered as `softphone`. This deliberately avoids Hold Slayer's own SIP
+listener, which answers `200 OK` to any REGISTER with no digest challenge.
+
+The `pjsua` CLI built alongside the Python bindings is the test device — same
+library stack as the gateway, no extra dependency. It needs an RPATH patch like
+the bindings did:
+
+```bash
+cp ~/src/pjproject/pjsip-apps/bin/pjsua-x86_64-pc-linux-gnu ~/.local/bin/pjsua
+patchelf --set-rpath $HOME/.local/lib ~/.local/bin/pjsua
+```
+
+Register it (config file avoids shell-quoting pain):
+
+```bash
+cat > softphone.cfg <<'EOF'
+--null-audio
+--auto-answer=200
+--max-calls=4
+--local-port=21070
+--id=sip:softphone@127.0.0.1
+--registrar=sip:127.0.0.1:21061
+--realm=asterisk
+--username=softphone
+--password=labphone
+--log-level=3
+EOF
+
+# pjsua is an interactive console app: it exits ~8s after start if stdin is
+# closed or /dev/null. Hold a fifo open on stdin — `script -qfc` and
+# `setsid </dev/null` both look like they work (registration succeeds) and
+# then the process dies, leaving a stale contact in Asterisk that routes
+# INVITEs to a port nobody is listening on.
+mkfifo sp.fifo
+setsid sh -c 'exec 3<>sp.fifo; pjsua --config-file softphone.cfg <&3 >softphone.log 2>&1' &
+```
+
+Verify — **check the port is actually bound**, not just that Asterisk holds a
+contact, since a stale registration outlives the process:
+
+```bash
+ss -lnup | grep 21070                                     # must be listening
+docker compose -f docker-compose.lab.yml exec asterisk \
+    asterisk -rx "pjsip show contacts"                    # must show softphone
+```
+
+Then place a call to `2001`. Both legs should show `Up` under one bridge id:
+
+```bash
+docker compose -f docker-compose.lab.yml exec asterisk \
+    asterisk -rx "core show channels concise"
+```
+
+> **`--realm=asterisk`, not `--realm='*'`** — the wildcard fails with
+> `PJSIP_EFAILEDCREDENTIAL` against Asterisk's digest challenge.
+
+> **Qualify is off** for this AOR (`qualify_frequency = 0`): the pjsua console
+> does not answer `OPTIONS`, so polling marks a working softphone `Unavail` and
+> the dialplan refuses to ring it. The `2001` guard therefore tests
+> `PJSIP_AOR(softphone,contact)` rather than `DEVICE_STATE`. A real hardphone
+> answers OPTIONS and can have qualify re-enabled.
+
 ## Useful commands
 
 ```bash

tests/lab/dialplan/extensions.conf

@@ -128,6 +128,23 @@ exten => 1008,1,NoOp(LAB 1008: silence)
  same => n,Wait(40)
  same => n,Hangup()
 
+; --- 2001: ring the registered softphone ----------------------------------
+; The transfer target. Asterisk is the registrar for devices, so the gateway
+; reaches a desk phone by dialling this rather than by registering it itself.
+; Fails fast when nothing is registered — a silent 30s ring would look like a
+; gateway bug rather than an absent softphone.
+exten => 2001,1,NoOp(LAB 2001: ring softphone)
+ ; Count registered contacts rather than DEVICE_STATE: device state follows
+ ; the OPTIONS qualify, which is off for this AOR (the pjsua CLI does not
+ ; answer OPTIONS), so a registered softphone would still read UNAVAILABLE.
+ same => n,GotoIf($[${PJSIP_AOR(softphone,contact)} = ""]?nodevice)
+ same => n,Dial(PJSIP/softphone,30)
+ same => n,Hangup()
+ same => n(nodevice),NoOp(LAB 2001: no softphone registered)
+ same => n,Answer()
+ same => n,Playback(lab-speech)
+ same => n,Hangup()
+
 ; --- echo test ------------------------------------------------------------
 ; Not a scenario — a debugging aid. Echoes audio back so you can confirm
 ; bidirectional RTP by ear when something looks wrong.

tests/lab/dialplan/pjsip.conf

@@ -34,14 +34,21 @@ local_net = {{ asterisk_local_net }}
 ; ---------------------------------------------------------------------------
 ; 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.
+; Identify the endpoint by source address *and port*. 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 came from sidesteps that.
+;
+; The port is essential when the softphone runs on the same host: a
+; host-only match claims *every* packet from that address, so the
+; softphone's REGISTER would be attributed to this endpoint and checked
+; against the gateway's password ("Failed to authenticate", confusingly).
+; Endpoints that authenticate by username (the softphone) must not be
+; covered by an identify block.
 [hold-slayer]
 type = identify
 endpoint = hold-slayer
-match = {{ asterisk_match_host }}
+match = {{ asterisk_match_host }}:{{ asterisk_gateway_port }}
 
 [hold-slayer]
 type = endpoint
@@ -68,6 +75,56 @@ auth_type = userpass
 username = {{ asterisk_sip_username }}
 password = {{ asterisk_sip_password }}
 
+; ---------------------------------------------------------------------------
+; Softphone endpoint — the transfer target
+; ---------------------------------------------------------------------------
+; Asterisk is the registrar for devices, not Hold Slayer. A softphone REGISTERs
+; here and the gateway transfers a live call to it by dialling extension 2001.
+;
+; Deliberate: Hold Slayer's own SIP listener answers 200 OK to any REGISTER
+; with no digest challenge, so anything on the network could register as a
+; device and receive transferred calls. Keeping registration in Asterisk means
+; the lab does not depend on that path, and the softphone is authenticated.
+;
+; Test with the pjsua CLI built alongside the Python bindings:
+;   pjsua --null-audio --auto-answer=200 \
+;         --id=sip:softphone@<asterisk-host> \
+;         --registrar=sip:<asterisk-host>:21061 \
+;         --realm='*' --username=softphone --password=<pw> \
+;         --local-port=<free port>
+
+[softphone]
+type = endpoint
+context = hold-slayer-lab
+disallow = all
+allow = ulaw
+allow = alaw
+auth = softphone-auth
+aors = softphone
+dtmf_mode = rfc4733
+direct_media = no
+force_rport = yes
+rewrite_contact = yes
+rtp_symmetric = yes
+
+[softphone-auth]
+type = auth
+auth_type = userpass
+username = {{ asterisk_softphone_username }}
+password = {{ asterisk_softphone_password }}
+
+[softphone]
+type = aor
+; The device's contact is learned from its REGISTER rather than configured —
+; a softphone's port is not known in advance.
+max_contacts = 1
+remove_existing = yes
+; No qualify: the pjsua CLI does not answer OPTIONS while sitting at its
+; console prompt, so polling marks a perfectly working softphone Unavail and
+; the dialplan refuses to ring it. Registration itself is the liveness signal
+; here. A real hardphone answers OPTIONS and can have qualify re-enabled.
+qualify_frequency = 0
+
 [hold-slayer]
 type = aor
 max_contacts = 2