; ---------------------------------------------------------------------------
; 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 *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 }}:{{ asterisk_gateway_port }}

[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 }}

; ---------------------------------------------------------------------------
; 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
remove_existing = yes
qualify_frequency = 60