# ── SIP Client Configuration ──────────────────────────────────
#
# Zorunlu: name, username, password, server
# Diğer tüm alanlar opsiyoneldir.

[web]
port = 9090
username = "admin"
password = "admin"

[commands_api]
port = 9099

[[accounts]]
name = "alice"
username = "alice"
password = "secret123"
register_retry_interval = 30
domain = "sip.example.com"
server = "192.168.1.1:5060"
sip_port = 5060
rtp_port_start = 8000
rtp_port_end = 8010
auth_method = "md5"
codec = "pcmu"

# ── Identity ─────────────────────────────────────────────────
display_name = "Alice Smith"
asserted_id = "sip:+44123456789@example.com"

# ── IVR / Auto-Attendant ─────────────────────────────────────
auto_answer = true
ivr_welcome = "welcome.wav"           # 8kHz 16-bit mono PCM WAV
ivr_timeout = 10                       # DTMF bekleme süresi (saniye)

# DTMF menüsü: tuş = aksiyon
# Format: transfer:URI | playback:dosya.wav | record:dosya.wav:saniye | hold | hangup
ivr_menu = { "1" = "transfer:sip:1001@sip.example.com", "2" = "transfer:sip:1002@sip.example.com", "3" = "playback:info.wav", "4" = "record:voicemail.wav:30", "5" = "hold", "*" = "hangup" }
ivr_default = "transfer:sip:operator@sip.example.com"

# ── Protocol ─────────────────────────────────────────────────
register_expiry = 3600
user_agent = "RustSIP/1.0"
dtmf_mode = "rfc2833"
early_media = true
session_timers = false

# ─────────────────────────────────────────────────────────────
[[accounts]]
name = "bob"
username = "bob"
password = "secret456"
domain = "sip.example.com"
server = "192.168.1.1:5060"
sip_port = 5070
rtp_port_start = 8012
rtp_port_end = 8022
auth_method = "md5"
codec = "pcma"
display_name = "Bob Jones"

[[accounts]]
name = "charlie"
username = "charlie"
password = "secret789"
domain = "pbx.other.com"
server = "10.0.0.50:5060"
sip_port = 0
rtp_port_start = 9000
rtp_port_end = 9010
auth_method = "none"
codec = "opus"
display_name = "Charlie"
asserted_id = "sip:charlie-trusted@pbx.other.com"
