active-call 0.3.22

A SIP/WebRTC voice agent
Documentation 
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addr = "0.0.0.0"
http_addr = "0.0.0.0:8080"
udp_port = 13050
# http_gzip = true
log_level = "debug"
#log_file = "/tmp/rustpbx.log"
# Paths can include a trailing * to skip whole subtrees
# http_access_skip_paths = ["/health", "/metrics*"]
media_cache_path = "./config/mediacache"

# external IP address for SIP signaling and media
# if server is behind NAT, set your public IP here (without port)
# external_ip = "1.2.3.4"

# rtp_start_port = 20000
# rtp_end_port = 30000

# setup your STUN/TURN servers here, for webrtc web clients
# [[ice_servers]]
# urls = ["stun:stun.l.google.com:19302"]

# [[ice_servers]]
# urls = ["turn:restsend.com:3478"]
# username = "user"
# credential = "pass"

sip_addr = "0.0.0.0"
sip_port = 13050     # don't use 5060

[sip_handler]
type = "webhook"
url = "http://localhost:8090/webhook"
method = "POST"

[recording]
enabled = true
auto_start = true

[callrecord]
type = "local"
root = "./config/cdr"