[general]
; This section sets global variables of the implementation

; port: int: SIP UDP port
;port=5060

; addr: ipaddress: IP address to bind to
;addr=0.0.0.0

; maxpkt: int: Maximum received packet size, 524 to 65528, default 1500
;maxpkt=1500

; buffer: int: Requested size of socket's receive buffer, 0 to use default
;buffer=0

; floodevents: int: How many SIP events retrieved in a row trigger flood warning
;floodevents=20

; maxforwards: int: Default Max-Forwards header, used to avoid looping calls
;maxforwards=20

; useragent: string: String to set in User-Agent or Server headers
;useragent=YATE/2.0.0

; realm: string: Authentication realm to offer in authentication requests
;realm=Yate

; transfer: bool: Allow handling the REFER message to perform transfers
;transfer=enable in server mode, disable in client mode

; registrar: bool: Allow the SIP module to receive registration requests
;registrar=enable in server mode, disable in client mode

; options: bool: Build and send a default 200 answer to OPTIONS requests
;options=enable

; prack: bool: Enable acknowledging provisional 1xx answers (RFC 3262)
;prack=disable

; info: bool: Accept incoming INFO messages
;info=enable

; fork: bool: Follow first forked 2xx answer on early dialogs
;fork=enable

; progress: bool: Send an "183 Session Progress" just after successfull routing
;progress=disable

; generate: bool: Allow Yate messages to send arbitrary SIP client transactions
;generate=disable

; nat: bool: Enable automatic NAT support
;nat=enable

; ignorevia: bool: Ignore Via headers and send answer back to the source
;  This violates RFC 3261 but is required to support NAT over UDP transport.
;ignorevia=enable

; lazy100: bool: Do not generate an initial "100 Trying" for non-INVITE
;  transactions unless a retransmission arrives before having a final answer
;lazy100=no

; dtmfinband: bool: Generate DTMF inband by default
;dtmfinband=no

; dtmfinfo: bool: Generate INFO messages to send keypad tones
;dtmfinfo=no

; rfc2833: bool: Offer RFC2833 telephone-event by default
;rfc2833=yes

; privacy: bool: Process and generate privacy related SIP headers
;privacy=disable

; secure: bool: Generate and accept RFC 4568 security descriptors for SRTP
;secure=disable

; forward_sdp: bool: Include the raw SDP body to be used as-is for forwarding RTP
;forward_sdp=disable

; rtp_localip: ipaddress: IP address to bind local RTP to, empty to guess best
;rtp_localip=

; rtp_start: bool: Start RTP when sending 200 on incoming instead of receiving ACK
;rtp_start=disable

; multi_ringing: bool: Accept provisional (1xx) messages even after 180 Ringing
;multi_ringing=disable

; refresh_nosdp: bool: Accept session refresh reINVITEs that lack a SDP offer
;refresh_nosdp=enable


[registrar]
; Controls the behaviour when acting as registrar

; expires_min: int: Minimum allowed expiration time in seconds
;expires_min=60

; expires_def: int: Default expiration time if not present in REGISTER request
;expires_def=600

; expires_max: int: Value used to limit the expiration time to something sane
;expires_max=3600

; auth_required: bool: Automatically challenge all clients for authentication
;auth_required=enable

; nat_refresh: int: Proposed client NAT refresh interval in seconds
;nat_refresh=25

; async_process: bool: Process registrations asynchronously in their own thread
;async_process=enable


[sip-t]
; Controls the SIP-T parameter handling

; isup: bool: Build outgoing or decode incoming application/isup bodies
; If enabled an incoming application/isup body will be decoded and added to
;  the engine message issued by the receiving channel
; If the channel needs to add more then one body to an outgoing message, a
;  multipart/mixed body will be attached to the message
; Defaults to disable
;isup=disable


[codecs]
; This section allows to individually enable or disable the codecs

; default: bool: Enable all unlisted codecs by default if a transcoder exists
;default=enable

; mulaw: bool: Companded-only G711 mu-law (PCMU/8000)
;mulaw=default

; alaw: bool: Companded-only G711 a-law (PCMU/8000)
;alaw=default

; gsm: bool: European GSM 06.10 (GSM/8000)
;gsm=default

; lpc10: bool: Linear Prediction Codec (LPC/8000)
;lpc10=default

; ilbc: bool: Internet Low Bandwidth Codec (iLBC/8000)
;ilbc=default

; amr: bool: Adaptive Multi-Rate 3GPP (AMR/8000)
;amr=default

; slin: bool: Signed Linear 16-bit uncompressed (L16/8000)
;slin=default

; g723: bool: ITU G.723 all variations (G723/8000)
;g723=default

; g726: bool: ITU G.726 32-bit (G726-32/8000)
;g726=default

; g728: bool: ITU G.728 all variations (G728/8000)
;g728=default

; g729: bool: ITU G.729 all variations (G729/8000)
;g729=default

; g729_annexb: bool: G.729 Annex B (VAD) support default (if not in SDP)
; NOTE: RFC 3555 specifies the default should be yes
;g729_annexb=no

; amr_octet: bool: Octet aligned AMR RTP payload default (if not in SDP)
; NOTE: RFC 4867 (and older 3267) specifies the default is bandwidth efficient
;amr_octet=no


[methods]
; Use this section to allow server processing of various SIP methods by
;  handling Yate messages.
; Each line has to be of the form:
;  methodname=boolean
; You must use lower case method names. The boolean value defaults to
;  true and allows automatically challenging the requests for authentication


[hacks]
; This section holds the dirty stuff required to work with some broken
;  implementations
;
; ilbc_forced: string: Format to force as iLBC, can be: ilbc20 or ilbc30
;ilbc_forced=
;
; ilbc_default: string: Format to use for iLBC when packetization is unknown
;ilbc_default=ilbc30

; g729_annexb: bool: Force G.729 Annex B support when parsing the SDP
;g729_annexb=

; ignore_missing_ack: bool: Ignore missing ACK on INVITE, don't drop the calls
;ignore_missing_ack=no

; 1xx_change_formats: bool: Provisional messages can change the formats list
;1xx_change_formats=yes

; ignore_sdp_port: bool: Ignore SDP changes if only the port is different
; This allows preserving the local RTP session and port
;ignore_sdp_port=no