709 lines
27 KiB
Plaintext
709 lines
27 KiB
Plaintext
; This file configures the SIP channel
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;
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; NOTES on UDP listeners
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; - Address/port can be changed and reloaded
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; - If address/port is changed for an enabled listener this will be destroyed and recreated
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; - When an UDP listener is destroyed all channels using it will be dropped and
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; all lines using it will be unregistered
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; - If the only configured listener is 'general' this one will be the default one
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; - After initializing the module will find for a default transport:
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; 1: First search for a default listener whose name is not 'general'
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; 2: Use 'general' if no other listener is set to be the default
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[general]
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; This section sets global variables of the implementation
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; It also configures a listener named 'general' who is always enabled and set as default
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; UDP transport (if type is udp)
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; The listener is always processed before other 'listener ' sections
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; ipv6_support: boolean: Enable or disable IPv6 support
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; This parameter is applied on reload
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; This parameter is ignored if yate was not built with IPv6 support
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; Defaults to no
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;ipv6_support=no
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; type: keyword: Listener type
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; Allowed values:
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; udp: Build an UDP listener
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; tcp: Build a TCP listener
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; tls: Build a TLS listener (encrypted TCP)
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; Defaults to udp if missing or invalid
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;type=
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; default: boolean: Specifiy if this is the default transport to use when none specified
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; Defaults to yes (unlike the other listeners)
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;default=yes
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; addr: ipaddress: IP address to bind to
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; Leave it empty to listen on all available interfaces
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; IPv6: An interface name can be added at the end of the address to bind on a specific
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; interface. This is mandatory for Link Local addresses (e.g. addr=fe80::1%eth0)
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;addr=
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; port: integer: Port to bind to
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; Defaults to 5060 for UDP and TCP, 5061 for TLS listener
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;port=5060
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; ipv6: boolean: Listen on IPv6 address(es)
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; Listen will fail if IPv6 support is not enabled or not supported
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; Defaults to 'yes' if IP address is an IPv6 one or 'no' otherwise
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;ipv6=no
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; udp_force_bind: boolean: Try to use a random port if failed to bind on configured one (UDP only)
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; Defaults to yes
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;udp_force_bind=yes
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; rtp_localip: ipaddress: IP address to bind local RTP to
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; This parameter is applied on reload
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; TCP/TLS: this parameter is applied on reload for new connections only
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; RTP local IP address will default to bound IP address if not binding on all interfaces
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; Explicitly set it to empty string to avoid using bound IP address
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; IPv6: An interface name can be added at the end of the address to bind on a specific
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; interface. This is mandatory for Link Local addresses (e.g. addr=fe80::1%eth0)
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;rtp_localip=
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; nat_address: ipaddress: IP address to advertise in SDP, empty to use the local RTP
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; This parameter is applied on reload
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; Set this parameter when you know your RTP is behind a NAT
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;nat_address=
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; backlog: integer: Maximum length of the queue of pending connections
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; This parameter is ignored for UDP listener
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; Set it to 0 for system maximum
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; Defaults to 5 if missing or invalid
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;backlog=5
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; sslcontext: string: SSL context if this is an encrypted connection
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; Ignored for non TLS listener, required for TLS listener
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;sslcontext=
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; maxpkt: int: Maximum received UDP packet size, 524 to 65528, default 1500
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; This parameter is applied on reload and can be overridden in UDP listener sections
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;maxpkt=1500
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; buffer: int: Requested size of UDP socket's receive buffer, 0 to use default
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; This can be overridden in UDP listener sections
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;buffer=0
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; tcp_maxpkt: int: Maximum received TCP packet size, 524 to 65528, default 4096
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; This parameter is applied on reload and can be overridden in TCP/TLS listener sections
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; The parameter is not applied on reload for already created listeners or connections
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;tcp_maxpkt=4096
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; tcp_out_rtp_localip: ipaddress: IP address to bind local RTP to for outgoing
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; TCP connections, empty to guess best
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; This parameter is applied on reload for new connections only
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; IPv6: An interface name can be added at the end of the address to bind on a specific
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; interface. This is mandatory for Link Local addresses (e.g. addr=fe80::1%eth0)
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;tcp_out_rtp_localip=
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; thread: keyword: Default priority of the SIP handling threads
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; Can be one of: lowest, low, normal, high, highest
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; High priorities need superuser privileges on POSIX operating systems
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; Low priorities are not recommended except for debugging
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;thread=normal
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; role: string: Role to be set in messages sent by connections using this listener
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; This parameter is applied on reload
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;role=
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; floodevents: int: How many SIP events retrieved in a row trigger a flood warning and the drop mechanism
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; for INVITE/REGISTER/SUBSCRIBE/OPTIONS messages if the flood protection is on.
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; NOTE! The drop mechanism is separately activated by the floodprotection setting which is on by default. Also,
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; setting this parameter to 0 will disable the flood warning and protection.
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;floodevents=100
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; floodprotection: bool: Activate the drop mechanism for INVITE/REGISTER/SUBSCRIBE/OPTIONS messages when
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; the number of SIP events retrieved in a row exceeds the number set for floodevents setting.
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; Other messages, as well as reINVITEs, will be allowed.
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; NOTE! This mechanism is activated by default, to disable it configure this parameter to false.
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;floodprotection=on
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; congestion_retry: int: Value of Retry-After header to set in case of engine congestion
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; Valid values 10 - 600, default 30 seconds
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;congestion_retry=30
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; maxforwards: int: Default Max-Forwards header, used to avoid looping calls
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;maxforwards=20
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; useragent: string: String to set in User-Agent or Server headers
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;useragent=YATE/2.0.0
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; realm: string: Authentication realm to offer in authentication requests
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;realm=Yate
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; transfer: bool/keyword: Allow handling the REFER message to perform transfers
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; 'chan-disable': Enable global transfer flag (advertise REFER is supported)
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; REFER handling is disabled in call legs
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; REFER handling MUST be explicitly enabled per call leg when routing
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; Parameter: 'enable_refer' for outgoing leg, 'ienable_refer' for incoming leg
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;transfer=enable in server mode, disable in client mode
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; refer_update: boolean: Emit a call.update with 'operation'='transfer' when handling REFER
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; If enabled the message MUST succeed (must be handled and return value not '-' or 'error')
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; REFER will fail if call.update fails
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; The handler may set a 'transferred' parameter to boolean 'true' to stop further REFER processing
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; and notify success to implicit subscription
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;refer_update=no
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; registrar: bool: Allow the SIP module to receive registration requests
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; OBSOLETE - please use "enable" in section [registar]
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;registrar=enable in server mode, disable in client mode
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; options: bool: Build and send a default 200 answer to OPTIONS requests
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; OBSOLETE - please use "enable" in section [options]
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;options=enable
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; update: bool: Enable receiving UPDATE transactions (RFC 3311)
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;update=disable
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; prack: bool: Enable acknowledging provisional 1xx answers (RFC 3262)
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;prack=disable
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; info: bool: Accept incoming INFO messages
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;info=enable
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; fork: bool: Follow first forked 2xx answer on early dialogs
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; This parameter is applied on reload
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;fork=enable
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; fork_early: bool: Also follow forked 1xx on early dialogs
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; Ignored if fork following on 2xx is disabled
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; This parameter is applied on reload
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;fork_early=disable
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; progress: bool: Send an "183 Session Progress" just after successfull routing
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;progress=disable
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; generate: bool: Allow Yate messages to send arbitrary SIP client transactions
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;generate=disable
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; nat: bool: Enable automatic NAT support
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;nat=enable
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; ignorevia: bool: Ignore Via headers and send answer back to the source
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; This violates RFC 3261 but is required to support NAT over UDP transport.
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; This parameter can be overridden in listener sections
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;ignorevia=enable
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; lazy100: bool: Do not generate an initial "100 Trying" for non-INVITE
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; transactions unless a retransmission arrives before having a final answer
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; This parameter is applied on reload
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;lazy100=no
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; t1: int: Value of SIP T1 timer in milliseconds
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; This is the RTT Estimate and several other SIP timers are derived from it
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; Valid values are between 100 and 5000, outside range uses the default of 500
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; This parameter is applied on reload
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;t1=500
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; t4: int: Value of SIP T4 timer in milliseconds
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; This is the maximum message lifetime, several other SIP timers are derived from it
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; It is enforced to be at least 3 * T1
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; Valid values are between 1000 and 25000, outside range uses the default of 5000
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; This parameter is applied on reload
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;t4=5000
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; check_allow_info: bool: Check 'Allow' header in INVITE and OK for INFO support
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; If enabled and INFO is not supported the 'info' dtmf method will be disabled
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; This parameter can be overridden from routing by 'ocheck_allow_info' for outgoing call leg
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; and 'icheck_allow_info' for incoming call leg
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; This parameter is ignored if info method is not enabled
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; This parameter is applied on reload for new calls only
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;check_allow_info=yes
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; missing_allow_info: bool: The default value for dtmf info support if
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; 'check_allow_info' is enabled and the 'Allow' header is missing
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; This parameter can be overridden from routing by 'omissing_allow_info' for outgoing call leg
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; and 'imissing_allow_info' for incoming call leg
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; This parameter is applied on reload for new calls only
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;missing_allow_info=enable
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; dtmfmethods: string: Comma separated list of methods used to send DTMFs
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; Allowed values in list:
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; info: Use SIP INFO if initial transaction finished
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; rfc2833: Use RFC 2833 signals if remote party advertised support
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; inband: Send tones in audio stream
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; The methods will be used in the listed order
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; Defaults to 'rfc2833,info,inband' if missing or empty
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; Invalid values are ignored
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; E.g.
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; 'info,foo' leads to 'info'
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; 'foo,foo1' leads to 'rfc2833,info,inband'
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; This parameter can be overridden from routing by 'odtmfmethods' for outgoing call leg
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; and 'idtmfmethods' for incoming call leg
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; Also, this parameter can be overridden in chan.dtmf messages by a 'methods' parameter
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; NOTE:
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; When overridden from chan.dtmf an empty or invalid 'methods' parameter will be ignored
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; Methods indicated in chan.dtmf message will be intersected with channel capabilities
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; unless an explicit boolean true 'methods_override' parameter is present
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; This parameter is applied on reload for new calls only
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;dtmfmethods=rfc2833,info,inband
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; honor_dtmf_detect: bool: Honor DTMF detected method when sending DTMFs
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; If enabled the channel will try to send a DTMF using the same method as received
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; If the detected method is not enabled it won't be used
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; This parameter can be overridden from routing by 'ohonor_dtmf_detect' for outgoing call leg
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; and 'ihonor_dtmf_detect' for incoming call leg
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; This parameter is applied on reload for new calls only
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; Defaults to enable
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;honor_dtmf_detect=enable
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; rfc2833: bool: Offer RFC2833 telephone-event 8KHz by default
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; A numeric payload >= 96 can be provided
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;rfc2833=yes
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; rfc2833_RATE: bool: Offer RFC2833 telephone-event for specific rate (non 8KHz) by default
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; A numeric payload >= 96 can be provided
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; Supported rates (parameters): rfc2833_16000, rfc2833_32000
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;rfc2833_RATE=yes
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; privacy: bool: Process and generate privacy related SIP headers
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;privacy=disable
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; secure: bool: Generate and accept RFC 4568 security descriptors for SRTP
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;secure=disable
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; forward_sdp: bool: Include the raw SDP body to be used as-is for forwarding RTP
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;forward_sdp=disable
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; forward_gpmd: bool: Propagate GPMD even when not forwarding RTP
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;forward_gpmd=disable
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; rtp_start: bool: Start RTP when sending 200 on incoming instead of receiving ACK
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;rtp_start=disable
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; multi_ringing: bool: Accept provisional (1xx) messages even after 180 Ringing
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;multi_ringing=disable
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; refresh_nosdp: bool: Accept session refresh reINVITEs that lack a SDP offer
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;refresh_nosdp=enable
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; update_target: bool: Update dialog target from Contact in reINVITE
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;update_target=disable
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; update_verify: bool: Use a message to verify if we should accept a reINVITE when proxying media
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;update_verify=disable
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; preventive_bye: bool: If possible send a BYE besides CANCEL for unanswered calls
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;preventive_bye=enable
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; auth_foreign: bool: Attempt to authenticate nonces not generated locally
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; This parameter is applied on reload
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;auth_foreign=disable
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;auth_copy_headers: string: Comma separated list of headers to be copied in user.auth message
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; This parameter is applied on reload
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;auth_copy_headers=
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; body_encoding: keyword: Encoding used for received generic binary bodies
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; Can be one of: base64, hex, hexs, raw
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;body_encoding=base64
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; async_generic: bool: Process generic SIP messages asynchronously in their own thread
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;async_generic=enable
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; flags: int: Miscellaneous SIP engine flags for broken implementations
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; See SIPMessage::Flags and SIPMessage::complete() in the source for gory details
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;flags=0
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; autochangeparty: bool: Automatically change remote ip/port when a channel receives
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; a response or a new transaction from a different address
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; E.g. if an INVITE sent to 1.2.3.4:5060 receives OK from 1.2.3.4:5080 the ACK
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; (and subsequent transactions) will be sent to 1.2.3.4:5080
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; Defaults to disable
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; This parameter is applied on reload
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;autochangeparty=disable
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; change_party_2xx: bool: Change party when handling 2xx response to INVITE
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; Defaults to disable
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; This parameter is applied on reload
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; This parameter is ignored if 'autochangeparty' is disabled
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;change_party_2xx=disable
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; ssl_certificate_file: string: File containing client SSL certificate to present
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; This parameter is used for outgoing encrypted connections if a certificate
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; is requested by the server during SSL negotiation
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; The file path is relative to configuration path
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; This parameter is applied on reload
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;ssl_certificate_file=
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; ssl_key_file: string: Optional file containing the key of the certificate
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; set in ssl_certificate_file
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; The file path is relative to configuration path
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; The certificate file must contain the key if this parameter is empty
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; This parameter is applied on reload
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;ssl_key_file=
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; sip_req_trans_count: integer: The number of times to transmit a sip request
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; when retransmission is required (e.g. on non reliable transports)
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; This parameter is applied on reload
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; Minimum allowed value is 2, maximum allowed value is 10
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; Defaults to 4 if missing, invalid or out of bounds
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;sip_req_trans_count=4
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; sip_rsp_trans_count: integer: The number of times to transmit a final response
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; to a sip request when retransmission is required
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; Retransmission is required for all responses to INVITE requests on non reliable
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; transports or 2xx responses over reliable transports
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; This parameter is applied on reload
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; Minimum allowed value is 2, maximum allowed value is 10
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; Defaults to 5 if missing, invalid or out of bounds
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;sip_rsp_trans_count=5
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; maxchans: int: Maximum number of channels running at once
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; A value of 0 specifies that there is no limit enforced.
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; Defaults to the value set by the maxchans setting from yate.conf
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;maxchans=
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; printmsg: boolean: Print SIP messages to output
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; This parameter is applied on reload
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; Defaults to yes
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;printmsg=yes
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; tcp_idle: integer: Interval (in seconds) allowed for an incoming TCP connection
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; to stay idle (nothing sent/received)
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; This parameter is applied on reload for new connections only
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; It may be overridden in listener sections
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; Defaults to 120
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; Minimum allowed value is calculated from SIP 'B' timer (which is 64 * t1 timer value)
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; expressed in seconds using the following formula: B * 3 / 2 (46 seconds for T1 default value)
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; Maximum allwed value is 600
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;tcp_idle=120
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; tcp_keepalive: integer: Interval (in seconds) to send keepalive on outgoing TCP connections
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; Defaults to 'tcp_idle' value
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;tcp_keepalive=tcp_idle
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; tcp_keepalive_first: integer: Interval (in seconds) to send first keepalive on
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; outgoing TCP connections
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; If set this parameter must be less than 'tcp_keepalive'
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;tcp_keepalive_first=0
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; ssdp_prefix: string: Prefix to use when handling SDP session level parameters
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; This parameter is used when setting them in yate messages or handling them from there
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; This parameter is applied on reload
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; Prefix used to set parsed SDP: <ssdp_prefix>_ (default: ssdp_)
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; Prefix used to update from yate messages: o<ssdp_prefix>_ (default: ossdp_)
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; When updated from yate messages the prefix must be set in 'ossdp-prefix' message parameter
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;ssdp_prefix=ssdp
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; initial_headers: boolean: Put all headers from initial requests in yate message
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; Handled for incoming channel preroute, user (un)register and messages sent on SIP
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; requests received outside a dialog
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; This parameter is applied on reload
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;initial_headers=no
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; reinvite_wait_initial: boolean: Wait for answered initial transaction termination when need to send
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; a re-INVITE and initial transaction was not terminated
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; Applicable for the inbound call leg
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; This parameter is handled when answer (200 OK) was sent to initial transaction
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; If enabled the module will not send an UPDATE even if supported by remote
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; This parameter can be overridden from routing
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; This parameter is applied on reload
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;reinvite_wait_initial=no
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; mixed_provisional: boolean: Accept mixed (non)reliable provisional responses to initial transaction
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; When enabled (default) the dialog will accept non reliable provisional messages
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; after receiving a reliable one
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; This parameter can be overridden from routing
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; This parameter is applied on reload
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;mixed_provisional=yes
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; warn_bind_fail_delay: integer/string: Delay failed to bind debug message
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; This parameter may be used when listener is going to bind on an IP which may become
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; available later
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; Values:
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; integer: Delay in milliseconds. Interval: 500..60000
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; 'start': Delay until engine starts (engine.start message is handled by module)
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;warn_bind_fail_delay=0
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; warn_no_default_udp_transport: boolean: Warn if there is not default UDP transport
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; This parameter is applied on reload
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;warn_no_default_udp_transport=yes
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; capture_filter: boolean. Enable global HEP3 capture of SIP packets
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; NOTE: This setting can be overridden by listener settings or by account settings
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; in case of outgoing TCP connections.
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; This setting applies on reload.
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;capture_filter=false
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; capture_agent: string, mandatory if capture_filter is set to true. Name of capture agent.
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; NOTE: This setting can be overridden by listener settings or by account settings
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; in case of outgoing TCP connections.
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; This is for internal tracking.
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;capture_agent=
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; capture_server: string, mandatory if capture_filter is set to true.
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; Name of HEP3 server where to send packets. The server with this name must be configured
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; in HEP3 module configuration.
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; NOTE: This setting can be overridden by listener settings or by account settings
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; in case of outgoing TCP connections.
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;capture_server=
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; capture_compress: boolean. Set to true to compress captured packets.
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; If not set, it will use the HEP server configuration 'compress' configured value.
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; NOTE: This setting can be overridden by listener settings or by account settings
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; in case of outgoing TCP connections.
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;capture_compress=false
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[options]
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; Controls the behaviour for SIP options retrieval
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; enable: bool: Allow the SIP module to receive OPTIONS requests
|
|
;enable=yes
|
|
|
|
|
|
[registrar]
|
|
; Controls the behaviour when acting as registrar
|
|
|
|
; enable: bool: Allow the SIP module to receive registration requests
|
|
;enable=yes in server mode, no in client mode
|
|
|
|
; 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
|
|
|
|
|
|
[message]
|
|
; Controls the behaviour for SIP messaging
|
|
|
|
; enable: bool: Allow the SIP module to receive MESSAGE requests
|
|
;enable=no
|
|
|
|
; auth_required: bool: Automatically challenge all senders for authentication
|
|
;auth_required=enable
|
|
|
|
; async_process: bool: Process SIP MESSAGE 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
|
|
|
|
; default_stereo: bool: Enable all unlisted stereo audio codecs by default if a transcoder exists
|
|
;default_stereo=no
|
|
|
|
; mulaw: bool: Companded-only G711 mu-law (PCMU/8000)
|
|
;mulaw=default
|
|
|
|
; alaw: bool: Companded-only G711 a-law (PCMU/8000)
|
|
;alaw=default
|
|
|
|
; clearmode: bool: Transparent 64kbit/s B channel (RFC4040)
|
|
;clearmode=default
|
|
|
|
; gsm: bool: European GSM 06.10 (GSM/8000)
|
|
;gsm=default
|
|
|
|
; gsm-efr: bool: European GSM 06.60 (GSM-EFR/8000)
|
|
;gsm-efr=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
|
|
|
|
; 2*mulaw: bool: Stereo Companded-only G711 mu-law (PCMU/8000)
|
|
;2*mulaw=default_stereo
|
|
|
|
; 2*alaw: bool: Stereo Companded-only G711 a-law (PCMU/8000)
|
|
;2*alaw=default_stereo
|
|
|
|
; 2*slin: bool: Stereo Signed Linear 16-bit uncompressed (L16/8000)
|
|
;2*slin=default_stereo
|
|
|
|
|
|
[methods]
|
|
; Use this section to allow server processing of various SIP methods by
|
|
; handling Yate messages with name "sip.methodname".
|
|
; 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
|
|
;
|
|
; Example for accepting SECRET with authentication and MESSAGE without:
|
|
; secret=yes
|
|
; message=no
|
|
|
|
|
|
[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
|
|
|
|
; sdp_implicit: bool: Assume application/sdp is supported if no Accept is present
|
|
;sdp_implicit=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
|
|
|
|
; ignore_sdp_addr: bool: Ignore SDP changes if only the address is different
|
|
; This allows preserving the local RTP session and port
|
|
;ignore_sdp_addr=no
|
|
|
|
|
|
;[listener name]
|
|
; This section configures a listener named 'name'
|
|
; If a listener named 'general' is configured (section 'listener general' exists) no listener
|
|
; will be setup from the 'general' section.
|
|
; The following parameters can be overridden from 'general' section:
|
|
; UDP: maxpkt, buffer
|
|
; TCP/TLS: tcp_maxpkt
|
|
; All: warn_bind_fail_delay
|
|
|
|
; type: keyword: Listener type
|
|
; Allowed values:
|
|
; udp: Build an UDP listener
|
|
; tcp: Build a TCP listener
|
|
; tls: Build a TLS listener (encrypted TCP)
|
|
; Defaults to udp if missing or invalid
|
|
;type=
|
|
|
|
; enable: boolean: Enable or disable this listener
|
|
; This parameter is applied on reload and defaults to yes
|
|
;enable=yes
|
|
|
|
; default: boolean: UDP only: specifiy if this is the default transport to use when none specified
|
|
; Defaults to no
|
|
;default=no
|
|
|
|
; udp_force_bind: boolean: UDP only: try to use a random port if failed to bind on configured one (UDP only)
|
|
; Defaults to yes
|
|
;udp_force_bind=yes
|
|
|
|
; addr: ipaddress: IP address to bind to
|
|
; Leave it empty to listen on all available interfaces
|
|
; IPv6: An interface name can be added at the end of the address to bind on a specific
|
|
; interface. This is mandatory for Link Local addresses (e.g. addr=fe80::1%eth0)
|
|
;addr=
|
|
|
|
; port: integer: Port to bind to
|
|
; Defaults to 5060 for UDP and TCP, 5061 for TLS listeners
|
|
;port=
|
|
|
|
; ipv6: boolean: Listen on IPv6 address(es)
|
|
; Listen will fail if IPv6 support is not enabled or not supported
|
|
; Defaults to 'yes' if IP address is an IPv6 one or 'no' otherwise
|
|
;ipv6=no
|
|
|
|
; rtp_localip: ipaddress: IP address to bind local RTP to
|
|
; This parameter is applied on reload
|
|
; TCP/TLS: this parameter is applied on reload for new connections only
|
|
; RTP local IP address will default to bound IP address if not binding on all interfaces
|
|
; Explicitly set it to empty string to avoid using bound IP address
|
|
; IPv6: An interface name can be added at the end of the address to bind on a specific
|
|
; interface. This is mandatory for Link Local addresses (e.g. addr=fe80::1%eth0)
|
|
;rtp_localip=
|
|
|
|
; backlog: integer: Maximum length of the queue of pending connections
|
|
; This parameter is ignored for UDP listeners
|
|
; Set it to 0 for system maximum
|
|
; Defaults to 5 if missing or invalid
|
|
;backlog=5
|
|
|
|
; sslcontext: string: SSL context if this is an encrypted connection
|
|
; Ignored for non TLS listeners, required for TLS listeners
|
|
;sslcontext=
|
|
|
|
; thread: keyword: Listener thread priority
|
|
; Can be one of: lowest, low, normal, high, highest
|
|
; High priorities need superuser privileges on POSIX operating systems
|
|
; Low priorities are not recommended except for debugging
|
|
;thread=normal
|
|
|
|
; role: string: Role to be set in messages sent by connections using this listener
|
|
; This parameter is applied on reload
|
|
;role=
|
|
|
|
; capture_filter: boolean. Enable HEP3 capture of packets on this listener.
|
|
; NOTE: for outgoing TCP connections, these settings must be made in accfile.conf.
|
|
; This setting applies on reload.
|
|
;capture_filter=false
|
|
|
|
; capture_agent: string, mandatory if capture_filter is set to true. Name of capture agent
|
|
; This is for internal tracking.
|
|
;capture_agent=
|
|
|
|
; capture_server: string, mandatory if capture_filter is set to true.
|
|
; Name of HEP3 server where to send packets. The server with this name must be configured
|
|
; in HEP3 module configuration
|
|
;capture_server=
|
|
|
|
; capture_compress: boolean. Set to true to compress captured packets.
|
|
; If not set, it will use the HEP server configuration 'compress' configured value
|
|
;capture_compress=false
|