menu.sip_settings = SIP Settings
module.sip_settings = SIP Settings
sip_settings.accept_outofcall_message = Allow Msg Request
sip_settings.accept_outofcall_message.tooltip = Disable this option to reject all MESSAGE requests outside of a call. By default, this option is enabled. When enabled, MESSAGE requests are passed in to the dial-plan.
sip_settings.allowguest = Allow Guest
sip_settings.allowguest.tooltip = Whether to allow guest calls.
sip_settings.allowtransfer = Allow Transfer
sip_settings.allowtransfer.tooltip = Disable all transfers (unless enabled in peers or users). Default is enabled. The Dial() options 't' and 'T' are not related as to whether SIP transfers are allowed or not.
sip_settings.alwaysauthreject = Always Reject
sip_settings.alwaysauthreject.tooltip = When an incoming INVITE or REGISTER is to be rejected, for any reason, always reject with an identical response. This reduces the ability of an attacker to scan for valid SIP usernames.
sip_settings.authfailureevents = Failure Events
sip_settings.authfailureevents.tooltip = Generate manager peer-status events when peer can't authenticate with Asterisk. Peer-status will be rejected.
sip_settings.autoframing = Auto Framing
sip_settings.autoframing.tooltip = Set packetization based on the remote endpoint's (ptime) preferences.
sip_settings.available_codecs = Available Codecs
sip_settings.bindaddr = Bind Address
sip_settings.bindaddr.tooltip = IP address and optional port to bind for this transport (0.0.0.0 binds to all).
sip_settings.bindport = Bind Port
sip_settings.codecs = Codecs
sip_settings.custom = Custom
sip_settings.custom_options = Custom Options
sip_settings.defaultexpiry = Default Expiry
sip_settings.defaultexpiry.tooltip = Default length of incoming/outgoing registration.
sip_settings.dumphistory = Dump History
sip_settings.dumphistory.tooltip = Dump SIP history at end of SIP dialogue SIP history is output to the DEBUG logging channel.
sip_settings.dynamic_exclude_static = Disallow Dynamic Hosts
sip_settings.dynamic_exclude_static.tooltip = Disallow all dynamic hosts from registering as any IP address used for statically defined hosts. This helps avoid the configuration error of allowing your users to register at the same address as a SIP provider.
sip_settings.externaddr = External Address
sip_settings.externaddr.tooltip = Specifies a static address[:port] to be used in SIP and SDP messages.
sip_settings.externhost = External Host
sip_settings.externhost.tooltip = Alternatively you can specify an external host, and Asterisk will perform DNS queries periodically. Not recommended for production environments, Use %s instead.
sip_settings.externrefresh = External Refresh
sip_settings.externrefresh.tooltip = How often to refresh %s if used (value in seconds).
sip_settings.faxdetect = Fax Detect
sip_settings.faxdetect.tooltip = Check to enable both CNG and T.38 detection. Default 'no'.
sip_settings.fax_settings = Fax Settings
sip_settings.g726nonstandard = G726-32 Audio
sip_settings.g726nonstandard.tooltip = If the peer negotiates G726-32 audio, use AAL2 packing order instead of RFC3551 packing order (this is required for Sipura and Grandstream ATAs, among others).
sip_settings.general = General
sip_settings.ip = IP Address
sip_settings.ip.invalid = Local Networks must have a valid IP address.
sip_settings.jbenabled = Enabled
sip_settings.jbenabled.tooltip = Enabled/Disabled the use of a jitter-buffer on the receiving side of a SIP channel.
sip_settings.jbforce = Force
sip_settings.jbforce.tooltip = Forces the use of a jitter-buffer on the receive side of a SIP channel.
sip_settings.jbimpl = Jitterbuffer implementation
sip_settings.jbimpl.adaptive = Adaptive
sip_settings.jbimpl.adaptive.tooltip = With variable size, actually the new jb of IAX2.
sip_settings.jbimpl.fixed = Fixed
sip_settings.jbimpl.fixed.tooltip = With size always equals to jb-max-size.
sip_settings.jbimpl.tooltip = Used on the receiving side of a SIP channel. Two implementations are currently available = Adaptive and Fixed.
sip_settings.jbmaxsize = Max Size
sip_settings.jbmaxsize.tooltip = Max length of the jitter-buffer in milliseconds.
sip_settings.jbresyncthreshold = Resynchronized
sip_settings.jbresyncthreshold.tooltip = Jump in the frame timestamps over which the jitter-buffer is resynchronized.
sip_settings.jbtargetextra = Target Extra
sip_settings.jbtargetextra.tooltip = This option only affects the jb when 'jbimpl = adaptive' is set. The option represents the number of milliseconds by which the new jitter buffer will pad its size.
sip_settings.jitter_buffer = Jitter Buffer
sip_settings.language = Language
sip_settings.language.tooltip = Default language setting for all users/peers. This may also be set for individual users/peers.
sip_settings.localNetwork = Local Networks
sip_settings.maxcallbitrate = Max Call BitRate
sip_settings.maxcallbitrate.tooltip = Maximum bitrate for video calls (default 384 kb/s).
sip_settings.maxexpiry = Max Expiry
sip_settings.maxexpiry.tooltip = Maximum allowed time of incoming registrations (seconds).
sip_settings.minexpiry = Min Expiry
sip_settings.minexpiry.tooltip = Minimum length of registrations (default 60).
sip_settings.mwiexpiry = MWI Expiry
sip_settings.mwiexpiry.tooltip = Expiry time for outgoing MWI subscriptions.
sip_settings.network = Network
sip_settings.tcp_enable = TCP Enable
sip_settings.tcp_enable.tooltip = Use TCP
sip_settings.tcp_bindaddr = TCP Bind Address
sip_settings.tcp_bindaddr.tooltip = IP address and optional port to bind for this transport.
sip_settings.addr = Address
sip_settings.port = Port
sip_settings.tls_enable = Enable TLS
sip_settings.tls_enable.tooltip = Use TLS
sip_settings.tls_bindaddr = TLS Bind Address
sip_settings.tls_bindaddr.tooltip = IP address and optional port to bind for this transport.
sip_settings.tls_dont_verify = TLS Do Not Verify
sip_settings.tls_dont_verify.tooltip = If set to yes, don't verify the servers certificate when acting as a client.
sip_settings.tls_certificate_id = TLS Certificate
sip_settings.tls_certificate_id.tooltip = The server's certificate file.
sip_settings.nat = NAT
sip_settings.netmask = Network Mask
sip_settings.netmask.empty = Local Networks require both an IP address and netmask.
sip_settings.netmask.invalid = Local Networks must have a valid netmask.
sip_settings.notification_settings = Notification Settings
sip_settings.notifycid = Notify CID
sip_settings.notifycid.tooltip = Control whether caller ID information is sent along with dialog-info+xml notifications (supported by snom phones).
sip_settings.notifyhold = Notify Hold
sip_settings.notifyhold.tooltip = Notify subscriptions on HOLD state (default: no).
sip_settings.notifyringing = Notify Ringing
sip_settings.notifyringing.tooltip = Control whether subscriptions already INUSE get sent RINGING when another call is sent (default: yes).
sip_settings.others = Others
sip_settings.param = Parameter
sip_settings.preferred_codec_only = Preferred Codec Only
sip_settings.preferred_codec_only.tooltip = Respond to a SIP invite with the single most preferred codec rather than advertising all joint codec capabilities. This limits the other side's codec choice to exactly what we prefer.
sip_settings.qualifyfreq = Qualify Frequency
sip_settings.qualifyfreq.tooltip = How often to check for the host to be up in seconds and reported in milliseconds with sip show settings.
sip_settings.recordhistory = Record History
sip_settings.recordhistory.tooltip = Turn On/Off Record SIP history.
sip_settings.registration_settings = Registration Settings
sip_settings.rtpholdtimeout = RTP Hold Timeout
sip_settings.rtpholdtimeout.tooltip = Terminate call if %d seconds of no RTP or RTCP activity on the audio channel.
sip_settings.rtptimeout = RTP Timeout
sip_settings.rtptimeout.tooltip = Terminate call if %d seconds of no RTP or RTCP activity on the audio channel.
sip_settings.rtp_timers = RTP Timers
sip_settings.security = Security
sip_settings.selected_codecs = Selected Codecs
sip_settings.sipdebug = SIP Debug
sip_settings.sipdebug.tooltip = Turn On/Off SIP debugging, from the moment the channel loads this configuration.
sip_settings.sip_debugging = SIP Debugging
sip_settings.sip_qos = SIP QoS
sip_settings.srvlookup = SRV Lookups
sip_settings.srvlookup.tooltip = Enabled/Disabled DNS SRV lookups on outbound calls.
sip_settings.t38pt_udptl = T.38 Fax Pass-Through
sip_settings.t38pt_udptl.tooltip = Enables T.38 with FEC error correction and overrides the other endpoints provided value to assume we can send 400 byte T.38 FAX packets to it.
sip_settings.tonezone = Tone Zone
sip_settings.tonezone.tooltip = Default tone zone for all users/peers.
sip_settings.tos_audio = Audio TOS
sip_settings.tos_audio.tooltip = Sets the TOS for RTP audio packets.
sip_settings.tos_sip = SIP TOS
sip_settings.tos_sip.tooltip = Sets the TOS for SIP packets.
sip_settings.tos_video = Video TOS
sip_settings.tos_video.tooltip = Sets the TOS for RTP video packets.
sip_settings.value = Value
sip_settings.video_settings = Video Settings
sip_settings.video_support = Video Support
sip_settings.video_support.tooltip = Turn on/off support for SIP video. You need to turn this on in this section to get any video support at all.
sip_settings.websocket_enabled = Enable Websocket
sip_settings.websocket_enabled.tooltip =
	Set to "No" to prevent SIP channels from listening to websockets. This needed when using SIP and PJSIP websockets on
	the same system.
sip_settings.autodomain = Auto-Domain
sip_settings.autodomain.tooltip =
	This is an important setting with respect to SIP domains. When it is set to "no", Asterisk will only recognise
	domains that were explicitly defined or will simply not support SIP domains at all (if there were no explicitly defined domains).
	If you set it to "yes" please be aware that Asterisk will create a domain based on the external IP address of your
	firewall as specified in the "externip" parameter. This might represent a compromise on your SIP security.
	If you don't want a domain to be created based on "externip", then set to "no" and explicitly add domains
	for your local (internal) IP address and for any other domains you require.
sip_settings.allowexternaldomains = Allow External Domains
sip_settings.allowexternaldomains.tooltip = Tells Asterisk whether or not to allow SIP-to-SIP calls to non-local domains.
sip_settings.sip_domains = SIP Domains
sip_settings.sip_domains.info =
	When used, they provide enhanced security because registrations will only be accepted when they come from an IP phone
	(or other SIP client) that is using one of the recognised domains. When Asterisk knows the identity of all its local
	SIP domains, this allows a higher level of security in the routing of SIP-to-SIP calls too
sip_settings.domain = Domain
