tooltips['sip_general'].en[0] = "Context: Default context for incoming calls"; tooltips['sip_general'].en[1] = "Realm for digest authentication: Realm for digest authentication.defaults to \'asterisk\'. If you set a system name in asterisk.conf, it defaults to that system name. Realms MUST be globally unique according to RFC 3261. Set this to your host name or domain name"; tooltips['sip_general'].en[2] = "UDP Port to bind to: SIP standard port is 5060"; tooltips['sip_general'].en[3] = "IP address to bind to: 0.0.0.0 binds to all"; tooltips['sip_general'].en[4] = "Domain: Comma separated list of domains which Asterisk is responsible for"; tooltips['sip_general'].en[5] = "Allow guest calls: Enable guest calls."; tooltips['sip_general'].en[6] = "Overlap dialing support: Enable dialing support"; tooltips['sip_general'].en[7] = "Allow Transfers: Enable Transfers"; tooltips['sip_general'].en[8] = "Enable DNS SRV lookups (on outbound calls): Enable DNS SRV lookups on calls"; tooltips['sip_general'].en[9] = "Pedantic: Enable slow, pedantic checking of Call-ID:s, multiline SIP headers and URI-encoded headers"; tooltips['sip_general'].en[10] = " Type of Service: "; tooltips['sip_general'].en[11] = " TOS for Signalling packets: Sets Type of Service for SIP packets"; tooltips['sip_general'].en[12] = "TOS for RTP audio packets: Sets Type of Service for RTP audio packets"; tooltips['sip_general'].en[13] = "TOS for RTP video packets: Sets Type of Service for RTP video packets"; tooltips['sip_general'].en[14] = " Max Registration/Subscription Time: Maximum duration (in seconds) of incoming registration/subscriptions we allow. Default 3600 seconds."; tooltips['sip_general'].en[15] = " Min Registration/Subscription Time: Minimum duration (in seconds) of registrations/subscriptions. Default 60 seconds"; tooltips['sip_general'].en[16] = " Default Incoming/Outgoing Registration Time: Default duration (in seconds) of incoming/outoing registration"; tooltips['sip_general'].en[17] = " Min RoundtripTime (T1 Time): Minimum roundtrip time for messages to monitored hosts, Defaults to 100 ms"; tooltips['sip_general'].en[18] = " Override Notify MIME Type: Allow overriding of mime type in MWI NOTIFY"; tooltips['sip_general'].en[19] = " Time between MWI Checks: Default Time between Mailbox checks for peers"; tooltips['sip_general'].en[20] = " Music On Hold Interpret: This option specifies a preference for which music on hold class this channel should listen to when put on hold if the music class has not been set on the channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer channel putting this one on hold did not suggest a music class"; tooltips['sip_general'].en[21] = " Music On Hold Suggest: This option specifies which music on hold class to suggest to the peer channel when this channel places the peer on hold. It may be specified globally or on a per-user or per-peer basis."; tooltips['sip_general'].en[22] = " Language: Default language setting for all users/peers"; tooltips['sip_general'].en[23] = " Enable Relaxed DTMF: Relax dtmf handling"; tooltips['sip_general'].en[24] = " RTP TimeOut: Terminate call if 60 seconds of no RTP activity when we're not on hold "; tooltips['sip_general'].en[25] = "RTP HoldTimeOut: Terminate call if 300 seconds of no RTP activity when we're on hold (must be > rtptimeout)"; tooltips['sip_general'].en[26] = "Trust Remote Party ID: If Remote-Party-ID should be trusted"; tooltips['sip_general'].en[27] = "Send Remote Party ID:If Remote-Party-ID should be sent"; tooltips['sip_general'].en[28] = "Generate In-Band Ringing: If we should generate in-band ringing always use \'never\' to never use in-band signalling, even in cases where some buggy devices might not render it. Default: never"; tooltips['sip_general'].en[29] = "Server UserAgent: Allows you to change the user agent string"; tooltips['sip_general'].en[30] = "Allow Nonlocal Redirect:If checked, allows 302 or REDIR to non-local SIP address Note that promiscredir when redirects are made to the local system will cause loops since Asterisk is incapable of performing a \'hairpin\' call"; tooltips['sip_general'].en[31] = "Add 'user=phone' to URI: If checked, \'user=phone\' is added to uri that contains a valid phone number"; tooltips['sip_general'].en[32] = "DTMF Mode: Set default dtmfmode for sending DTMF. Default: rfc2833H"; tooltips['sip_general'].en[33] = "Send Compact SIP Headers: send compact sip headers"; tooltips['sip_general'].en[34] = " SIP Video Related:"; tooltips['sip_general'].en[35] = "Max Bitrate (kb/s):Maximum bitrate for video calls (default 384 kb/s)"; tooltips['sip_general'].en[36] = "Support for SIP Video:Turn on support for SIP video"; tooltips['sip_general'].en[37] = "Generate Manager Events: Generate manager events when sip ua performs events (e.g. hold)"; tooltips['sip_general'].en[38] = "Reject NonMatching Invites: When an incoming INVITE or REGISTER is to be rejected, for any reason, always reject with \'401 Unauthorized\' instead of letting the requester know whether there was a matching user or peer for their request"; tooltips['sip_general'].en[39] = "NonStandard G.726 Support: 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). This is contrary to the RFC3551 specification, the peer _should_ be negotiating AAL2-G726-32 instead"; tooltips['sip_general'].en[40] = " T.38 FAX Passthrough Support:"; tooltips['sip_general'].en[41] = "T.38 fax (UDPTL) Passthrough:Enables T.38 fax (UDPTL) passthrough on SIP to SIP calls"; tooltips['sip_general'].en[42] = "Sip Debugging:"; tooltips['sip_general'].en[43] = "Enable SIP debugging: Turn on SIP debugging by default "; tooltips['sip_general'].en[44] = "Record SIP History: Record SIP history by default"; tooltips['sip_general'].en[45] = "Dump SIP History: Dump SIP history at end of SIP dialogue"; tooltips['sip_general'].en[46] = "Status Notifications (Subscriptions):"; tooltips['sip_general'].en[47] = "Subscribe Context: Set a specific context for SUBSCRIBE requests. Useful to limit subscriptions to local extensions"; tooltips['sip_general'].en[48] = "Allow Subscribe: Support for subscriptions."; tooltips['sip_general'].en[49] = "Notify on Ringing: Notify subscriptions on RINGING state"; tooltips['sip_general'].en[50] = "Outbound SIP Registrations:"; tooltips['sip_general'].en[51] = "Register: Register as a SIP user agent to a SIP proxy (provider)"; tooltips['sip_general'].en[52] = "Register TimeOut: Retry registration calls at every \'x\' seconds (default 20)"; tooltips['sip_general'].en[53] = "Register Attempts: Number of registration attempts before we give up; 0 = continue foreverp"; tooltips['sip_general'].en[54] = "NAT Support:"; tooltips['sip_general'].en[55] = "Extern ip:Address that we're going to put in outbound SIP messages if we're behind a NAT"; tooltips['sip_general'].en[56] = "Extern Host:Alternatively you can specify an external host, and Asterisk will perform DNS queries periodically. Not recommended for production environments! Use externip instead"; tooltips['sip_general'].en[57] = "Extern Refresh: How often to refresh externhost if used. You may specify a local network in the field below"; tooltips['sip_general'].en[58] = "Local Network Address: \'192.168.0.0/255.255.0.0\' : All RFC 1918 addresses are local networks, \'10.0.0.0/255.0.0.0\' : Also RFC1918, \'172.16.0.0/12\' : Another RFC1918 with CIDR notation, \'169.254.0.0/255.255.0.0\' : Zero conf local network"; tooltips['sip_general'].en[59] = "NAT mode:Global NAT settings (Affects all peers and users); yes = Always ignore info and assume NAT; no = Use NAT mode only according to RFC3581; never = Never attempt NAT mode or RFC3581 support; route = Assume NAT, don't send rport"; tooltips['sip_general'].en[60] = "Allow RTP Reinvite:Asterisk by default tries to redirect the RTP media stream (audio) to go directly from the caller to the callee. Some devices do not support this (especially if one of them is behind a NAT)."; tooltips['sip_general'].en[61] = "Realtime Support:"; tooltips['sip_general'].en[62] = "Auto-Expire Friends: Auto-Expire friends created on the fly on the same schedule as if it had just registered? (yes|no|) If set to yes, when the registration expires, the friend will vanish from the configuration until requested again. If set to an integer, friends expire within this number of seconds instead of the registration interval."; tooltips['sip_general'].en[63] = "Cache Friends: Cache realtime friends by adding them to the internal list "; tooltips['sip_general'].en[64] = "Save SysName: Save systemname in realtime database at registration"; tooltips['sip_general'].en[65] = "Send Registry Updates: Send registry updates to database using realtime?"; tooltips['sip_general'].en[66] = "Ignore Expired Peers: Enabling this setting has two functions:

For non-realtime peers, when their registration expires, the information will _not_ be removed from memory or the Asterisk database if you attempt to place a call to the peer, the existing information will be used in spiteof it having expired

For realtime peers, when the peer is retrieved from realtime storage, the registration information will be used regardless of whether it has expired or not; if it expires while the realtime peer is still in memory (due to caching or other reasons), the information will not be removed from realtime storage

"; tooltips['sip_general'].en[67] = "SIP Domain Support:"; tooltips['sip_general'].en[68] = "Domain: List of \'allowed\' domains"; tooltips['sip_general'].en[69] = "From Domain: When making outbound SIP INVITEs to non-peers, use your primary domain \'identity\' for From: headers instead of just your IP address. This is to be polite and it may be a mandatory requirement for some destinations which do not have a prior account relationship with your server. "; tooltips['sip_general'].en[70] = "Auto Domain:Turn this on to have Asterisk add local host name and local IP to domain list."; tooltips['sip_general'].en[71] = "Allow External Domains:Allow requests for domains not serviced by this server "; tooltips['sip_general'].en[72] = "Allow External Invites: Enable INVITE and REFER to non-local domains "; tooltips['sip_general'].en[73] = " Jitter Buffer Configuration: "; tooltips['sip_general'].en[74] = "Enable Jitter Buffer: Enables the use of a jitterbuffer on the receiving side of a SIP channel."; tooltips['sip_general'].en[75] = "Force Jitter Buffer:Forces the use of a jitterbuffer on the receive side of a SIP channel "; tooltips['sip_general'].en[76] = "Log Frames:Enables jitterbuffer frame logging."; tooltips['sip_general'].en[77] = "Max Jitter Buffer: Max length of the jitterbuffer in milliseconds"; tooltips['sip_general'].en[78] = "Resync Threshold: Jump in the frame timestamps over which the jitterbuffer is resynchronized. Useful to improve the quality of the voice, with big jumps in/broken timestamps, usualy sent from exotic devices and programs. Defaults to 1000."; tooltips['sip_general'].en[79] = "Implementation: Jitterbuffer implementation, used on the receiving side of a SIP channel. Two implementations are currenlty available - \'fixed\' (with size always equals to jbmaxsize) and \'adaptive\' (with variable size, actually the new jb of IAX2)";