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pabx-app/admin/khomp.tpl

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Commit inicial versao-1.7.11.4
2 years ago
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
;;;;;;;;;;;;;; This is the configuration file for the Khomp ;;;;;;;;;;;;;;;
;;;;;;;;;;;;;;;;;;;;;;;;;;; channel version 4.2 ;;;;;;;;;;;;;;;;;;;;;;;;;;;
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
;;;;
;;;; Section for main general configurations
;;;;
[globals]
;;
;; Enable/disable the dial string compatibility with DAHDI in Dial string form,
;; if enabled all Khomp dial will use the dial string like DAHDI.
;;
;; possible syntaxes:
;;
;; Dial(Khomp/<channel#>[c|r<cadence#>|d]/<extension>)
;; Dial(Khomp/(g|G|r|R)<group#>[c|r<cadence#>|d]/<extension>)
;; Dial(Khomp/i<span#>/<extension>)
;;
;; (default = no)
; dial-string-like-dahdi = no
;;
;; Defines the behaviour of the module load function if communication to
;; Khomp boards and/or devices cannot be started. Possible values:
;;
;; failure - Return failure and halts Asterisk execution.
;; skip - Skips loading this module.
;;
;; (default = failure)
; load-error = failure
;;
;; Defines the incoming context for calls on E1 channels. Some wildcards are
;; accepted, and described in the bottom of this section.
;;
;; (default = khomp-DD-LL)
; context-digital = khomp-DD-LL
;;
;; Defines the incoming context for calls on FXS channels. Some wildcards are
;; accepted, and described in the bottom of this section.
;;
;; (default = khomp-DD-CC)
; context-fxs = khomp-DD-CC
;;
;; Defines the "alternative" context for calls on FXS channels, which will be
;; checked if the main context does not match for a call. Some wildcards are
;; accepted, and described in the bottom of this section.
;;
;; (default = khomp-DD)
; context-fxs-alt = khomp-DD
;;
;; Defines the incoming context for calls on FXO channels. Some wildcards are
;; accepted, and described in the bottom of this section.
;;
;; (default = khomp-DD-CC)
; context-fxo = khomp-DD-CC
;;
;; Defines the "alternative" context for calls on FXO channels, which will be
;; checked if the main context does not match for a call. Some wildcards are
;; accepted, and described in the bottom of this section.
;;
;; (default = khomp-DD)
; context-fxo-alt = khomp-DD
;;
;; Defines the incoming context for calls on GSM channels. Some wildcards are
;; accepted, and described in the bottom of this section.
;;
;; (default = khomp-DD-CC)
; context-gsm-call = khomp-DD-CC
;;
;; Defines the "alternative" context for calls on GSM channels, which will be
;; checked if the main context does not match for a call. Some wildcards are
;; accepted, and described in the bottom of this section.
;;
;; (default = khomp-DD)
; context-gsm-call-alt = khomp-DD
;;
;; Defines the incoming context for messages on GSM channels. Some wildcards are
;; accepted, and described in the bottom of this section.
;;
;; (default = khomp-sms-DD-CC)
; context-gsm-sms = khomp-sms-DD-CC
;;
;; Defines the incoming context for execution of a waiting call on GSM channels,
;; used when some KGSM channel is already handling calls and another incoming
;; call arrives in the same channel.
;;
;; The channel type used for execution is a special Khomp_Wait channel, which
;; does not have any audio nor can be answered/dialed. Some wildcards are
;; accepted on the context name, and described in the bottom of this section.
;;
;; Use "none" for disabling this feature.
;;
;; (default = khomp-wait-DD-CC)
; context-gsm-wait = khomp-wait-DD-CC
;;
;; Defines the incoming context for calls on Passive Record boards (KPR). Some
;; wildcards are accepted, and described in the bottom of this section.
;;
;; (default = khomp-DD-CC)
; context-pr = khomp-DD-CC
;;
;; Set the logging of messages to console. Possible values (to set more than one,
;; separate the values with comma):
;;
;; errors -- Error messages, when something goes really wrong.
;; warnings -- Warnings, used when something might not be going as expected.
;; messages -- Generic messages, used to indicate some information.
;; events -- Show received K3L events as console messages.
;; commands -- Show sent K3L commands as console messages.
;; audio -- Enable messages for K3L audio events (very verbose!).
;; modem -- Enable messages for data received from KGSM modems.
;; link -- Enable logging of link status changes.
;; standard -- Special, enable default messages (RECOMMENDED).
;; all -- Special, enable ALL messages (should not be used naively).
;;
;; (default = standard)
; log-to-console = standard
;;
;; Set the logging of messages to disk. Possible values (to set more than one,
;; separate the values with comma):
;;
;; errors -- Error messages, when something goes really wrong.
;; warnings -- Warnings, used when something might not be going as expected.
;; messages -- Generic messages, used to indicate some information.
;; events -- Record received K3L events as log messages.
;; commands -- Record sent K3L commands as log messages.
;; audio -- Enable messages for K3L audio events (very verbose!).
;; modem -- Enable messages for data received from KGSM modems.
;; link -- Enable logging of link status changes.
;; functions -- Enable debugging for functions. Should not be used naively!
;; threads -- Enable debugging for threads. Should not be used naively!
;; locks -- Enable debugging for locks. Should not be used naively!
;; stream-notice -- Enable debugging for stream changes, failures, overruns, underruns.
;; Should not be used naively!
;; stream-debug -- Enable debugging for streams (DO NOT USE THIS!).
;;
;; standard -- Special, enable default messages (RECOMMENDED).
;; debugging -- Special, enable debug messages (should not be used naively).
;; all -- Special, enable ALL messages (DO NOT USE THIS!).
;;
;; (default = standard)
; log-to-disk = standard
;;
;; Sets the global orig (CALLERID) base for FXS boards. This number is added
;; to a sequencial number, which is incremented for each FXS board and FXS
;; channel in the system.
;;
;; For more example of how to use this option, see channel README file,
;; section 'Opcoes do application Dial', item 'Dial(Khomp/r304)'.
;;
;; (default = 00)
; fxs-global-orig = 00
;;
;; Defines the way audio RX synchorization is handled.
;;
;; Possible values:
;;
;; softtimer-thread - Creates a thread which signals all channels for
;; pending data each (audio-packet-length/8) ms.
;; Default mode of operation, works on virtually any
;; system using default configuration.
;;
;; softtimer-multi-thread - Same as above, but creates one thread for each
;; device (board or module).
;;
;; softtimer-kernel - Available for kernel 2.6.22+ and {g,eg}libc 2.7+,
;; uses timerfd syscalls for creating timers for
;; synchronizing the audio stream reported to Asterisk.
;;
;; auto (default) - Selects best method for synchronization, currently just
;; sets the option for 'softtimer-thread'.
; audio-rx-sync = auto
;;
;; Defines the length (in BYTES) for A-Law audio packets sent to Asterisk. Can
;; be used to change the packet size for applications and/or channels that need
;; fixed-sized data at a specific size (e.g.: iaxmodem, some SIP devices).
;; DO NOT CHANGE THIS OPTION NAIVELY, AS IT MAY RESULT IN UNDESIRED DELAY DUE
;; TO EXTRA BUFFERING NEEDED FOR PACKET SIZE CONVERSION.
;;
;; Some useful values:
;;
;; 16 ms => 128 bytes (default, no extra buffering needed)
;;
;; 10 ms => 80 bytes (minimum)
;; 20 ms => 160 bytes
;; 30 ms => 240 bytes (maximum)
;;
;; Minimum step is 8 bytes (1ms).
; audio-packet-length = 128
;;
;; Defines if the activation and deactivation of Kommuter is done automatically by
;; the channel, or manually by the user. Possible values:
;;
;; auto -- Khomp channel driver starts all kommuter devices at initialization
;; and stops them if the module is unloaded.
;;
;; manual -- The user must explicity call the CLI command < khomp kommuter on/off >,
;; that starts or stops the kommuter devices connected to this machine.
;;
;; (default = auto)
; kommuter-activation = auto
;;
;; Defines the default value for the Kommuter watchdog in seconds.
;; All kommuters conected to this machine will be initialized with this value,
;; and will commute the links after reaching this time with no response of the channel.
;; The minimum is 0 (will never commute the links), and maximum value is 255 seconds.
;;
;; (default = 10)
; kommuter-timeout = 10
;;
;; When adjusted to some DTMF digit sequence, it will define those as the digits
;; to be used to initialize a call transfer using PBX-to-PBX signaling.
;;
;; (default = empty)
; user-transfer-digits =
;;;;;;;;;;;;;;;;;;;;;; CONTEXTS WILDCARDS ;;;;;;;;;;;;;;;;;;;;;;
;;
;; For incoming contexts, you can use the following wildcards:
;;
;; 'DD' (device number): the sequence number of the board on the
;; system (can be checked using "khomp summary", valid for all
;; board models);
;;
;; 'LL' (link number): the sequence number of link where the call
;; has arrived on the board. valid only for E1 boards.
;;
;; 'SSSS' (serial number): the serial number of the board (can
;; be checked using "khomp summary", and it's valid for all
;; board models);
;;
;; 'CC' (channel number): the channel number on which the call
;; or message has arrived (valid for FXO, FXS and GSM boards);
;;
;; 'CCC' (channel number): same as above, but valid only for E1
;; channels;
;;
;;;;
;;;; Section for main configurations about channels.
;;;;
;;;; In this section you can apply any of the configurations below to a specific
;;;; channel selection, by using the customized section [channels-<string>],
;;;; where <string> is a custom dialstring, as the one used on the Dial() application.
;;;; The other channels will have the configurations defined on the [channels]
;;;; section, while the specific channels defined here will have these options
;;;; overwritten. For more information about the dialstrings allowed
;;;; check the Khomp channel driver manual.
;;;; ex: [channels-b0l0]
;;;; drop-collect-call = yes
;;;;
;;;; [channels-b1c0+b1c1]
;;;; input-volume = +2
;;;; output-volume = +2
[channels]
;;
;; Enable/disable generalized echo cancellation in the channel, for calls
;; passing inside Asterisk (disabled for bridged calls).
;;
;; (default = yes)
; echo-canceller = yes
;;
;; Enable/disable AGC (auto gain control). Should be used carefully, as it
;; can make line noise really loud.
;;
;; (default = no)
; auto-gain-control = no
;;
;; Enable/disable sending DTMFs out-band as a way to pass detected DTMFs to
;; Asterisk. Needed if Asterisk generates digits for us in Dial (option 'D')
;; or is being used for IVR ("URA", in pt_BR).
;;
;; (default = yes)
; out-of-band-dtmfs = yes
;;
;; Enable/disable DTMF suppression. WARNING: If you disable this, DTMFs will
;; not be suppressed anymore! You should only use this option if you know what
;; you are doing.
;;
;; (default = yes)
; dtmf-suppression = yes
;;
;; Adjust connection automagically if a FAX tone is detected.
;;
;; (default = yes)
; auto-fax-adjustment = yes
;;
;; Time (is seconds) since connection, when we may detect FAX tone and perform
;; automagical adjustment. After this, FAX tone detection is ignored.
;;
;; Possible values: 0 - 9999
;;
;; (default = 30).
; fax-adjustment-timeout = 30
;;
;; Enable/disable pulse detection (reported as DTMF digits).
;;
;; (default = no)
; pulse-forwarding = no
;;
;; Enable correct standard following for R2/MFC protocol.
;;
;; NOTE: This option is disabled by default as R2 protocol timeout for condition
;; reporting is only 7 seconds, and for correct operation we need to send a
;; fake "Line Free" condition if Asterisk is used in the midle of two R2 links
;; (forwarding calls from one to another) or if incoming calls may take long to
;; receive RINGING or BUSY status.
;;
;; You can set this to "yes" to have a correct behaviour for condition
;; obtaining/reporting, but only if Asterisk is directly connected to a CO (central
;; office) or a fast signaling PBX, or incoming calls are handled quickly enogth by
;; dialplan logic - IOW, you must GUARANTEE there will be no more than 7 seconds from
;; call arrival to call progress reporting.
;;
;; (default = no)
; r2-strict-behaviour = no
;;
;; Set the delay (in ms) after sending ringback condition where audio stream
;; should be opened for the channel. Limited to 25ms minimum, 500ms maximum.
;;
;; (default = 250)
; r2-preconnect-wait = 250
;;
;; Chooses the method for disconnecting calls on the R2/MFC protocol.
;;
;; NOTE: This option is adjusted to "force-disconnect" by default, as disconnecting
;; incoming calls in the standard way may result in 90 seconds delay for releasing the
;; timeslot, thus leading to an under-utilization of the available lines. This option
;; is non-ITU.T, but allowed by Brazillian variant of R2/MFC.
;;
;; This option can be changed to "release" for a more agressive disconnect method, if
;; the "force-disconnect" is not releasing calls fast enought, or to "disconnect" for
;; attaining strictly to the international R2 standard.
;;
;; WARNING: Do not use the option 'release' unless you really know what you are doing!
;; This option instructs the channel to have a complete non-standard behaviour,
;; which could result in signaling errors for the remote site.
;;
;; Available options:
;;
;; force-disconnect Use force disconnection (signals AB as '00')
;; disconnect Use standard disconnection (signals AB as '11')
;; release Release the channel w/o disconnecting (signals AB as '10')
;;
;; (default = "force-disconnect")
; r2-hangup-behaviour = force-disconnect
;;
;; Set output volume level. Possible values:
;;
;; * '+ N' = increase N times;
;; * '- N' = decrease N times.
;; * '0' = leave default.
;;
;; (default = 0)
; output-volume = 0
;;
;; Set input volume level. Can only be used if AGC (and not pulse detection)
;; is enabled on the board configuration. Possible values:
;;
;; * '+ N' = increase N times;
;; * '- N' = decrease N times.
;; * '0' = leave default.
;;
;; (default = 0)
; input-volume = 0
;;
;; Sets the default AMA flags, affecting the categorization of entries in the
;; call detail records.
;;
;; (default = default)
; amaflags = default
;;
;; Defines call groups for calls of all channel.
;;
;; (default = 0)
; callgroup = 0
;;
;; Set the default group that can pickup fellow workers' calls.
;;
;; (default = 0)
; pickupgroup = 0
;;
;; Sets the account code for calls placed on the channel. The account code may
;; be any alphanumeric string.
;;
;; (default = <empty>)
; accountcode =
;;
;; Set the language of the channel (useful for selecting audio messages of a
;; specific language on answer).
;;
;; (default = <empty>)
; language = pt_BR
;;
;; Set the music on hold class of the channel (useful for selecting a group of
;; songs to play on hold).
;;
;; (default = default)
; mohclass = default
;;
;; Sets the CallerID detection type for FXO boards using Asterisk(r) interface.
;;
;; NOTE: This option is not used for BINA DTMF, which is detected by the
;; board DSPs, and is always enabled.
;;
;; Possible values:
;;
;; bell Bell FSK standard
;; v23 V23 FSK standard
;; none Disable the detection
;;
;; (default = bell)
; fxo-fsk-detection = bell
;; Sets the timeout for CallerID detection (is ms).
;;
;; (default = 2000)
; fxo-fsk-timeout = 2000
;;
;; Sets the amount of time to wait before sending FLASH command on KUserTransfer
;; application. Possible values are between 500 and 5000 miliseconds.
;;
;; (default = 1000)
; fxo-user-xfer-delay = 1000
;;
;; Enable or disable sending originator number throught BINA DTMF signalization
;; to a FXS branch during call setup.
;; (default = no)
; fxs-send-bina-dtmf = no
;;
;; Enable or disable sending originator number throught FSK signalization to a
;; FXS branch during call setup.
;; (default = no)
; fxs-send-fsk = no
;;
;; Enables/disables the use of sharp (#) as an end-of-number digit for immediate
;; dialing. This does not affect special numbers which start on sharp, like '#8'.
;;
;; (default = no)
; immediate-sharp-dial = no
;;
;; Sets the numbers (separated by comma) in which the standard dialtone will
;; be changed to CO dialtone (central office tone) when they are pressed.
;;
;; (default = <empty>)
; co-dialtone-digits = 0,90
;;
;; Sets the cadence name for standard dialtone (PBX dialtone by default).
;;
;; (default = std-dialtone)
; dialtone-cadence = pbx-dialtone
;;
;; Sets the cadence name for voicemail dialtone, used for the branch
;; associated with an user when he/she has voicemail.
;;
;; (default = vm-dialtone)
; vm-dialtone-cadence = vm-dialtone
;;
;; Sets the cadence name for CO dialtone.
;;
;; (default = co-dialtone)
; co-dialtone-cadence = co-dialtone
;;
;; This is the delay time to really disconnect a channel after the disconnect
;; event arrive. If a connect event comes up in this interval, then the
;; disconnection is ignored and the call keeps going on. Values are in
;; miliseconds.
;;
;; (default = 0)
; disconnect-delay = 0
;;
;; This timer controls the delay associated with ringback generation in the
;; Khomp channel, when the other side *does not send audio* - in other words,
;; this is used when calling peers located at VoIP channels.
;; Values are in milliseconds.
;;
;; (default = 1500)
; delay-ringback-co = 1500
;;
;; This timer controls the delay associated with ringback generation in the
;; Khomp channel when the other side report a continuous stream of audio in
;; silence - in other words, this is used when the audio is present but does
;; not have any tone. Values are in milliseconds.
;;
;; (default = 2500)
; delay-ringback-pbx = 2500
;;
;; Defines if the channel should optimize audio delay by droping longstanding
;; packets from audio buffer. This guarantees the minimal audio delay for the
;; user, and avoid delays associated with miscoded SIP clients. However,
;; depending on the system's scheduling policy (some 2.6 kernel releases),
;; this may result on excessive drop of packets, and audible audio skipping.
;; This should not be changed naively.
;;
;; (default = no)
; optimize-audio-path = no
;;
;; Defines if the channel should ignore some uncommon DTMF digits detected by
;; the board (A, B, C and D). This reduces the number of false positives which
;; may happen sometimes, without affecting correctness on traditional scenarios.
;; However, if you need to pass those digits througth the board, you may need
;; to set this option to 'no'.
;;
;; (default = yes)
; ignore-letter-dtmfs = yes
;;
;; For KFXO boards, defines if audio will be allowed being sent into outgoing
;; calls before it has been answered.
;;
;; (default = yes)
; fxo-send-pre-audio = yes
;;
;; When there is a request to indicate busy for an incoming KFXO call, the
;; ringing channel is taken off hook and then placed on hook, so it now goes
;; to answered state and we can send audio throught the line.
;; If the off/on hook time interval is too short, the PSTN may ignore it, and
;; keep the line in a ringing state. If it is too long, the call may be charged.
;; The option below defines the delay between the line being answered and being
;; disconnected, in miliseconds (from 50ms to 90000ms).
;;
;; (default = 1250)
; fxo-busy-disconnection = 1250
;;
;; Defines the timeout, in seconds, between digits of a FXS board's extension.
;;
;; (default = 7)
; fxs-digit-timeout = 7
;;
;; Enables/Disables the action of dropping collect calls. If enabled, all
;; collect calls will be dropped no matter what KDropCollectCall is set to.
;; This feature is not available for KGSM-USB devices.
;;
;; (default = no)
; drop-collect-call = no
;;
;; Configures which calls detected by call analyzer will be dropped
;; when the "drop" option is passed on the Dial application.
;; Available only in digital signaling (E1 or GSM).
;; Possible values:
;;
;; message_box,human_answer,answering_machine,carrier_message,unknown
;;
;; (default = <empty>)
; drop-options =
;;
;; When enabled, this channel will be elegible for transfer using PBX-to-PBX signaling
;; if the underlying signalization supports this feature (ISDN, Line Side, CAS EL7,
;; E1 LC, or FXO signalings).
;;
;; (default = yes)
; user-transfer-enable = yes
;;
;; Defines the facility used by the ISDN protocol for QSig transfers. Possible values:
;;
;; qsig-ct Use QSig CT transfer (preferred method).
;; qsig-ct-rr Use QSig CT transfer with reroute.
;; qsig-ct-pr Use QSig CT transfer eith path replace.
;; qsig-ssct Use QSig SSCT transfer.
;; any Use any method.
;; none Do not transfer ISDN calls.
;;
;; (default = qsig-ct)
; qsig-transfer-facility = qsig-ct
;;
;; Defines the flash behaviour: attended transfer or pendulum.
;;
;; Possible values:
;;
;; xfer Use atxfer option from features.conf.
;; pendulum Use pendulum (if enabled).
;; auto If enabled, use pendulum, if not, use xfer.
;;
;; (default = xfer)
;; flash-behaviour = xfer
;;
;; Defines the DTMF digit sequence used to answer a waiting call (placing the current
;; on hold), place the active call on hold and start a new outgoing call, and to
;; alternate between answered calls.
;;
;; (default = empty)
; pendulum-digits =
;; Defines the channels where the pendulum feature for outgoing will be enabled automatically.
;;
;; Possible values:
;;
;; none No channel will have the feature by default.
;; fxs Enable the feature for FXS channels only.
;; all All channels will have the feature.
;;
;; (default = fxs)
pendulum-native = none
;; Defines the channels where the pendulum feature for incoming will be enabled automatically
;; (removing the need to use "@" in the Dial string for activation).
;;
;; Possible values:
;;
;; none No channel will have the feature by default.
;; fxs Enable the feature for FXS channels only.
;; all All channels will have the feature.
;;
;; (default = fxs)
;;
;; NOTE: In case of signallings or technologies where a single channel does not
;; necessarily corresponds to a specific user or destination, the "@"
;; notation is needed for specifying the destination number to the
;; allocation procedures. For more info, please check the Manual.
pendulum-incoming = none
;; Defines the timeout for an waiting call to stay in the ringing state without being
;; answered. After this, the waiting call is dropped, and the waiting cadence is stopped.
;;
;; (default = 20000ms)
; pendulum-timeout = 20000
;; Defines if channels should be enabled or not. Useful in specific channel sections.
;; Use for debugging purpouses only, not recommended for production environments.
;;
;; (default = yes)
; enabled = yes
;;;;
;;;; Section for configuring allocation groups.
;;;;
[groups]
;; In this section, you should define options using the following syntax:
;;
;; <groupname> = <allocation-string>[:<context>]
;;
;; ex: 'pstn = b0l0 + b1c38' (without quotes)
;; ex: 'pstn = b0l0 + b1c38:from-pstn' (without quotes)
;;
;; You may define your own groups. In the example above, the group
;; called pstn can be used in the Dial string as Dial(Khomp/Gpstn/...)
;; or Dial(Khomp/gpstn/...). As a result, the allocation string will be
;; replaced with the one associated with the group "pstn". This is the same
;; of doing Dial(Khomp/b0l0 + b1c38/...).
;; In the second example, a context is also defined which can be used in
;; extensions.conf for inbound calls on that allocation string's range.
;;
;; If, and only if, choose compatibility mode like DAHDI in compatibily session, your
;; groups must follow the syntax:
;; <groupname> = <channel#>-<channel#>[:<context>] | <channel#>, <channel#>[:<context>]
;;
;; ex: '1 = 1-30' (without quotes)
;; ex: '1 = 1,2,3,4-10:from-pstn' (without quotes)
;;
;; Or if no group was added, the groups for compatibily mode like DAHDI will be set automatically
;; with full range of channels at the each board/link, and each board/link will be a group,
;; per example, for a E1 only board, will be generated two groups, first with a 1 to 30 channels (link 1), and
;; second group of 31 to 60 channels (link 2).
;;;;
;;;; Section for configuring cadences used on FXS boards and the whole
;;;; channel (fast busy, ringback tones, etc).
;;;;
;E1
;1=b0l0
;GSM
;Vazio
;1=b1c2-4
;Claro
;2=b1c0-7
;OI
;3=b1c5-6
;Vivo
;4=b1c7
{GRUPOS}
[cadences]
;; Default value for cadences. You may define your own cadences, and
;; also use them in the Dial arguments as "ring_cadence=your_cadence_name".
;;
;; "0,0" means a continuous dialtone, only valid for audio generation
;; (i.e. cannot be used for ringing an FXS channel).
;;
;; (default as defined below, independent of compatibility mode,
;; and only to be used in Dial arguments)
; co-dialtone = 0,0
; vm-dialtone = 1000,100,100,100
; pbx-dialtone = 1000,100
; fast-busy = 100,100
; ringback = 1000,4000
; waiting-call = 100,100,100,3700
; ring = 1000,4000
;; If choose compatibily mode like DAHDI in compatibility session, the
;; cadences must be a sequential number starts by 1, to be used in dial string.
;;
;; ex: Dial(Khomp/1r1/9901)
;;
;; Note: Negative values are not accepted, to configuring where the caller ID spill is
;; to be placed, use the properly configuration section.
;;
;; (default as defined below)
; 1 = 125, 125, 2000, 4000
; 2 = 250, 250, 500, 1000
; 3 = 125, 125, 125, 125
; 4 = 1000, 500, 2500, 5000
;;;;
;;;; Section for configuring CALLERID's associated with FXS boards.
;;;;
[fxs-branches]
;; In this section, you should define options using the following syntax:
;;
;; 'orig_prefix = serial number 0, serial number 1, ...'
;;
;; ex: '800 = 1234,4535' (without quotes)
{SERIES}
;;
;; In the example above (assuming KFXS-SPX boards 1234 and 4535), the
;; branches will be numbered from 800 to 829 in board 1234, and from
;; 830 to 859 in board 4535.
;;;;
;;;; Section for configuring FXS hotlines.
;;;;
[fxs-hotlines]
;; In this section, you should define options using the following syntax:
;;
;; 'orig_prefix = destination_number'
;;
;; ex: '804 = 32332933' (without quotes)
;;
;; In the example above, the branch numbered 804 will call the number
;; 3233-2933 (Khomp support number) every time the FXS branch goes off hook.
{HOTLINES}
;;;
;;; Section for configuring specific options for FXS branches.
;;;
[fxs-options]
;; In this section, you should define options using the following syntax:
;;
;; 'orig_prefix = option1:value | option2:value | option3:value' ...
;;
;; ex: '804 = pickupgroup:1,4-15 | output-volume:+2' (without quotes)
;;
{RAMAIS}
;; In the example above, the branch numbered 804 will have specific
;; configuration for 'pickupgroup' and default output volume set to +2.
;;
;; Possible values to options is:
;; pickupgroup, callgroup, context, input-volume, output-volume language,
;; mohclass, amaflags, accountcode, calleridnum, calleridname, mailbox.