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chan_dahdi.conf.sample
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;
; DAHDI Telephony Configuration file
;
; You need to restart Asterisk to re-configure the DAHDI channel
; CLI> module reload chan_dahdi.so
; will reload the configuration file, but not all configuration options
; are re-configured during a reload (signalling, as well as PRI and
; SS7-related settings cannot be changed on a reload).
;
; This file documents many configuration variables. Normally unless you know
; what a variable means or that it should be changed, there's no reason to
; un-comment those lines.
;
; Examples below that are commented out (those lines that begin with a ';' but
; no space afterwards) typically show a value that is not the default value,
; but would make sense under certain circumstances. The default values are
; usually sane. Thus you should typically not touch them unless you know what
; they mean or you know you should change them.
[trunkgroups]
;
; Trunk groups are used for NFAS connections.
;
; Group: Defines a trunk group.
; trunkgroup => <trunkgroup>,<dchannel>[,<backup1>...]
;
; trunkgroup is the numerical trunk group to create
; dchannel is the DAHDI channel which will have the
; d-channel for the trunk.
; backup1 is an optional list of backup d-channels.
;
;trunkgroup => 1,24,48
;trunkgroup => 1,24
;
; Spanmap: Associates a span with a trunk group
; spanmap => <dahdispan>,<trunkgroup>[,<logicalspan>]
;
; dahdispan is the DAHDI span number to associate
; trunkgroup is the trunkgroup (specified above) for the mapping
; logicalspan is the logical span number within the trunk group to use.
; if unspecified, no logical span number is used.
;
;spanmap => 1,1,1
;spanmap => 2,1,2
;spanmap => 3,1,3
;spanmap => 4,1,4
[channels]
;
; Default language
;
;language=en
;
; Context for incoming calls. Defaults to 'default'
;
context=public
;
; Switchtype: Only used for PRI.
;
; national: National ISDN 2 (default)
; dms100: Nortel DMS100
; 4ess: AT&T 4ESS
; 5ess: Lucent 5ESS
; euroisdn: EuroISDN (common in Europe)
; ni1: Old National ISDN 1
; qsig: Q.SIG
;
;switchtype=euroisdn
;
; MSNs for ISDN spans. Asterisk will listen for the listed numbers on
; incoming calls and ignore any calls not listed.
; Here you can give a comma separated list of numbers or dialplan extension
; patterns. An empty list disables MSN matching to allow any incoming call.
; Only set on PTMP CPE side of ISDN span if needed.
; The default is an empty list.
;msn=
;
; Some switches (AT&T especially) require network specific facility IE.
; Supported values are currently 'none', 'sdn', 'megacom', 'tollfreemegacom', 'accunet'
;
; nsf cannot be changed on a reload.
;
;nsf=none
;
;service_message_support=yes
; Enable service message support for channel. Must be set after switchtype.
;
; Dialing options for ISDN (i.e., Dial(DAHDI/g1/exten/options)):
; R Reverse Charge Indication
; Indicate to the called party that the call will be reverse charged.
; K(n) Keypad digits n
; Send out the specified digits as keypad digits.
;
; PRI Dialplan: The ISDN-level Type Of Number (TON) or numbering plan, used for
; the dialed number. Leaving this as 'unknown' (the default) works for most
; cases. In some very unusual circumstances, you may need to set this to
; 'dynamic' or 'redundant'.
;
; unknown: Unknown
; private: Private ISDN
; local: Local ISDN
; national: National ISDN
; international: International ISDN
; dynamic: Dynamically selects the appropriate dialplan using the
; prefix settings.
; redundant: Same as dynamic, except that the underlying number is not
; changed (not common)
;
; pridialplan cannot be changed on reload.
;pridialplan=unknown
;
; PRI Local Dialplan: Only RARELY used for PRI (sets the calling number's
; numbering plan). In North America, the typical use is sending the 10 digit
; callerID number and setting the prilocaldialplan to 'national' (the default).
; Only VERY rarely will you need to change this.
;
; unknown: Unknown
; private: Private ISDN
; local: Local ISDN
; national: National ISDN
; international: International ISDN
; from_channel: Use the CALLERID(ton) value from the channel.
; dynamic: Dynamically selects the appropriate dialplan using the
; prefix settings.
; redundant: Same as dynamic, except that the underlying number is not
; changed (not common)
;
; prilocaldialplan cannot be changed on reload.
;prilocaldialplan=national
;
; PRI Connected Line Dialplan: Sets the connected party number's numbering plan.
;
; unknown: Unknown
; private: Private ISDN
; local: Local ISDN
; national: National ISDN
; international: International ISDN
; from_channel: Use the CONNECTEDLINE(ton) value from the channel.
; dynamic: Dynamically selects the appropriate dialplan using the
; prefix settings.
; redundant: Same as dynamic, except that the underlying number is not
; changed (not common)
;
; pricpndialplan cannot be changed on reload.
;pricpndialplan=from_channel
;
; pridialplan may be also set at dialtime, by prefixing the dialed number with
; one of the following letters:
; U - Unknown
; I - International
; N - National
; L - Local (Net Specific)
; S - Subscriber
; V - Abbreviated
; R - Reserved (should probably never be used but is included for completeness)
;
; Additionally, you may also set the following NPI bits (also by prefixing the
; dialed string with one of the following letters):
; u - Unknown
; e - E.163/E.164 (ISDN/telephony)
; x - X.121 (Data)
; f - F.69 (Telex)
; n - National
; p - Private
; r - Reserved (should probably never be used but is included for completeness)
;
; You may also set the prilocaldialplan in the same way, but by prefixing the
; Caller*ID Number rather than the dialed number.
; Please note that telcos which require this kind of additional manipulation
; of the TON/NPI are *rare*. Most telco PRIs will work fine simply by
; setting pridialplan to unknown or dynamic.
;
;
; PRI caller ID prefixes based on the given TON/NPI (dialplan)
; This is especially needed for EuroISDN E1-PRIs
;
; None of the prefix settings can be changed on reload.
;
; sample 1 for Germany
;internationalprefix = 00
;nationalprefix = 0
;localprefix = 0711
;privateprefix = 07115678
;unknownprefix =
;
; sample 2 for Germany
;internationalprefix = +
;nationalprefix = +49
;localprefix = +49711
;privateprefix = +497115678
;unknownprefix =
;
; PRI resetinterval: sets the time in seconds between restart of unused
; B channels; defaults to 'never'.
;
;resetinterval = 3600
;
; Enable per ISDN span to force a RESTART on a channel that returns a cause
; code of PRI_CAUSE_REQUESTED_CHAN_UNAVAIL(44). If this option is enabled
; and the reason the peer rejected the call with cause 44 was that the
; channel is stuck in an unavailable state on the peer, then this might
; help release the channel. It is worth noting that the next outgoing call
; Asterisk makes will likely try the same channel again.
;
; NOTE: Sending a RESTART in response to a cause 44 is not required
; (nor prohibited) by the standards and is likely a primitive chan_dahdi
; response to call collisions (glare) and buggy peers. However, there
; are telco switches out there that ignore the RESTART and continue to
; send calls to the channel in the restarting state.
; Default no.
;
;force_restart_unavailable_chans=yes
;
; Assume inband audio may be present when a SETUP ACK message is received.
; Q.931 Section 5.1.3 says that in scenarios with overlap dialing, when a
; dialtone is sent from the network side, progress indicator 8 "Inband info
; now available" MAY be sent to the CPE if no digits were received with
; the SETUP. It is thus implied that the ie is mandatory if digits came
; with the SETUP and dialtone is needed.
; This option should be enabled, when the network sends dialtone and you
; want to hear it, but the network doesn't send the progress indicator when
; needed.
;
; NOTE: For Q.SIG setups this option should be enabled when outgoing overlap
; dialing is also enabled because Q.SIG does not send the progress indicator
; with the SETUP ACK.
; Default no.
;
;inband_on_setup_ack=yes
;
; Assume inband audio may be present when a PROCEEDING message is received.
; Q.931 Section 5.1.2 says the network cannot assume that the CPE side has
; attached to the B channel at this time without explicitly sending the
; progress indicator ie informing the CPE side to attach to the B channel
; for audio. However, some non-compliant ISDN switches send a PROCEEDING
; without the progress indicator ie indicating inband audio is available and
; assume that the CPE device has connected the media path for listening to
; ringback and other messages.
; Default no.
;
;inband_on_proceeding=yes
;
; Overlap dialing mode (sending overlap digits)
; Cannot be changed on a reload.
;
; incoming: incoming direction only
; outgoing: outgoing direction only
; no: neither direction
; yes or both: both directions
;
;overlapdial=yes
; Send/receive ISDN display IE options. The display options are a comma separated
; list of the following options:
;
; block: Do not pass display text data.
; Q.SIG: Default for send/receive.
; ETSI CPE: Default for send.
; name_initial: Use display text in SETUP/CONNECT messages as the party name.
; Default for all other modes.
; name_update: Use display text in other messages (NOTIFY/FACILITY) for COLP name
; update.
; name: Combined name_initial and name_update options.
; text: Pass any unused display text data as an arbitrary display message
; during a call. Sent text goes out in an INFORMATION message.
;
; * Default is an empty string for legacy behavior.
; * The name options are not recommended for Q.SIG since Q.SIG already
; supports names.
; * The send block is the only recommended setting for CPE mode since Q.931 uses
; the display IE only in the network to user direction.
;
; display_send and display_receive cannot be changed on reload.
;
;display_send=
;display_receive=
; Allow sending an ISDN Malicious Caller ID (MCID) request on this span.
; Default disabled
;
;mcid_send=yes
; Send ISDN date/time IE in CONNECT message option. Only valid on NT spans.
;
; no: Do not send date/time IE in CONNECT message.
; date: Send date only.
; date_hh Send date and hour.
; date_hhmm Send date, hour, and minute.
; date_hhmmss Send date, hour, minute, and second.
;
; Default is an empty string which lets libpri pick the default
; date/time IE send policy.
;
;datetime_send=
; Send ISDN conected line information.
;
; block: Do not send any connected line information.
; connect: Send connected line information on initial connect.
; update: Same as connect but also send any updates during a call.
; Updates happen if the call is transferred. (Default)
;
;colp_send=update
; Allow inband audio (progress) when a call is DISCONNECTed by the far end of a PRI
;
;inbanddisconnect=yes
;
; Allow a held call to be transferred to the active call on disconnect.
; This is useful on BRI PTMP NT lines where an ISDN phone can simulate the
; transfer feature of an analog phone.
; The default is no.
;hold_disconnect_transfer=yes
; BRI PTMP layer 1 presence.
; You should normally not need to set this option.
; You may need to set this option if your telco brings layer 1 down when
; the line is idle.
; required: Layer 1 presence required for outgoing calls. (default)
; ignore: Ignore alarms from DAHDI about this span.
; (Layer 1 and 2 will be brought back up for an outgoing call.)
; NOTE: You will not be able to detect physical line problems
; until an outgoing call is attempted and fails.
;
;layer1_presence=ignore
; BRI PTMP layer 2 persistence.
; You should normally not need to set this option.
; You may need to set this option if your telco brings layer 1 down when
; the line is idle.
; <blank>: Use libpri default.
; keep_up: Bring layer 2 back up if peer takes it down.
; leave_down: Leave layer 2 down if peer takes it down. (Libpri default)
; (Layer 2 will be brought back up for an outgoing call.)
;
;layer2_persistence=leave_down
; PRI Out of band indications.
; Enable this to report Busy and Congestion on a PRI using out-of-band
; notification. Inband indication, as used by Asterisk doesn't seem to work
; with all telcos.
;
; outofband: Signal Busy/Congestion out of band with RELEASE/DISCONNECT
; inband: Signal Busy/Congestion using in-band tones (default)
;
; priindication cannot be changed on a reload.
;
;priindication = outofband
;
; If you need to override the existing channels selection routine and force all
; PRI channels to be marked as exclusively selected, set this to yes.
;
; priexclusive cannot be changed on a reload.
;
;priexclusive = yes
;
;
; If you need to use the logical channel mapping with your Q.SIG PRI instead
; of the physical mapping you must use the qsigchannelmapping option.
;
; logical: Use the logical channel mapping
; physical: Use physical channel mapping (default)
;
;qsigchannelmapping=logical
;
; If you wish to ignore remote hold indications (and use MOH that is supplied over
; the B channel) enable this option.
;
;discardremoteholdretrieval=yes
;
; ISDN Timers
; All of the ISDN timers and counters that are used are configurable. Specify
; the timer name, and its value (in ms for timers).
; K: Layer 2 max number of outstanding unacknowledged I frames (default 7)
; N200: Layer 2 max number of retransmissions of a frame (default 3)
; T200: Layer 2 max time before retransmission of a frame (default 1000 ms)
; T203: Layer 2 max time without frames being exchanged (default 10000 ms)
; T305: Wait for DISCONNECT acknowledge (default 30000 ms)
; T308: Wait for RELEASE acknowledge (default 4000 ms)
; T309: Maintain active calls on Layer 2 disconnection (default 6000 ms)
; EuroISDN: 6000 to 12000 ms, according to (N200 + 1) x T200 + 2s
; May vary in other ISDN standards (Q.931 1993 : 90000 ms)
; T313: Wait for CONNECT acknowledge, CPE side only (default 3000 ms)
;
; T-RESPONSE: Maximum time to wait for a typical APDU response. (default 4000 ms)
; This is an implementation timer when the standard does not specify one.
; T-ACTIVATE: Request supervision timeout. (default 10000 ms)
; T-RETENTION: Maximum time to wait for user A to activate call-completion. (default 30000 ms)
; Used by ETSI PTP, ETSI PTMP, and Q.SIG as the cc_offer_timer.
; T-CCBS1: T-STATUS timer equivalent for CC user A status. (default 4000 ms)
; T-CCBS2: Maximum time the CCBS service will be active (default 45 min in ms)
; T-CCBS3: Maximum time to wait for user A to respond to user B availability. (default 20000 ms)
; T-CCBS5: Network B CCBS supervision timeout. (default 60 min in ms)
; T-CCBS6: Network A CCBS supervision timeout. (default 60 min in ms)
; T-CCNR2: Maximum time the CCNR service will be active (default 180 min in ms)
; T-CCNR5: Network B CCNR supervision timeout. (default 195 min in ms)
; T-CCNR6: Network A CCNR supervision timeout. (default 195 min in ms)
; CC-T1: Q.SIG CC request supervision timeout. (default 30000 ms)
; CCBS-T2: Q.SIG CCBS supervision timeout. (default 60 min in ms)
; CCNR-T2: Q.SIG CCNR supervision timeout. (default 195 min in ms)
; CC-T3: Q.SIG CC Maximum time to wait for user A to respond to user B availability. (default 30000 ms)
;
;pritimer => t200,1000
;pritimer => t313,4000
;
; CC PTMP recall mode:
; specific - Only the CC original party A can participate in the CC callback
; global - Other compatible endpoints on the PTMP line can be party A in the CC callback
;
; cc_ptmp_recall_mode cannot be changed on a reload.
;
;cc_ptmp_recall_mode = specific
;
; CC Q.SIG Party A (requester) retain signaling link option
; retain Require that the signaling link be retained.
; release Request that the signaling link be released.
; do_not_care The responder is free to choose if the signaling link will be retained.
;
;cc_qsig_signaling_link_req = retain
;
; CC Q.SIG Party B (responder) retain signaling link option
; retain Prefer that the signaling link be retained.
; release Prefer that the signaling link be released.
;
;cc_qsig_signaling_link_rsp = retain
;
; See ccss.conf.sample for more options. The timers described by ccss.conf.sample
; are not used by ISDN for the native protocol since they are defined by the
; standards and set by pritimer above.
;
; To enable transmission of facility-based ISDN supplementary services (such
; as caller name from CPE over facility), enable this option.
; Cannot be changed on a reload.
;
;facilityenable = yes
;
; This option enables Advice of Charge pass-through between the ISDN PRI and
; Asterisk. This option can be set to any combination of 's', 'd', and 'e' which
; represent the different variants of Advice of Charge, AOC-S, AOC-D, and AOC-E.
; Advice of Charge pass-through is currently only supported for ETSI. Since most
; AOC messages are sent on facility messages, the 'facilityenable' option must
; also be enabled to fully support AOC pass-through.
;
;aoc_enable=s,d,e
;
; When this option is enabled, a hangup initiated by the ISDN PRI side of the
; asterisk channel will result in the channel delaying its hangup in an
; attempt to receive the final AOC-E message from its bridge. The delay
; period is configured as one half the T305 timer length. If the channel
; is not bridged the hangup will occur immediatly without delay.
;
;aoce_delayhangup=yes
; pritimer cannot be changed on a reload.
;
; Signalling method. The default is "auto". Valid values:
; auto: Use the current value from DAHDI.
; em: E & M
; em_e1: E & M E1
; em_w: E & M Wink
; featd: Feature Group D (The fake, Adtran style, DTMF)
; featdmf: Feature Group D (The real thing, MF (domestic, US))
; featdmf_ta: Feature Group D (The real thing, MF (domestic, US)) through
; a Tandem Access point
; featb: Feature Group B (MF (domestic, US))
; fgccama: Feature Group C-CAMA (DP DNIS, MF ANI)
; fgccamamf: Feature Group C-CAMA MF (MF DNIS, MF ANI)
; fxs_ls: FXS (Loop Start)
; fxs_gs: FXS (Ground Start)
; fxs_ks: FXS (Kewl Start)
; fxo_ls: FXO (Loop Start)
; fxo_gs: FXO (Ground Start)
; fxo_ks: FXO (Kewl Start)
; pri_cpe: PRI signalling, CPE side
; pri_net: PRI signalling, Network side
; bri_cpe: BRI PTP signalling, CPE side
; bri_net: BRI PTP signalling, Network side
; bri_cpe_ptmp: BRI PTMP signalling, CPE side
; bri_net_ptmp: BRI PTMP signalling, Network side
; sf: SF (Inband Tone) Signalling
; sf_w: SF Wink
; sf_featd: SF Feature Group D (The fake, Adtran style, DTMF)
; sf_featdmf: SF Feature Group D (The real thing, MF (domestic, US))
; sf_featb: SF Feature Group B (MF (domestic, US))
; e911: E911 (MF) style signalling
; ss7: Signalling System 7
; mfcr2: MFC/R2 Signalling. To specify the country variant see 'mfcr2_variant'
;
; The following are used for Radio interfaces:
; fxs_rx: Receive audio/COR on an FXS kewlstart interface (FXO at the
; channel bank)
; fxs_tx: Transmit audio/PTT on an FXS loopstart interface (FXO at the
; channel bank)
; fxo_rx: Receive audio/COR on an FXO loopstart interface (FXS at the
; channel bank)
; fxo_tx: Transmit audio/PTT on an FXO groundstart interface (FXS at
; the channel bank)
; em_rx: Receive audio/COR on an E&M interface (1-way)
; em_tx: Transmit audio/PTT on an E&M interface (1-way)
; em_txrx: Receive audio/COR AND Transmit audio/PTT on an E&M interface
; (2-way)
; em_rxtx: Same as em_txrx (for our dyslexic friends)
; sf_rx: Receive audio/COR on an SF interface (1-way)
; sf_tx: Transmit audio/PTT on an SF interface (1-way)
; sf_txrx: Receive audio/COR AND Transmit audio/PTT on an SF interface
; (2-way)
; sf_rxtx: Same as sf_txrx (for our dyslexic friends)
; ss7: Signalling System 7
;
; signalling of a channel can not be changed on a reload.
;
;signalling=fxo_ls
;
; If you have an outbound signalling format that is different from format
; specified above (but compatible), you can specify outbound signalling format,
; (see below). The 'signalling' format specified will be the inbound signalling
; format. If you only specify 'signalling', then it will be the format for
; both inbound and outbound.
;
; outsignalling can only be one of:
; em, em_e1, em_w, sf, sf_w, sf_featd, sf_featdmf, sf_featb, featd,
; featdmf, featdmf_ta, e911, fgccama, fgccamamf
;
; outsignalling cannot be changed on a reload.
;
;signalling=featdmf
;
;outsignalling=featb
;
; For Feature Group D Tandem access, to set the default CIC and OZZ use these
; parameters (Will not be updated on reload):
;
;defaultozz=0000
;defaultcic=303
;
; A variety of timing parameters can be specified as well
; The default values for those are "-1", which is to use the
; compile-time defaults of the DAHDI kernel modules. The timing
; parameters, (with the standard default from DAHDI):
;
; prewink: Pre-wink time (default 50ms)
; preflash: Pre-flash time (default 50ms)
; wink: Wink time (default 150ms)
; flash: Flash time (default 750ms)
; start: Start time (default 1500ms)
; rxwink: Receiver wink time (default 300ms)
; rxflash: Receiver flashtime (default 1250ms)
; debounce: Debounce timing (default 600ms)
;
; None of them will update on a reload.
;
; How long generated tones (DTMF and MF) will be played on the channel
; (in milliseconds).
;
; This is a global, rather than a per-channel setting. It will not be
; updated on a reload.
;
;toneduration=100
;
; Whether or not to do distinctive ring detection on FXO lines:
;
;usedistinctiveringdetection=yes
;
; enable dring detection after caller ID for those countries like Australia
; where the ring cadence is changed *after* the caller ID spill:
;
;distinctiveringaftercid=yes
;
; Whether or not to use caller ID:
;
usecallerid=yes
;
; Type of caller ID signalling in use
; bell = bell202 as used in US (default)
; v23 = v23 as used in the UK
; v23_jp = v23 as used in Japan
; dtmf = DTMF as used in Denmark, Sweden and Netherlands
; smdi = Use SMDI for caller ID. Requires SMDI to be enabled (usesmdi).
;
;cidsignalling=v23
;
; What signals the start of caller ID
; ring = a ring signals the start (default)
; polarity = polarity reversal signals the start
; polarity_IN = polarity reversal signals the start, for India,
; for dtmf dialtone detection; using DTMF.
; (see https://wiki.asterisk.org/wiki/display/AST/Caller+ID+in+India)
; dtmf = causes monitor loop to look for dtmf energy on the
; incoming channel to initate cid acquisition
;
;cidstart=polarity
;
; When cidstart=dtmf, the energy level on the line used to trigger dtmf cid
; acquisition. This number is compared to the average over a packet of audio
; of the absolute values of 16 bit signed linear samples. The default is set
; to 256. The choice of 256 is arbitrary. The value you should select should
; be high enough to prevent false detections while low enough to insure that
; no dtmf spills are missed.
;
;dtmfcidlevel=256
;
; Whether or not to hide outgoing caller ID (Override with *67 or *82)
; (If your dialplan doesn't catch it)
;
;hidecallerid=yes
;
; Enable if you need to hide just the name and not the number for legacy PBX use.
; Only applies to PRI channels.
;hidecalleridname=yes
;
; On UK analog lines, the caller hanging up determines the end of calls. So
; Asterisk hanging up the line may or may not end a call (DAHDI could just as
; easily be re-attaching to a prior incoming call that was not yet hung up).
; This option changes the hangup to wait for a dialtone on the line, before
; marking the line as once again available for use with outgoing calls.
; Specified in milliseconds, not set by default.
;waitfordialtone=1000
;
; For analog lines, enables Asterisk to use dialtone detection per channel
; if an incoming call was hung up before it was answered. If dialtone is
; detected, the call is hung up.
; no: Disabled. (Default)
; yes: Look for dialtone for 10000 ms after answer.
; <number>: Look for dialtone for the specified number of ms after answer.
; always: Look for dialtone for the entire call. Dialtone may return
; if the far end hangs up first.
;
;dialtone_detect=no
;
; The following option enables receiving MWI on FXO lines. The default
; value is no.
; The mwimonitor can take the following values
; no - No mwimonitoring occurs. (default)
; yes - The same as specifying fsk
; fsk - the FXO line is monitored for MWI FSK spills
; fsk,rpas - the FXO line is monitored for MWI FSK spills preceded
; by a ring pulse alert signal.
; neon - The fxo line is monitored for the presence of NEON pulses
; indicating MWI.
; When detected, an internal Asterisk MWI event is generated so that any other
; part of Asterisk that cares about MWI state changes is notified, just as if
; the state change came from app_voicemail.
; For FSK MWI Spills, the energy level that must be seen before starting the
; MWI detection process can be set with 'mwilevel'.
;
;mwimonitor=no
;mwilevel=512
;
; This option is used in conjunction with mwimonitor. This will get executed
; when incoming MWI state changes. The script is passed 2 arguments. The
; first is the corresponding configured mailbox, and the second is 1 or 0,
; indicating if there are messages waiting or not.
; Note: app_voicemail mailboxes are in the form of mailbox@context.
;
; /usr/local/bin/dahdinotify.sh 501@mailboxes 1
;
;mwimonitornotify=/usr/local/bin/dahdinotify.sh
;
; The following keyword 'mwisendtype' enables various VMWI methods on FXS lines (if supported).
; The default is to send FSK only.
; The following options are available;
; 'rpas' Ring Pulse Alert Signal, alerts intelligent phones that a FSK message is about to be sent.
; 'lrev' Line reversed to indicate messages waiting.
; 'hvdc' 90Vdc OnHook DC voltage to indicate messages waiting.
; 'hvac' or 'neon' 90Vac OnHook AC voltage to light Neon bulb.
; 'nofsk' Disables FSK MWI spills from being sent out.
; It is feasible that multiple options can be enabled.
;mwisendtype=rpas,lrev
;
; Whether or not to enable call waiting on internal extensions
; With this set to 'yes', busy extensions will hear the call-waiting
; tone, and can use hook-flash to switch between callers. The Dial()
; app will not return the "BUSY" result for extensions.
;
callwaiting=yes
;
; Configure the number of outstanding call waiting calls for internal ISDN
; endpoints before bouncing the calls as busy. This option is equivalent to
; the callwaiting option for analog ports.
; A call waiting call is a SETUP message with no B channel selected.
; The default is zero to disable call waiting for ISDN endpoints.
;max_call_waiting_calls=0
;
; Allow incoming ISDN call waiting calls.
; A call waiting call is a SETUP message with no B channel selected.
;allow_call_waiting_calls=no
; Configure the ISDN span to indicate MWI for the list of mailboxes.
; You can give a comma separated list of up to 8 mailboxes per span.
; An empty list disables MWI.
;
; The default is an empty list.
;mwi_mailboxes=vm-mailbox{,vm-mailbox}
; vm-mailbox = Internal voicemail mailbox identifier.
; Note: app_voicemail mailboxes must be in the form of mailbox@context.
;mwi_mailboxes=501@mailboxes,502@mailboxes
; Configure the ISDN mailbox number sent over the span for MWI mailboxes.
; The position of the number in the list corresponds to the position in
; mwi_mailboxes. If either position in mwi_mailboxes or mwi_vm_boxes is
; empty then that position is disabled.
;
; The default is an empty list.
;mwi_vm_boxes=mailbox_number{,mailbox_number}
;mwi_vm_boxes=501,502
; Configure the ISDN span voicemail controlling numbers for MWI mailboxes.
; What number to call for a user to retrieve voicemail messages.
;
; You can give a comma separated list of numbers. The position of the number
; corresponds to the position in mwi_mailboxes. If a position is empty then
; the last number is reused.
;
; For example:
; mwi_vm_numbers=700,,800,,900
; is equivalent to:
; mwi_vm_numbers=700,700,800,800,900,900,900,900
;
; The default is no number.
;mwi_vm_numbers=
; Whether or not restrict outgoing caller ID (will be sent as ANI only, not
; available for the user)
; Mostly use with FXS ports
; Does nothing. Use hidecallerid instead.
;
;restrictcid=no
;
; Whether or not to use the caller ID presentation from the Asterisk channel
; for outgoing calls.
; See dialplan function CALLERID(pres) for more information.
; Only applies to PRI and SS7 channels.
;
usecallingpres=yes
;
; Some countries (UK) have ring tones with different ring tones (ring-ring),
; which means the caller ID needs to be set later on, and not just after
; the first ring, as per the default (1).
;
;sendcalleridafter = 2
;
;
; Support caller ID on Call Waiting
;
callwaitingcallerid=yes
;
; Support three-way calling
;
threewaycalling=yes
;
; For FXS ports (either direct analog or over T1/E1):
; Support flash-hook call transfer (requires three way calling)
; Also enables call parking (overrides the 'canpark' parameter)
;
; For digital ports using ISDN PRI protocols:
; Support switch-side transfer (called 2BCT, RLT or other names)
; This setting must be enabled on both ports involved, and the
; 'facilityenable' setting must also be enabled to allow sending
; the transfer to the ISDN switch, since it sent in a FACILITY
; message.
; NOTE: This should be disabled for NT PTMP mode. Phones cannot
; have tromboned calls pushed down to them.
;
transfer=yes
;
; Allow call parking
; ('canpark=no' is overridden by 'transfer=yes')
;
canpark=yes
; Sets the default parking lot for call parking.
; This is setable per channel.
; Parkinglots are configured in features.conf
;
;parkinglot=plaza
;
; Support call forward variable
;
cancallforward=yes
;
; Whether or not to support Call Return (*69, if your dialplan doesn't
; catch this first)
;
callreturn=yes
;
; Stutter dialtone support: If voicemail is received in the mailbox then
; taking the phone off hook will cause a stutter dialtone instead of a
; normal one.
;
; Note: app_voicemail mailboxes must be in the form of mailbox@context.
;
;mailbox=1234@context
;
; Enable echo cancellation
; Use either "yes", "no", or a power of two from 32 to 256 if you wish to
; actually set the number of taps of cancellation.
;
; Note that when setting the number of taps, the number 256 does not translate
; to 256 ms of echo cancellation. echocancel=256 means 256 / 8 = 32 ms.
;
; Note that if any of your DAHDI cards have hardware echo cancellers,
; then this setting only turns them on and off; numeric settings will
; be treated as "yes". There are no special settings required for
; hardware echo cancellers; when present and enabled in their kernel
; modules, they take precedence over the software echo canceller compiled
; into DAHDI automatically.
;
;
echocancel=yes
;
; Some DAHDI echo cancellers (software and hardware) support adjustable
; parameters; these parameters can be supplied as additional options to
; the 'echocancel' setting. Note that Asterisk does not attempt to
; validate the parameters or their values, so if you supply an invalid
; parameter you will not know the specific reason it failed without
; checking the kernel message log for the error(s) put there by DAHDI.
;
;echocancel=128,param1=32,param2=0,param3=14
;
; Generally, it is not necessary (and in fact undesirable) to echo cancel when
; the circuit path is entirely TDM. You may, however, change this behavior
; by enabling the echo canceller during pure TDM bridging below.
;
echocancelwhenbridged=yes
;
; In some cases, the echo canceller doesn't train quickly enough and there
; is echo at the beginning of the call. Enabling echo training will cause
; DAHDI to briefly mute the channel, send an impulse, and use the impulse
; response to pre-train the echo canceller so it can start out with a much
; closer idea of the actual echo. Value may be "yes", "no", or a number of
; milliseconds to delay before training (default = 400)
;
; WARNING: In some cases this option can make echo worse! If you are
; trying to debug an echo problem, it is worth checking to see if your echo
; is better with the option set to yes or no. Use whatever setting gives
; the best results.
;
; Note that these parameters do not apply to hardware echo cancellers.
;
;echotraining=yes
;echotraining=800
;
; If you are having trouble with DTMF detection, you can relax the DTMF
; detection parameters. Relaxing them may make the DTMF detector more likely
; to have "talkoff" where DTMF is detected when it shouldn't be.
;
;relaxdtmf=yes
;
; Hardware gain settings increase/decrease the analog volume level on a channel.
; The values are in db (decibels) and can be adjusted in 0.1 dB increments.
; A positive number increases the volume level on a channel, and a negavive
; value decreases volume level.
;
; Hardware gain settings are only possible on hardware with analog ports
; because the gain is done on the analog side of the analog/digital conversion.
;
; When hardware gains are disabled, Asterisk will NOT touch the gain setting
; already configured in hardware.
;
; hwrxgain: Hardware receive gain for the channel (into Asterisk).
; Default: disabled
; hwtxgain: Hardware transmit gain for the channel (out of Asterisk).
; Default: disabled
;
;hwrxgain=disabled
;hwtxgain=disabled
;hwrxgain=2.0
;hwtxgain=3.0
;
; Software gain settings digitally increase/decrease the volume level on a channel.
; The values are in db (decibels). A positive number increases the volume
; level on a channel, and a negavive value decreases volume level.
;
; Software gains work on the digital side of the analog/digital conversion
; and thus can also work with T1/E1 cards.
;
; rxgain: Software receive gain for the channel (into Asterisk). Default: 0.0
; txgain: Software transmit gain for the channel (out of Asterisk).
; Default: 0.0
;
; cid_rxgain: Add this gain to rxgain when Asterisk expects to receive
; a Caller ID stream.
; Default: 5.0 .
;
;rxgain=2.0
;txgain=3.0
;
; Dynamic Range Compression: You can also enable dynamic range compression
; on a channel. This will digitally amplify quiet sounds while leaving louder
; sounds untouched. This is useful in situations where a linear gain setting
; would cause clipping. Acceptable values are in the range of 0.0 to around
; 6.0 with higher values causing more compression to be done.
;
; rxdrc: dynamic range compression for the rx channel. Default: 0.0
; txdrc: dynamic range compression for the tx channel. Default: 0.0
;
;rxdrc=1.0
;txdrc=4.0
;
; Logical groups can be assigned to allow outgoing roll-over. Groups range
; from 0 to 63, and multiple groups can be specified. By default the
; channel is not a member of any group.
;
; Note that an explicit empty value for 'group' is invalid, and will not
; override a previous non-empty one. The same applies to callgroup and
; pickupgroup as well.
;
group=1
;
; Ring groups (a.k.a. call groups) and pickup groups. If a phone is ringing
; and it is a member of a group which is one of your pickup groups, then
; you can answer it by picking up and dialing *8#. For simple offices, just
; make these both the same. Groups range from 0 to 63.
;
callgroup=1
pickupgroup=1
;
; Named ring groups (a.k.a. named call groups) and named pickup groups.
; If a phone is ringing and it is a member of a group which is one of your
; named pickup groups, then you can answer it by picking up and dialing *8#.
; For simple offices, just make these both the same.
; The number of named groups is not limited.
;
;namedcallgroup=engineering,sales,netgroup,protgroup
;namedpickupgroup=sales
; Channel variables to be set for all calls from this channel
;setvar=CHANNEL=42
;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
; cause the given audio file to
; be played upon completion of
; an attended transfer to the
; target of the transfer.
;
; Specify whether the channel should be answered immediately or if the simple
; switch should provide dialtone, read digits, etc.
; Note: If immediate=yes the dialplan execution will always start at extension
; 's' priority 1 regardless of the dialed number!
;
;immediate=yes
;
; Specify whether flash-hook transfers to 'busy' channels should complete or
; return to the caller performing the transfer (default is yes).
;
;transfertobusy=no
; Calls will have the party id user tag set to this string value.
;
;cid_tag=
; With this set, you can automatically append the MSN of a party
; to the cid_tag. An '_' is used to separate the tag from the MSN.
; Applies to ISDN spans.
; Default is no.
;
; Table of what number is appended:
; outgoing incoming
; net dialed caller
; cpe caller dialed
;
;append_msn_to_cid_tag=no
; caller ID can be set to "asreceived" or a specific number if you want to
; override it. Note that "asreceived" only applies to trunk interfaces.
; fullname sets just the
;
; fullname: sets just the name part.
; cid_number: sets just the number part:
;
;callerid = 123456
;
;callerid = My Name <2564286000>
; Which can also be written as:
;cid_number = 2564286000
;fullname = My Name
;
;callerid = asreceived
;
; should we use the caller ID from incoming call on DAHDI transfer?
;
;useincomingcalleridondahditransfer = yes
;
; Add a description for the channel which can be shown through the Asterisk
; console when executing the 'dahdi show channels' command is run.
;
;description=Phone located in lobby
;
; AMA flags affects the recording of Call Detail Records. If specified
; it may be 'default', 'omit', 'billing', or 'documentation'.
;
;amaflags=default
;
; Channels may be associated with an account code to ease
; billing
;
;accountcode=lss0101