~singpolyma/asterisk

ref: refs/heads/master asterisk/configs/samples/chan_dahdi.conf.sample -rw-r--r-- 66.9 KiB
1c6d77d7Stephen Paul Weber Fix presence from component sub-addresses a month ago
                                                                                
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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
;
; ADSI (Analog Display Services Interface) can be enabled on a per-channel
; basis if you have (or may have) ADSI compatible CPE equipment
;
;adsi=yes
;
; SMDI (Simplified Message Desk Interface) can be enabled on a per-channel
; basis if you would like that channel to behave like an SMDI message desk.
; The SMDI port specified should have already been defined in smdi.conf.  The
; default port is /dev/ttyS0.
;
;usesmdi=yes
;smdiport=/dev/ttyS0
;
; On trunk interfaces (FXS) and E&M interfaces (E&M, Wink, Feature Group D
; etc, it can be useful to perform busy detection either in an effort to
; detect hangup or for detecting busies.  This enables listening for
; the beep-beep busy pattern.
;
;busydetect=yes
;
; If busydetect is enabled, it is also possible to specify how many busy tones
; to wait for before hanging up.  The default is 3, but it might be
; safer to set to 6 or even 8.  Mind that the higher the number, the more
; time that will be needed to hangup a channel, but lowers the probability
; that you will get random hangups.
;
;busycount=6
;
; If busydetect is enabled, it is also possible to specify the cadence of your
; busy signal.  In many countries, it is 500msec on, 500msec off.  Without
; busypattern specified, we'll accept any regular sound-silence pattern that
; repeats <busycount> times as a busy signal.  If you specify busypattern,
; then we'll further check the length of the sound (tone) and silence, which
; will further reduce the chance of a false positive.
;
;busypattern=500,500
;
; NOTE: In make menuselect, you'll find further options to tweak the busy
; detector.  If your country has a busy tone with the same length tone and
; silence (as many countries do), consider enabling the
; BUSYDETECT_COMPARE_TONE_AND_SILENCE option.
;
; To further detect which hangup tone your telco provider is sending, it is
; useful to use the dahdi_monitor utility to record the audio that main/dsp.c
; is receiving after the caller hangs up.
;
; For FXS (FXO signalled) ports
;   switch the line polarity to signal the connected PBX that an outgoing
;   call was answered by the remote party.
; For FXO (FXS signalled) ports
;   watch for a polarity reversal to mark when a outgoing call is
;   answered by the remote party.
;
;answeronpolarityswitch=yes
;
; For FXS (FXO signalled) ports
;   switch the line polarity to signal the connected PBX that the current
;   call was "hung up" by the remote party
; For FXO (FXS signalled) ports
;   In some countries, a polarity reversal is used to signal the disconnect of a
;   phone line.  If the hanguponpolarityswitch option is selected, the call will
;   be considered "hung up" on a polarity reversal.
;
;hanguponpolarityswitch=yes
;
; polarityonanswerdelay: minimal time period (ms) between the answer
;                        polarity switch and hangup polarity switch.
;                        (default: 600ms)
;
; On trunk interfaces (FXS) it can be useful to attempt to follow the progress
; of a call through RINGING, BUSY, and ANSWERING.   If turned on, call
; progress attempts to determine answer, busy, and ringing on phone lines.
; This feature is HIGHLY EXPERIMENTAL and can easily detect false answers,
; so don't count on it being very accurate.
;
; Few zones are supported at the time of this writing, but may be selected
; with "progzone".
;
; progzone also affects the pattern used for buzydetect (unless
; busypattern is set explicitly). The possible values are:
;   us (default)
;   ca (alias for 'us')
;   cr (Costa Rica)
;   br (Brazil, alias for 'cr')
;   uk
;
; This feature can also easily detect false hangups. The symptoms of this is
; being disconnected in the middle of a call for no reason.
;
;callprogress=yes
;progzone=uk
;
; Set the tonezone. Equivalent of the defaultzone settings in
; /etc/dahdi/system.conf. This sets the tone zone by number.
; Note that you'd still need to load tonezones (loadzone in
; /etc/dahdi/system.conf).
; The default is -1: not to set anything.
;tonezone = 0 ; 0 is US
;
; FXO (FXS signalled) devices must have a timeout to determine if there was a
; hangup before the line was answered.  This value can be tweaked to shorten
; how long it takes before DAHDI considers a non-ringing line to have hungup.
;
; ringtimeout will not update on a reload.
;
;ringtimeout=8000
;
; For FXO (FXS signalled) devices, whether to use pulse dial instead of DTMF
; Pulse digits from phones (FXS devices, FXO signalling) are always
; detected.
;
;pulsedial=yes
;
; For fax detection, uncomment one of the following lines.  The default is *OFF*
;
;faxdetect=both
;faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no
;
; When 'faxdetect' is enabled, one could use 'faxdetect_timeout' to disable fax
; detection after the specified number of seconds into a call.  Be aware that
; outgoing analog channels may consider the channel is answered immediately
; when dialing completes.  Analog does not have a reliable method of detecting
; when the far end answers.  Zero disables the timeout.
; Default is 0 to disable the timeout.
;
;faxdetect_timeout=30
;
; When 'faxdetect' is used, one could use 'faxbuffers' to configure the DAHDI
; transmit buffer policy.  The default is *OFF*.  When this configuration
; option is used, the faxbuffer policy will be used for the life of the call
; after a fax tone is detected.  The faxbuffer policy is reverted after the
; call is torn down.  The sample below will result in 6 buffers and a full
; buffer policy.
;
;faxbuffers=>6,full
;
; When FXO signalling (FXS device, e.g. analog phone) is used, overlap dialing
; is typically used. Asterisk has several configurable (per-channel) timeouts
; to know how long to wait for the next digit. All the values are in
; milliseconds.
; * firstdigit_timeout: a longer timeout before any digit is dialed.
;   By default: 16 seconds.
; * interdigit_timeout: timeout for next digits, if the current number dialed
;   does not match a number in the current context. Default: 8 seconds.
; * matchdigit_timeout: timeout for next digits, if the current number dialed
;   matches a number in the current context. Default: 3 seconds.
;
;firstdigit_timeout=16000
;interdigit_timeout=8000
;matchdigit_timeout=3000
;
; Configure the default number of DAHDI buffers and the transmit policy to use.
; This can be used to eliminate data drops when scheduling jitter prevents
; Asterisk from writing to a DAHDI channel regularly. Most users will probably
; want "faxbuffers" instead of "buffers".
;
; The policies are:
; immediate - DAHDI will immediately start sending the data to the hardware after
;             Asterisk writes to the channel. This is the default mode. It
;             introduces the least amount of latency but has an increased chance for
;             hardware under runs if Asterisk is not able to keep the DAHDI write
;             queue from going empty.
; half      - DAHDI will wait until half of the configured buffers are full before
;             starting to transmit. This adds latency to the audio but reduces
;             the chance of under runs. Essentially, this is like an in-kernel jitter
;             buffer.
; full      - DAHDI will not start transmitting until all buffers are full.
;             Introduces the most amount of latency and is susceptible to over
;             runs from the Asterisk process.
;
; The receive policy is never changed. DAHDI will always pass up audio as soon
; as possible.
;
; The default number of buffers is 4 (from jitterbuffers) and the default policy
; is immediate.
;
;buffers=4,immediate
;
; This option specifies what to do when the channel's bridged peer puts the
; ISDN channel on hold.  Settable per logical ISDN span.
; moh:          Generate music-on-hold to the remote party.
; notify:       Send hold notification signaling to the remote party.
;               For ETSI PTP and ETSI PTMP NT links.
;               (The notify setting deprecates the mohinterpret=passthrough setting.)
; hold:         Use HOLD/RETRIEVE signaling to release the B channel while on hold.
;               For ETSI PTMP TE links.
;
;moh_signaling=moh
;
; 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.
;
; This option may be set globally or on a per-channel basis.
;
;mohinterpret=default
;
; This option specifies which music on hold class to suggest to the peer channel
; when this channel places the peer on hold.  This option may be set globally,
; or on a per-channel basis.
;
;mohsuggest=default
;
; PRI channels can have an idle extension and a minunused number.  So long as
; at least "minunused" channels are idle, chan_dahdi will try to call "idledial"
; on them, and then dump them into the PBX in the "idleext" extension (which
; is of the form exten@context).  When channels are needed the "idle" calls
; are disconnected (so long as there are at least "minidle" calls still
; running, of course) to make more channels available.  The primary use of
; this is to create a dynamic service, where idle channels are bundled through
; multilink PPP, thus more efficiently utilizing combined voice/data services
; than conventional fixed mappings/muxings.
;
; Those settings cannot be changed on reload.
;
;idledial=6999
;idleext=6999@dialout
;minunused=2
;minidle=1
;
;
; ignore_failed_channels: Continue even if some channels failed to configure.
; True by default. Disable this if you can guarantee that DAHDI starts before
; Asterisk and want to be sure chan_dahdi will not start with broken
; configuration.
;
;ignore_failed_channels = false
;
; Configure jitter buffers in DAHDI (each one is 20ms, default is 4)
; This is set globally, rather than per-channel.
;
;jitterbuffers=4
;
; ----------------------------- JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes              ; Enables the use of a jitterbuffer on the receiving side of a
                              ; DAHDI channel. Defaults to "no". An enabled jitterbuffer will
                              ; be used only if the sending side can create and the receiving
                              ; side can not accept jitter. The DAHDI channel can't accept jitter,
                              ; thus an enabled jitterbuffer on the receive DAHDI side will always
                              ; be used if the sending side can create jitter.

; jbmaxsize = 200             ; Max length of the jitterbuffer in milliseconds.

; jbresyncthreshold = 1000    ; 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, usually sent from exotic devices
                              ; and programs. Defaults to 1000.

; jbimpl = fixed              ; Jitterbuffer implementation, used on the receiving side of a DAHDI
                              ; channel. Two implementations are currently available - "fixed"
                              ; (with size always equals to jbmax-size) and "adaptive" (with
                              ; variable size, actually the new jb of IAX2). Defaults to fixed.

; jbtargetextra = 40          ; 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. the default is 40, so without
                              ; modification, the new jitter buffer will set its size to the jitter
                              ; value plus 40 milliseconds. increasing this value may help if your
                              ; network normally has low jitter, but occasionally has spikes.

; jblog = no                  ; Enables jitterbuffer frame logging. Defaults to "no".
; ----------------------------------------------------------------------------------
;
; You can define your own custom ring cadences here.  You can define up to 8
; pairs.  If the silence is negative, it indicates where the caller ID spill is
; to be placed.  Also, if you define any custom cadences, the default cadences
; will be turned off.
;
; This setting is global, rather than per-channel. It will not update on
; a reload.
;
; Syntax is:  cadence=ring,silence[,ring,silence[...]]
;
; These are the default cadences:
;
;cadence=125,125,2000,-4000
;cadence=250,250,500,1000,250,250,500,-4000
;cadence=125,125,125,125,125,-4000
;cadence=1000,500,2500,-5000
;
; Each channel consists of the channel number or range.  It inherits the
; parameters that were specified above its declaration.
;
;
;callerid="Green Phone"<(256) 428-6121>
;description=Reception Phone			; add a description for 'dahdi show channels'
;channel => 1
;callerid="Black Phone"<(256) 428-6122>
;description=Courtesy Phone
;channel => 2
;callerid="CallerID Phone" <(630) 372-1564>
;description=					; reset the description for following channels
;channel => 3
;callerid="Pac Tel Phone" <(256) 428-6124>
;channel => 4
;callerid="Uniden Dead" <(256) 428-6125>
;channel => 5
;callerid="Cortelco 2500" <(256) 428-6126>
;channel => 6
;callerid="Main TA 750" <(256) 428-6127>
;channel => 44
;
; For example, maybe we have some other channels which start out in a
; different context and use E & M signalling instead.
;
;context=remote
;signaling=em
;channel => 15
;channel => 16

;signalling=em_w
;
; All those in group 0 I'll use for outgoing calls
;
; Strip most significant digit (9) before sending
;
;stripmsd=1
;callerid=asreceived
;group=0
;signalling=fxs_ls
;channel => 45

;signalling=fxo_ls
;group=1
;callerid="Joe Schmoe" <(256) 428-6131>
;channel => 25
;callerid="Megan May" <(256) 428-6132>
;channel => 26
;callerid="Suzy Queue" <(256) 428-6233>
;channel => 27
;callerid="Larry Moe" <(256) 428-6234>
;channel => 28
;
; Sample PRI (CPE) config:  Specify the switchtype, the signalling as either
; pri_cpe or pri_net for CPE or Network termination, and generally you will
; want to create a single "group" for all channels of the PRI.
;
; switchtype cannot be changed on a reload.
;
; switchtype = national
; signalling = pri_cpe
; group = 2
; channel => 1-23

;  Used for distinctive ring support for x100p.
;  You can see the dringX patterns is to set any one of the dringXcontext fields
;  and they will be printed on the console when an inbound call comes in.
;
;  dringXrange is used to change the acceptable ranges for "tone offsets".  Defaults to 10.
;  Note: a range of 0 is NOT what you might expect - it instead forces it to the default.
;  A range of -1 will force it to always match.
;  Anything lower than -1 would presumably cause it to never match.
;
;dring1=95,0,0
;dring1context=internal1
;dring1range=10
;dring2=325,95,0
;dring2context=internal2
;dring2range=10
; If no pattern is matched here is where we go.
;context=default
;channel => 1

; AMI alarm event reporting
;reportalarms=channels
;Possible values are:
;channels - report each channel alarms (current behavior, default for backward compatibility)
;spans - report an "SpanAlarm" event when the span of any configured channel is alarmed
;all - report channel and span alarms (aggregated behavior)
;none - do not report any alarms.

; ---------------- Options for use with signalling=ss7 -----------------
; None of them can be changed by a reload.
;
; Variant of SS7 signalling:
; Options are itu and ansi
;ss7type = itu

; SS7 Called Nature of Address Indicator
;
; unknown:        Unknown
; subscriber:     Subscriber
; national:       National
; international:  International
; dynamic:        Dynamically selects the appropriate dialplan
;
;ss7_called_nai=dynamic
;
; SS7 Calling Nature of Address Indicator
;
; unknown:        Unknown
; subscriber:     Subscriber
; national:       National
; international:  International
; dynamic:        Dynamically selects the appropriate dialplan
;
;ss7_calling_nai=dynamic
;
;
; sample 1 for Germany
;ss7_internationalprefix = 00
;ss7_nationalprefix = 0
;ss7_subscriberprefix =
;ss7_unknownprefix =
;

; This option is used to disable automatic sending of ACM when the call is started
; in the dialplan.  If you do use this option, you will need to use the Proceeding()
; application in the dialplan to send ACM or enable ss7_autoacm below.
;ss7_explicitacm=yes

; Use this option to automatically send ACM when the call rings or is answered and
; has not seen proceeding yet. If you use this option, you should disable ss7_explicitacm.
; You may still use Proceeding() to explicitly send an ACM from the dialplan.
;ss7_autoacm=yes

; Create the linkset with all CICs in hardware remotely blocked state.
;ss7_initialhwblo=yes

; This option is whether or not to trust the remote echo control indication.  This means
; that in cases where echo control is reported by the remote end, we will trust them and
; not enable echo cancellation on the call.
;ss7_use_echocontrol=yes

; This option is to set what our echo control indication is to the other end.  Set to
; yes to indicate that we are using echo cancellation or no if we are not.
;ss7_default_echocontrol=yes

; All settings apply to linkset 1
;linkset = 1

; Set the Signaling Link Code (SLC) for each sigchan.
; If you manually set any you need to manually set all.
; Should be defined before sigchan.
; The default SLC starts with zero and increases for each defined sigchan.
;slc=

; Point code of the linkset.  For ITU, this is the decimal number
; format of the point code.  For ANSI, this can either be in decimal
; number format or in the xxx-xxx-xxx format
;pointcode = 1

; Point code of node adjacent to this signalling link (Possibly the STP between you and
; your destination).  Point code format follows the same rules as above.
;adjpointcode = 2

; Default point code that you would like to assign to outgoing messages (in case of
; routing through STPs, or using A links).  Point code format follows the same rules
; as above.
;defaultdpc = 3

; Begin CIC (Circuit indication codes) count with this number
;cicbeginswith = 1

; What the MTP3 network indicator bits should be set to.  Choices are
; national, national_spare, international, international_spare
;networkindicator=international

; First signalling channel
;sigchan = 48

; Additional signalling channel for this linkset (So you can have a linkset
; with two signalling links in it).  It seems like a silly way to do it, but
; for linksets with multiple signalling links, you add an additional sigchan
; line for every additional signalling link on the linkset.
;sigchan = 96

; Channels to associate with CICs on this linkset
;channel = 25-47
;

; Set this option if you wish to send an Information Request Message (INR) request
; if no calling party number is specified. This will attempt to tell the other end
; to send it anyways. Should be defined after sigchan.
;inr_if_no_calling=yes

; Set this to set whether or not the originating access is (non) ISDN in the forward and
; backward call indicators. Should be defined after sigchan
;non_isdn_access=yes

; This sets the number of binary places to shift the CIC when doing load balancing between
; sigchans on a linkset. Should be defined after sigchan. Default 0
;sls_shift = 0

; Send custom cause_location value
; Should be defined after sigchan. Default 1 (private local)
;cause_location=1

; SS7 timers (ISUP and MTP3) should be explicitly defined for each linkset to be used.
; For a full list of supported timers and their default values (applicable for both ITU
; and ANSI) see ss7.timers
; Should be defined after sigchan
;#include ss7.timers

; For more information on setting up SS7, see the README file in libss7 or
; https://wiki.asterisk.org/wiki/display/AST/Signaling+System+Number+7
; ----------------- SS7 Options ----------------------------------------

; ---------------- Options for use with signalling=mfcr2 --------------

; MFC-R2 signaling has lots of variants from country to country and even sometimes
; minor variants inside the same country. The only mandatory parameters here are:
; mfcr2_variant, mfcr2_max_ani and mfcr2_max_dnis.
; IT IS RECOMMENDED that you leave the default values (leaving it commented) for the
; other parameters unless you have problems or you have been instructed to change some
; parameter. OpenR2 library uses the mfcr2_variant parameter to try to determine the
; best defaults for your country, also refer to the OpenR2 package directory
; doc/asterisk/ where you can find sample configurations for some countries. If you
; want to contribute your configs for a particular country send them to the e-mail
; of the primary OpenR2 developer that you can find in the AUTHORS file of the OpenR2 package

; MFC/R2 variant. This depends on the OpenR2 supported variants
; A list of values can be found by executing the openr2 command r2test -l
; some valid values are:
; ar (Argentina)
; br (Brazil)
; mx (Mexico)
; ph (Philippines)
; itu (per ITU spec)
; mfcr2_variant=mx

; Max amount of ANI to ask for
; mfcr2_max_ani=10

; Max amount of DNIS to ask for
; mfcr2_max_dnis=4

; whether or not to get the ANI before getting DNIS.
; some telcos require ANI first some others do not care
; if this go wrong, change this value
; mfcr2_get_ani_first=no

; Caller Category to send
; national_subscriber
; national_priority_subscriber
; international_subscriber
; international_priority_subscriber
; collect_call
; usually national_subscriber works just fine
; you can change this setting from the dialplan
; by setting the variable MFCR2_CATEGORY
; (remember to set _MFCR2_CATEGORY from originating channels)
; MFCR2_CATEGORY will also be a variable available in your context
; on incoming calls set to the value received from the far end
; mfcr2_category=national_subscriber

; Call logging is stored at the Asterisk
; logging directory specified in asterisk.conf
; plus mfcr2/<whatever you put here>
; if you specify 'span1' here and asterisk.conf has
; as logging directory /var/log/asterisk then the full
; path to your MFC/R2 call logs will be /var/log/asterisk/mfcr2/span1
; (the directory will be automatically created if not present already)
; remember to set mfcr2_call_files=yes
; mfcr2_logdir=span1

; whether or not to drop call files into mfcr2_logdir
; mfcr2_call_files=yes|no

; MFC/R2 valid logging values are: all,error,warning,debug,notice,cas,mf,stack,nothing
; error,warning,debug and notice are self-descriptive
; 'cas' is for logging ABCD CAS tx and rx
; 'mf' is for logging of the Multi Frequency tones
; 'stack' is for very verbose output of the channel and context call stack, only useful
; if you are debugging a crash or want to learn how the library works. The stack logging
; will be only enabled if the openr2 library was compiled with -DOR2_TRACE_STACKS
; You can mix up values, like: loglevel=error,debug,mf to log just error, debug and
; multi frequency messages
; 'all' is a special value to log all the activity
; 'nothing' is a clean-up value, in case you want to not log any activity for
; a channel or group of channels
; BE AWARE that the level of output logged will ALSO depend on
; the value you have in logger.conf, if you disable output in logger.conf
; then it does not matter you specify 'all' here, nothing will be logged
; so logger.conf has the last word on what is going to be logged
; mfcr2_logging=all

; MFC/R2 value in milliseconds for the MF timeout. Any negative value
; means 'default', smaller values than 500ms are not recommended
; and can cause malfunctioning. If you experience protocol error
; due to MF timeout try incrementing this value in 500ms steps
; mfcr2_mfback_timeout=-1

; MFC/R2 value in milliseconds for the metering pulse timeout.
; Metering pulses are sent by some telcos for some R2 variants
; during a call presumably for billing purposes to indicate costs,
; however this pulses use the same signal that is used to indicate
; call hangup, therefore a timeout is sometimes required to distinguish
; between a *real* hangup and a billing pulse that should not
; last more than 500ms, If you experience call drops after some
; minutes of being stablished try setting a value of some ms here,
; values greater than 500ms are not recommended.
; BE AWARE that choosing the proper protocol mfcr2_variant parameter
; implicitly sets a good recommended value for this timer, use this
; parameter only when you *really* want to override the default, otherwise
; just comment out this value or put a -1
; Any negative value means 'default'.
; mfcr2_metering_pulse_timeout=-1

; Brazil uses a special calling party category for collect calls (llamadas por cobrar)
; instead of using the operator (as in Mexico). The R2 spec in Brazil says a special GB tone
; should be used to reject collect calls. If you want to ALLOW collect calls specify 'yes',
; if you want to BLOCK collect calls then say 'no'. Default is to block collect calls.
; (see also 'mfcr2_double_answer')
; mfcr2_allow_collect_calls=no

; This feature is related but independent of mfcr2_allow_collect_calls
; Some PBX's require a double-answer process to block collect calls, if
; you ever have problems blocking collect calls using Group B signals (mfcr2_allow_collect_calls=no)
; then you may want to try with mfcr2_double_answer=yes, this will cause that every answer signal
; is changed by answer->clear back->answer (sort of a flash)
; (see also 'mfcr2_allow_collect_calls')
; mfcr2_double_answer=no

; This feature allows to skip the use of Group B/II signals and go directly
; to the accepted state for incoming calls
; mfcr2_immediate_accept=no

; You most likely dont need this feature. Default is yes.
; When this is set to yes, all calls that are offered (incoming calls) which
; DNIS is valid (exists in extensions.conf) and pass collect call validation
; will be accepted with a Group B tone (either call with charge or not, depending on mfcr2_charge_calls)
; with this set to 'no' then the call will NOT be accepted on offered, and the call will start its
; execution in extensions.conf without being accepted until the channel is answered (either with Answer() or
; any other application resulting in the channel being answered).
; This can be set to 'no' if your telco or PBX needs the hangup cause to be set accurately
; when this option is set to no you must explicitly accept the call with DAHDIAcceptR2Call
; or implicitly through the Answer() application.
; mfcr2_accept_on_offer=yes

; Skip request of calling party category and ANI
; you need openr2 >= 1.2.0 to use this feature
; mfcr2_skip_category=no

; WARNING: advanced users only! I really mean it
; this parameter is commented by default because
; YOU DON'T NEED IT UNLESS YOU REALLY GROK MFC/R2
; READ COMMENTS on doc/r2proto.conf in openr2 package
; for more info
; mfcr2_advanced_protocol_file=/path/to/r2proto.conf

; Brazil use a special signal to force the release of the line (hangup) from the
; backward perspective. When mfcr2_forced_release=no, the normal clear back signal
; will be sent on hangup, which is OK for all mfcr2 variants I know of, except for
; Brazilian variant, where the central will leave the line up for several seconds (30, 60)
; which sometimes is not what people really want. When mfcr2_forced_release=yes, a different
; signal will be sent to hangup the call indicating that the line should be released immediately
; mfcr2_forced_release=no

; Whether or not report to the other end 'accept call with charge'
; This setting has no effect with most telecos, usually is safe
; leave the default (yes), but once in a while when interconnecting with
; old PBXs this may be useful.
; Concretely this affects the Group B signal used to accept calls
; The application DAHDIAcceptR2Call can also be used to decide this
; in the dial plan in a per-call basis instead of doing it here for all calls
; mfcr2_charge_calls=yes

; ---------------- END of options to be used with signalling=mfcr2

; Configuration Sections
; ~~~~~~~~~~~~~~~~~~~~~~
; You can also configure channels in a separate chan_dahdi.conf section. In
; this case the keyword 'channel' is not used. Instead the keyword
; 'dahdichan' is used (as in users.conf) - configuration is only processed
; in a section where the keyword dahdichan is used. It will only be
; processed in the end of the section. Thus the following section:
;
;[phones]
;echocancel = 64
;dahdichan = 1-8
;group = 1
;
; Is somewhat equivalent to the following snippet in the section
; [channels]:
;
;echocancel = 64
;group = 1
;channel => 1-8
;
; When starting a new section almost all of the configuration values are
; copied from their values at the end of the section [channels] in
; chan_dahdi.conf and [general] in users.conf - one section's configuration
; does not affect another one's.
;
; Instead of letting common configuration values "slide through" you can
; use configuration templates to easily keep the common part in one
; place and override where needed.
;
;[phones](!)
;echocancel = yes
;group = 0,4
;callgroup = 3
;pickupgroup = 3
;threewaycalling = yes
;transfer = yes
;context = phones
;faxdetect = incoming
;
;[phone-1](phones)
;dahdichan = 1
;callerid = My Name <501>
;mailbox = 501@mailboxes
;
;
;[fax](phones)
;dahdichan = 2
;faxdetect = no
;context = fax
;
;[phone-3](phones)
;dahdichan = 3
;pickupgroup = 3,4