Advertisement
Not a member of Pastebin yet?
Sign Up,
it unlocks many cool features!
- ;
- ; DAHDI telephony
- ;
- ; Configuration file
- ;
- ; You need to restart Asterisk to re-configure the DAHDI channel
- ; CLI> 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 unrem lines.
- ;
- ; remmed-out examples below (those lines that begin with a ';' but no
- ; space afterwards) typically show a value that is not the defauult value,
- ; but would make sense under cetain 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 or GR-303 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 calls. Defaults to 'default'
- ;
- ;context=incoming
- ;
- ; 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
- ;
- ; 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
- ;
- ; PRI Dialplan: The ISDN-level Type Of Number (TON) or numbering plan, used for
- ; the dialed number. For most installations, leaving this as 'unknown' (the
- ; default) works in the most cases. In some very unusual circumstances, you
- ; may need to set this to 'dynamic' or 'redundant'. Note that if you set one
- ; of the others, you will be unable to dial another class of numbers. For
- ; example, if you set 'national', you will be unable to dial local or
- ; international numbers.
- ;
- ; 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.
- ;
- ; Neither pridialplan nor prilocaldialplan can be changed on reload.
- ;
- ; unknown: Unknown
- ; private: Private ISDN
- ; local: Local ISDN
- ; national: National ISDN
- ; international: International ISDN
- ; dynamic: Dynamically selects the appropriate dialplan
- ; redundant: Same as dynamic, except that the underlying number is not
- ; changed (not common)
- ;
- ;pridialplan=unknown
- ;prilocaldialplan=national
- ;
- ; pridialplan may be also set at dialtime, by prefixing the dialled 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
- ; dialled 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 dialled 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
- ;
- ; 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
- ;
- ; Allow inband audio (progress) when a call is RELEASEd by the far end of a PRI
- ;
- ;inbanddisconnect=yes
- ;
- ; 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
- ;
- ; 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 -1,
- ; Asterisk clears calls)
- ; 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)
- ;
- ;pritimer => t200,1000
- ;pritimer => t313,4000
- ;
- ; 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
- ;
- ; 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
- ; gr303fxoks_net: GR-303 Signalling, FXO Loopstart, Network side
- ; gr303fxsks_cpe: GR-303 Signalling, FXS Loopstart, CPE 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
- ;
- ; 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 doc/India-CID.txt)
- ;
- ;cidstart=polarity
- ;
- ; 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
- ;
- ; The following option enables receiving MWI on FXO lines. The default
- ; value is no. When this is enabled, and MWI notification indicates on or off,
- ; the script specified by the mwimonitornotify option is executed. Also, an
- ; internal Asterisk MWI event will be generated so that any other part of
- ; Asterisk that cares about MWI state changes will get notified, just as if
- ; the state change came from app_voicemail. 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 mailbox, and the second is 1 or 0, indicating if
- ; there are messages waiting or not.
- ;
- ;mwimonitornotify=/usr/local/bin/dahdinotify.sh
- ;
- ; 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
- ;
- ; 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.
- ;
- transfer=yes
- ;
- ; Allow call parking
- ; ('canpark=no' is overridden by 'transfer=yes')
- ;
- canpark=yes
- ;
- ; 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 a mailbox is specified without a voicemail
- ; context, then when voicemail is received in a mailbox in the default
- ; voicemail context in voicemail.conf, taking the phone off hook will cause a
- ; stutter dialtone instead of a normal one.
- ;
- ; If a mailbox is specified *with* a voicemail context, the same will result
- ; if voicemail received in mailbox in the specified voicemail context.
- ;
- ; for default voicemail context, the example below is fine:
- ;
- ;mailbox=1234
- ;
- ; for any other voicemail context, the following will produce the stutter tone:
- ;
- ;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=256
- ;
- ; 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
- ;
- ; You may also set the default receive and transmit gains (in dB)
- ;
- ; Gain Settings: increasing / decreasing the volume level on a channel.
- ; The values are in db (decibells). A positive number
- ; increases the volume level on a channel, and a
- ; negavive value decreases volume level.
- ;
- ; There are several independent gain settings:
- ; rxgain: gain for the rx (receive - into Asterisk) channel. Default: 0.0
- ; txgain: gain for the tx (transmit - out of Asterisk Asterisk) channel.
- ; Default: 0.0
- ; cid_rxgain: set the gain just for the caller ID sounds Asterisk
- ; emits. Default: 5.0 .
- rxgain=0.0
- txgain=0.0
- ;
- ; 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
- ; Channel variable to be set for all calls from this channel
- ;setvar=CHANNEL=42
- ;
- ; 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
- ;
- ; 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
- ;
- ; 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 ztmonitor utility to record the audio that main/dsp.c
- ; is receiving after the caller hangs up.
- ;
- ; Use a polarity reversal to mark when a outgoing call is answered by the
- ; remote party.
- ;
- ;answeronpolarityswitch=yes
- ;
- ; 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
- ;
- ; 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.
- ;
- ; If this option is set to "passthrough", then the hold message will always be
- ; passed through as signalling instead of generating hold music locally. This
- ; setting is only valid when used on a channel that uses digital signalling.
- ;
- ; 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
- ;
- ; 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.
- ;
- ; For GR-303, CRV's are created like channels except they must start with the
- ; trunk group followed by a colon, e.g.:
- ;
- ; crv => 1:1
- ; crv => 2:1-2,5-8
- ;
- ;
- ;callerid="Green Phone"<(256) 428-6121>
- ;channel => 1
- ;callerid="Black Phone"<(256) 428-6122>
- ;channel => 2
- ;callerid="CallerID Phone" <(630) 372-1564>
- ;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
- ; ---------------- 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.
- ;ss7_explictacm=yes
- ; All settings apply to linkset 1
- ;linkset = 1
- ; 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
- ;
- ; For more information on setting up SS7, see the README file in libss7 or
- ; the doc/ss7.txt file in the Asterisk source tree.
- ; ----------------- SS7 Options ----------------------------------------
- #include dahdi-channels.conf
Advertisement
Add Comment
Please, Sign In to add comment
Advertisement