Asterisk config chan_dahdi.conf
Configuration file for Chan_DAHDi lookups in Asterisk
;
;
; 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.
;
;
; 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
;
;
; 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=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
;
; 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.
;
; 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.
;
;
;pridialplan=unknown
;prilocaldialplan=national
;
; pridialplan may be also set at dialtime, by prefixing the dialled number with
; 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
; 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 =
;
;
;resetinterval = 3600
;
; Overlap dialing mode (sending overlap digits)
; Cannot be changed on a reload.
;
;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.
;
; 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).
;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.
;
; fgccamamf Feature Group C-CAMA MF (MF DNIS, MF ANI)
; 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.
;
; 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
;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
; 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
;
;usedistinctiveringdetection=yes
;
; enable dring detection after caller ID for those countries like Australia
;distinctiveringaftercid=yes
;
usecallerid=yes
;
; Hide the name part and leave just the number part of the caller ID
; string. Only applies to PRI channels.
;hidecalleridname=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
;
; 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
;
;restrictcid=no
;
; Whether or not use the caller ID presentation for the outgoing call that the
; calling switch is sending.
;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
;
; Also enables call parking (overrides the 'canpark' parameter)
;
; 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
;
; 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.
;
;mailbox=1234
;
;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)
;
; 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)
;
; increases the volume level on a channel, and a
; negavive value decreases volume level.
;
;rxgain=2.0
;txgain=3.0
;
; 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
;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
; cause the given audio file to
; be played upon completion of
; an attended transfer.
;
; Specify whether the channel should be answered immediately or if the simple
; switch should provide dialtone, read digits, etc.
;
;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
;
;callerid = 123456
;
;callerid = My Name <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
;
; 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
;
; 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
; 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.conf . This sets the tone zone by number.
; Note that you'd still need to load tonezones (loadzone in dahdi.conf).
;
; 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.
;
;mohinterpret=default
;
; This option specifies which music on hold class to suggest to the peer channel
; when this channel places the peer on hold.
;
;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.
; 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.
;
;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
;
;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
;sigalling=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
;
; 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.
; 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.
;
;ss7type = itu
; SS7 Called Nature of Address Indicator
;
;ss7_called_nai=dynamic
;
; SS7 Calling Nature of Address Indicator
;
;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 ----------------------------------------
; Configuration Sections
; ~~~~~~~~~~~~~~~~~~~~~~
; You can also configure channels in a separate 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
;phones
;echocancel = 64
;dahdichan = 1-8
;group = 1
;
; Is somewhat equivalent to the following snippet in the section
;
;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
; 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
;
group=1
context=inside-users
signalling=fxo_ks
callerid="Joe the Plomer" <123>
mailbox="123"
callwaiting=yes
threewaycalling=yes
transfer=yes
channel =>25
;
group=4
context=from_pstn
signalling=fxs_ks
callerid=asreceived
callwaiting=o
channel => 28
;
;configuration for PRI
group=5
context=from_pri_provider
echocancel=yes
echocancelwhenbridged=no
echotraining=yes
switchtype=5ess
signaling=pri_cpe
channel => 1-23
;
;
;
;
See also
Sample chan_dahdi.conf
;
;
; 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 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
;
; 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
; 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
; 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
;
;pridialplan=unknown
;prilocaldialplan=national
;
; pridialplan may be also set at dialtime, by prefixing the dialled number with
- one of the following letters
; 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)
; 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
;
;resetinterval = 3600
;
; Overlap dialing mode (sending overlap digits)
; Cannot be changed on a reload.
;
;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,
- 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
- featb
- Feature Group B (MF (domestic, US))
; 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
- fxs_tx
- Transmit audio/PTT on an FXS loopstart interface (FXO at the
- fxo_rx
- Receive audio/COR on an FXO loopstart interface (FXS at the
- fxo_tx
- Transmit audio/PTT on an FXO groundstart interface (FXS at
- 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
- 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
- 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
; 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
;
; Hide the name part and leave just the number part of the caller ID
; string. Only applies to PRI channels.
;hidecalleridname=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
;
; 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
;
;restrictcid=no
;
; Whether or not use the caller ID presentation for the outgoing call that the
; calling switch is sending.
- See README.callingpres. FIXME
- file no longer exists.
;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)
; Also enables call parking (overrides the 'canpark' parameter)
;
- For digital ports using ISDN PRI protocols
; 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
; 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=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
; 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.
; 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=2.0
;txgain=3.0
;
; 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
;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
; cause the given audio file to
; be played upon completion of
; an attended 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
;
;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
;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
; 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
- (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
; 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.conf . This sets the tone zone by number.
; Note that you'd still need to load tonezones (loadzone in dahdi.conf).
- The default is -1
- not to set anything.
;
; 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.
;
;mohinterpret=default
;
; This option specifies which music on hold class to suggest to the peer channel
; when this channel places the peer on hold.
;
;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.
; 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
;sigalling=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
; 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.
; 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
;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 ----------------------------------------
; Configuration Sections
; ~~~~~~~~~~~~~~~~~~~~~~
; You can also configure channels in a separate 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
;
;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
; 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
;
group=1
context=inside-users
signalling=fxo_ks
callerid="Joe the Plomer" <123>
mailbox="123"
callwaiting=yes
threewaycalling=yes
transfer=yes
channel =>25
;
group=4
context=from_pstn
signalling=fxs_ks
callerid=asreceived
callwaiting=o
channel => 28
;
;configuration for PRI
group=5
context=from_pri_provider
echocancel=yes
echocancelwhenbridged=no
echotraining=yes
switchtype=5ess
signaling=pri_cpe
channel => 1-23
;
;
;
;
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