Im curious why it dosent show dahdi show channels ? As below
pbx*CLI> dahdi show channels
Chan Extension Context Language MOH Interpret Blocked State
pseudo default default In Service
Below is the chan_dahdi.conf file.
root@pbx:~ $ cat /etc/asterisk/chan_dahdi.conf
; DAHDI Telephony Configuration file
; You need to restart Asterisk to re-configure the DAHDI channel
; CLI> module reload chan_dahdi.so
; will reload the configuration file, but not all configuration options
; are re-configured during a reload (signalling, as well as PRI and
; SS7-related settings cannot be changed on a reload).
; This file documents many configuration variables. Normally unless you know
; what a variable means or that it should be changed, there’s no reason to
; un-comment those lines.
; Examples below that are commented out (those lines that begin with a ‘;’ but
; no space afterwards) typically show a value that is not the default value,
; but would make sense under certain circumstances. The default values are
; usually sane. Thus you should typically not touch them unless you know what
; they mean or you know you should change them.
; Trunk groups are used for NFAS or GR-303 connections.
; Group: Defines a trunk group.
; trunkgroup => ,[,…]
; 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 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
; Default language
; Context for calls. Defaults to ‘default’
; 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
; 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.
; 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 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
; 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
; sample 2 for Germany
;internationalprefix = +
;nationalprefix = +49
;localprefix = +49711
;privateprefix = +497115678
; 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
; Allow inband audio (progress) when a call is RELEASEd by the far end of a PRI
; PRI Out of band indications.
; Enable this to report Busy and Congestion on a PRI using out-of-band
; notification. Inband indication, as used by Asterisk doesn’t seem to work
; with all telcos.
; outofband: Signal Busy/Congestion out of band with RELEASE/DISCONNECT
; inband: Signal Busy/Congestion using in-band tones (default)
; priindication cannot be changed on a reload.
;priindication = outofband
; If you need to override the existing channels selection routine and force all
; PRI channels to be marked as exclusively selected, set this to yes.
; priexclusive cannot be changed on a reload.
;priexclusive = yes
; If you need to use the logical channel mapping with your Q.SIG PRI instead
; of the physical mapping you must use the qsigchannelmapping option.
; logical: Use the logical channel mapping
; physical: Use physical channel mapping (default)
; If you wish to ignore remote hold indications (and use MOH that is supplied over
; the B channel) enable this option.
; 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
; mfcr2: MFC/R2 Signalling. To specify the country variant see ‘mfcr2_variant’
; The following are used for Radio interfaces:
; fxs_rx: Receive audio/COR on an FXS kewlstart interface (FXO at the
; channel bank)
; fxs_tx: Transmit audio/PTT on an FXS loopstart interface (FXO at the
; channel bank)
; fxo_rx: Receive audio/COR on an FXO loopstart interface (FXS at the
; channel bank)
; fxo_tx: Transmit audio/PTT on an FXO groundstart interface (FXS at
; the channel bank)
; em_rx: Receive audio/COR on an E&M interface (1-way)
; em_tx: Transmit audio/PTT on an E&M interface (1-way)
; em_txrx: Receive audio/COR AND Transmit audio/PTT on an E&M interface
; 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.
; 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.
; For Feature Group D Tandem access, to set the default CIC and OZZ use these
; parameters (Will not be updated on reload):
; 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.
; Whether or not to do distinctive ring detection on FXO lines:
; enable dring detection after caller ID for those countries like Australia
; where the ring cadence is changed after the caller ID spill:
; Whether or not to use caller ID:
; Hide the name part and leave just the number part of the caller ID
; string. Only applies to PRI channels.
; 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).
; 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)
; Whether or not to hide outgoing caller ID (Override with *67 or *82)
; (If your dialplan doesn’t catch it)
; On UK analog lines, the caller hanging up determines the end of calls. So
; Asterisk hanging up the line may or may not end a call (DAHDI could just as
; easily be re-attaching to a prior incoming call that was not yet hung up).
; This option changes the hangup to wait for a dialtone on the line, before
; marking the line as once again available for use with outgoing calls.
; The following option enables receiving MWI on FXO lines. The default
; value is no.
; The mwimonitor can take the following values
; no - No mwimonitoring occurs. (default)
; yes - The same as specifying fsk
; fsk - the FXO line is monitored for MWI FSK spills
; fsk,rpas - the FXO line is monitored for MWI FSK spills preceded
; by a ring pulse alert signal.
; neon - The fxo line is monitored for the presence of NEON pulses
; indicating MWI.
; When detected, an internal Asterisk MWI event is generated so that any other
; part of Asterisk that cares about MWI state changes is notified, just as if
; the state change came from app_voicemail.
; For FSK MWI Spills, the energy level that must be seen before starting the
; MWI detection process can be set with ‘mwilevel’.
; 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.
; The following keyword ‘mwisendtype’ enables various VMWI methods on FXS lines (if supported).
; The default is to send FSK only.
; The following options are available;
; ‘rpas’ Ring Pulse Alert Signal, alerts intelligent phones that a FSK message is about to be sent.
; ‘lrev’ Line reversed to indicate messages waiting.
; ‘hvdc’ 90Vdc OnHook DC voltage to indicate messages waiting.
; ‘hvac’ or ‘neon’ 90Vac OnHook AC voltage to light Neon bulb.
; ‘nofsk’ Disables FSK MWI spills from being sent out.
; It is feasible that multiple options can be enabled.
; 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.
; 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.
; 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.
; 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
; Support three-way calling
; 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
; Allow call parking
; (‘canpark=no’ is overridden by ‘transfer=yes’)
; Support call forward variable
; Whether or not to support Call Return (*69, if your dialplan doesn’t
; catch this first)
; 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:
; for any other voicemail context, the following will produce the stutter tone:
; 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.
; 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.
; 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.
; 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.
; 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.
; 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 .
; 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.
; 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.
; Channel variable to be set for all calls from this channel
;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
; ‘s’ priority 1 regardless of the dialed number!
; Specify whether flash-hook transfers to ‘busy’ channels should complete or
; return to the caller performing the transfer (default is yes).
; 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’.
; Channels may be associated with an account code to ease
; ADSI (Analog Display Services Interface) can be enabled on a per-channel
; basis if you have (or may have) ADSI compatible CPE equipment
; 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.
; 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.
; 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.
; 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 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.
; 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.
; 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.
; 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’)
; This feature can also easily detect false hangups. The symptoms of this is
; being disconnected in the middle of a call for no reason.
; 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
; 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.
; For FXO (FXS signalled) devices, whether to use pulse dial instead of DTMF
; Pulse digits from phones (FXS devices, FXO signalling) are always
; For fax detection, uncomment one of the following lines. The default is OFF
; When ‘faxdetect’ is used, one could use ‘faxbuffers’ to configure the DAHDI
; transmit buffer policy. The default is OFF. When this configuration
; option is used, the faxbuffer policy will be used for the life of the call
; after a fax tone is detected. The faxbuffer policy is reverted after the
; call is torn down. The sample below will result in 6 buffers and a full
; buffer policy.
; 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.
; 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.
; 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.
; Configure jitter buffers in DAHDI (each one is 20ms, default is 4)
; This is set globally, rather than per-channel.
;------------------------------ 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:
; 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.
;channel => 15
;channel => 16
; All those in group 0 I’ll use for outgoing calls
; Strip most significant digit (9) before sending
;channel => 45
;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.
; If no pattern is matched here is where we go.
;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 Calling Nature of Address Indicator
; unknown: Unknown
; subscriber: Subscriber
; national: National
; international: International
; dynamic: Dynamically selects the appropriate dialplan
; sample 1 for Germany
;ss7_internationalprefix = 00
;ss7_nationalprefix = 0
; 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.
; 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
; 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 ----------------------------------------
; ---------------- Options for use with signalling=mfcr2 --------------
; MFC-R2 signaling has lots of variants from country to country and even sometimes
; minor variants inside the same country. The only mandatory parameters here are:
; mfcr2_variant, mfcr2_max_ani and mfcr2_max_dnis.
; IT IS RECOMMENDED that you leave the default values (leaving it commented) for the
; other parameters unless you have problems or you have been instructed to change some
; parameter. OpenR2 library uses the mfcr2_variant parameter to try to determine the
; best defaults for your country, also refer to the OpenR2 package directory
; doc/asterisk/ where you can find sample configurations for some countries. If you
; want to contribute your configs for a particular country send them to the e-mail
; of the primary OpenR2 developer that you can find in the AUTHORS file of the OpenR2 package
; MFC/R2 variant. This depends on the OpenR2 supported variants
; A list of values can be found by executing the openr2 command r2test -l
; some valid values are:
; ar (Argentina)
; br (Brazil)
; mx (Mexico)
; ph (Philippines)
; itu (per ITU spec)
; Max amount of ANI to ask for
; Max amount of DNIS to ask for
; whether or not to get the ANI before getting DNIS.
; some telcos require ANI first some others do not care
; if this go wrong, change this value
; Caller Category to send
; usually national_subscriber works just fine
; you can change this setting from the dialplan
; by setting the variable MFCR2_CATEGORY
; (remember to set _MFCR2_CATEGORY from originating channels)
; MFCR2_CATEGORY will also be a variable available in your context
; on incoming calls set to the value received from the far end
; Call logging is stored at the Asterisk
; logging directory specified in asterisk.conf
; plus mfcr2/
; if you specify ‘span1’ here and asterisk.conf has
; as logging directory /var/log/asterisk then the full
; path to your MFC/R2 call logs will be /var/log/asterisk/mfcr2/span1
; (the directory will be automatically created if not present already)
; remember to set mfcr2_call_files=yes
; whether or not to drop call files into mfcr2_logdir
; MFC/R2 valid logging values are: all,error,warning,debug,notice,cas,mf,stack,nothing
; error,warning,debug and notice are self-descriptive
; ‘cas’ is for logging ABCD CAS tx and rx
; ‘mf’ is for logging of the Multi Frequency tones
; ‘stack’ is for very verbose output of the channel and context call stack, only useful
; if you are debugging a crash or want to learn how the library works. The stack logging
; will be only enabled if the openr2 library was compiled with -DOR2_TRACE_STACKS
; You can mix up values, like: loglevel=error,debug,mf to log just error, debug and
; multi frequency messages
; ‘all’ is a special value to log all the activity
; ‘nothing’ is a clean-up value, in case you want to not log any activity for
; a channel or group of channels
; BE AWARE that the level of output logged will ALSO depend on
; the value you have in logger.conf, if you disable output in logger.conf
; then it does not matter you specify ‘all’ here, nothing will be logged
; so logger.conf has the last word on what is going to be logged
; MFC/R2 value in milliseconds for the MF timeout. Any negative value
; means ‘default’, smaller values than 500ms are not recommended
; and can cause malfunctioning. If you experience protocol error
; due to MF timeout try incrementing this value in 500ms steps
; MFC/R2 value in milliseconds for the metering pulse timeout.
; Metering pulses are sent by some telcos for some R2 variants
; during a call presumably for billing purposes to indicate costs,
; however this pulses use the same signal that is used to indicate
; call hangup, therefore a timeout is sometimes required to distinguish
; between a real hangup and a billing pulse that should not
; last more than 500ms, If you experience call drops after some
; minutes of being stablished try setting a value of some ms here,
; values greater than 500ms are not recommended.
; BE AWARE that choosing the proper protocol mfcr2_variant parameter
; implicitly sets a good recommended value for this timer, use this
; parameter only when you really want to override the default, otherwise
; just comment out this value or put a -1
; Any negative value means ‘default’.
; Brazil uses a special calling party category for collect calls (llamadas por cobrar)
; instead of using the operator (as in Mexico). The R2 spec in Brazil says a special GB tone
; should be used to reject collect calls. If you want to ALLOW collect calls specify ‘yes’,
; if you want to BLOCK collect calls then say ‘no’. Default is to block collect calls.
; (see also ‘mfcr2_double_answer’)
; This feature is related but independent of mfcr2_allow_collect_calls
; Some PBX’s require a double-answer process to block collect calls, if
; you ever have problems blocking collect calls using Group B signals (mfcr2_allow_collect_calls=no)
; then you may want to try with mfcr2_double_answer=yes, this will cause that every answer signal
; is changed by answer->clear back->answer (sort of a flash)
; (see also ‘mfcr2_allow_collect_calls’)
; This feature allows to skip the use of Group B/II signals and go directly
; to the accepted state for incoming calls
; You most likely dont need this feature. Default is yes.
; When this is set to yes, all calls that are offered (incoming calls) which
; DNIS is valid (exists in extensions.conf) and pass collect call validation
; will be accepted with a Group B tone (either call with charge or not, depending on mfcr2_charge_calls)
; with this set to ‘no’ then the call will NOT be accepted on offered, and the call will start its
; execution in extensions.conf without being accepted until the channel is answered (either with Answer() or
; any other application resulting in the channel being answered).
; This can be set to ‘no’ if your telco or PBX needs the hangup cause to be set accurately
; when this option is set to no you must explicitly accept the call with DAHDIAcceptR2Call
; or implicitly through the Answer() application.
; Skip request of calling party category and ANI
; you need openr2 >= 1.2.0 to use this feature
; WARNING: advanced users only! I really mean it
; this parameter is commented by default because
; YOU DON’T NEED IT UNLESS YOU REALLY GROK MFC/R2
; READ COMMENTS on doc/r2proto.conf in openr2 package
; for more info
; Brazil use a special signal to force the release of the line (hangup) from the
; backward perspective. When mfcr2_forced_release=no, the normal clear back signal
; will be sent on hangup, which is OK for all mfcr2 variants I know of, except for
; Brazilian variant, where the central will leave the line up for several seconds (30, 60)
; which sometimes is not what people really want. When mfcr2_forced_release=yes, a different
; signal will be sent to hangup the call indicating that the line should be released immediately
; Whether or not report to the other end ‘accept call with charge’
; This setting has no effect with most telecos, usually is safe
; leave the default (yes), but once in a while when interconnecting with
; old PBXs this may be useful.
; Concretely this affects the Group B signal used to accept calls
; The application DAHDIAcceptR2Call can also be used to decide this
; in the dial plan in a per-call basis instead of doing it here for all calls
; ---------------- END of options to be used with signalling=mfcr2
; Configuration Sections
; You can also configure channels in a separate chan_dahdi.conf section. In
; this case the keyword ‘channel’ is not used. Instead the keyword
; ‘dahdichan’ is used (as in users.conf) - configuration is only processed
; in a section where the keyword dahdichan is used. It will only be
; processed in the end of the section. Thus the following section:
;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
; chan_dahdi.conf and [general] in users.conf - one section’s configuration
; does not affect another one’s.
; Instead of letting common configuration values “slide through” you can
; use configuration templates to easily keep the common part in one
; place and override where needed.
;echocancel = yes
;group = 0,4
;callgroup = 3
;pickupgroup = 3
;threewaycalling = yes
;transfer = yes
;context = phones
;faxdetect = incoming
;dahdichan = 1
;callerid = My Name <501>
;mailbox = 501@mailboxes
;dahdichan = 2
;faxdetect = no
;context = fax
;dahdichan = 3
;pickupgroup = 3,4
Please advice …