From 8f216ea83af04ed5970b13bb1169d0e32801e229 Mon Sep 17 00:00:00 2001 From: Jeff Peeler Date: Mon, 30 Jun 2008 22:34:08 +0000 Subject: rename zapata.conf.sample to chan_dahdi.conf.sample git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@126675 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- configs/chan_dahdi.conf.sample | 981 +++++++++++++++++++++++++++++++++++++++++ configs/zapata.conf.sample | 981 ----------------------------------------- 2 files changed, 981 insertions(+), 981 deletions(-) create mode 100644 configs/chan_dahdi.conf.sample delete mode 100644 configs/zapata.conf.sample (limited to 'configs') diff --git a/configs/chan_dahdi.conf.sample b/configs/chan_dahdi.conf.sample new file mode 100644 index 000000000..f08bca9ab --- /dev/null +++ b/configs/chan_dahdi.conf.sample @@ -0,0 +1,981 @@ +; +; DAHDI telephony +; +; Configuration file +; +; You need to restart Asterisk to re-configure the DAHDI channel +; CLI> reload chan_dahdi.so +; will reload the configuration file, +; but not all configuration options are +; re-configured during a reload (signalling, as well as +; PRI and SS7-related settings cannot be changed on a +; reload. +; +; This file documents many configuration variables. Normally unless you +; know what a variable means or that it should be changed, there's no +; reason to unrem lines. +; +; remmed-out examples below (those lines that begin with a ';' but no +; space afterwards) typically show a value that is not the defauult value, +; but would make sense under cetain circumstances. The default values +; are usually sane. Thus you should typically not touch them unless you +; know what they mean or you know you should change them. + + +[trunkgroups] +; +; Trunk groups are used for NFAS or GR-303 connections. +; +; Group: Defines a trunk group. +; trunkgroup => ,[,...] +; +; trunkgroup 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 + +[channels] +; +; Default language +; +;language=en +; +; Context for calls. Defaults to 'default' +; +;context=incoming +; +; Switchtype: Only used for PRI. +; +; national: National ISDN 2 (default) +; dms100: Nortel DMS100 +; 4ess: AT&T 4ESS +; 5ess: Lucent 5ESS +; euroisdn: EuroISDN (common in Europe) +; ni1: Old National ISDN 1 +; qsig: Q.SIG +; +;switchtype=euroisdn +; +; Some switches (AT&T especially) require network specific facility IE +; supported values are currently 'none', 'sdn', 'megacom', 'tollfreemegacom', 'accunet' +; +; nsf cannot be changed on a reload. +; +;nsf=none +; +; PRI Dialplan: The ISDN-level Type Of Number (TON) or numbering plan, used for +; the dialed number. For most installations, leaving this as 'unknown' (the +; default) works in the most cases. In some very unusual circumstances, you +; may need to set this to 'dynamic' or 'redundant'. Note that if you set one +; of the others, you will be unable to dial another class of numbers. For +; example, if you set 'national', you will be unable to dial local or +; international numbers. +; +; PRI Local Dialplan: Only RARELY used for PRI (sets the calling number's +; numbering plan). In North America, the typical use is sending the 10 digit +; callerID number and setting the prilocaldialplan to 'national' (the default). +; Only VERY rarely will you need to change this. +; +; Neither pridialplan nor prilocaldialplan can be changed on reload. +; +; unknown: Unknown +; private: Private ISDN +; local: Local ISDN +; national: National ISDN +; international: International ISDN +; dynamic: Dynamically selects the appropriate dialplan +; redundant: Same as dynamic, except that the underlying number is not +; changed (not common) +; +;pridialplan=unknown +;prilocaldialplan=national +; +; pridialplan may be also set at dialtime, by prefixing the dialled number with +; one of the following letters: +; U - Unknown +; I - International +; N - National +; L - Local (Net Specific) +; S - Subscriber +; V - Abbreviated +; R - Reserved (should probably never be used but is included for completeness) +; +; Additionally, you may also set the following NPI bits (also by prefixing the +; dialled string with one of the following letters): +; u - Unknown +; e - E.163/E.164 (ISDN/telephony) +; x - X.121 (Data) +; f - F.69 (Telex) +; n - National +; p - Private +; r - Reserved (should probably never be used but is included for completeness) +; +; You may also set the prilocaldialplan in the same way, but by prefixing the +; Caller*ID Number, rather than the dialled number. Please note that telcos +; which require this kind of additional manipulation of the TON/NPI are *rare*. +; Most telco PRIs will work fine simply by setting pridialplan to unknown or +; dynamic. +; +; +; PRI caller ID prefixes based on the given TON/NPI (dialplan) +; This is especially needed for EuroISDN E1-PRIs +; +; None of the prefix settings can be changed on reload. +; +; sample 1 for Germany +;internationalprefix = 00 +;nationalprefix = 0 +;localprefix = 0711 +;privateprefix = 07115678 +;unknownprefix = +; +; sample 2 for Germany +;internationalprefix = + +;nationalprefix = +49 +;localprefix = +49711 +;privateprefix = +497115678 +;unknownprefix = +; +; PRI resetinterval: sets the time in seconds between restart of unused +; B channels; defaults to 'never'. +; +;resetinterval = 3600 +; +; Overlap dialing mode (sending overlap digits) +; Cannot be changed on a reload. +; +;overlapdial=yes +; +; PRI Out of band indications. +; Enable this to report Busy and Congestion on a PRI using out-of-band +; notification. Inband indication, as used by Asterisk doesn't seem to work +; with all telcos. +; +; outofband: Signal Busy/Congestion out of band with RELEASE/DISCONNECT +; inband: Signal Busy/Congestion using in-band tones (default) +; +; priindication cannot be changed on a reload. +; +;priindication = outofband +; +; If you need to override the existing channels selection routine and force all +; PRI channels to be marked as exclusively selected, set this to yes. +; +; priexclusive cannot be changed on a reload. +; +;priexclusive = yes +; +; ISDN Timers +; All of the ISDN timers and counters that are used are configurable. Specify +; the timer name, and its value (in ms for timers). +; K: Layer 2 max number of outstanding unacknowledged I frames (default 7) +; N200: Layer 2 max number of retransmissions of a frame (default 3) +; T200: Layer 2 max time before retransmission of a frame (default 1000 ms) +; T203: Layer 2 max time without frames being exchanged (default 10000 ms) +; T305: Wait for DISCONNECT acknowledge (default 30000 ms) +; T308: Wait for RELEASE acknowledge (default 4000 ms) +; T309: Maintain active calls on Layer 2 disconnection (default -1, +; Asterisk clears calls) +; EuroISDN: 6000 to 12000 ms, according to (N200 + 1) x T200 + 2s +; May vary in other ISDN standards (Q.931 1993 : 90000 ms) +; T313: Wait for CONNECT acknowledge, CPE side only (default 3000 ms) +; +;pritimer => t200,1000 +;pritimer => t313,4000 +; +; To enable transmission of facility-based ISDN supplementary services (such +; as caller name from CPE over facility), enable this option. +; Cannot be changed on a reload. +; +;facilityenable = yes +; +; pritimer cannot be changed on a reload. +; +; Signalling method. The default is "auto". Valid values: +; auto: Use the current value from DAHDI. +; em: E & M +; em_e1: E & M E1 +; em_w: E & M Wink +; featd: Feature Group D (The fake, Adtran style, DTMF) +; featdmf: Feature Group D (The real thing, MF (domestic, US)) +; featdmf_ta: Feature Group D (The real thing, MF (domestic, US)) through +; a Tandem Access point +; featb: Feature Group B (MF (domestic, US)) +; fgccama Feature Group C-CAMA (DP DNIS, MF ANI) +; fgccamamf Feature Group C-CAMA MF (MF DNIS, MF ANI) +; fxs_ls: FXS (Loop Start) +; fxs_gs: FXS (Ground Start) +; fxs_ks: FXS (Kewl Start) +; fxo_ls: FXO (Loop Start) +; fxo_gs: FXO (Ground Start) +; fxo_ks: FXO (Kewl Start) +; pri_cpe: PRI signalling, CPE side +; pri_net: PRI signalling, Network side +; gr303fxoks_net: GR-303 Signalling, FXO Loopstart, Network side +; gr303fxsks_cpe: GR-303 Signalling, FXS Loopstart, CPE side +; sf: SF (Inband Tone) Signalling +; sf_w: SF Wink +; sf_featd: SF Feature Group D (The fake, Adtran style, DTMF) +; sf_featdmf: SF Feature Group D (The real thing, MF (domestic, US)) +; sf_featb: SF Feature Group B (MF (domestic, US)) +; e911: E911 (MF) style signalling +; ss7: Signalling System 7 +; +; The following are used for Radio interfaces: +; fxs_rx: Receive audio/COR on an FXS kewlstart interface (FXO at the +; channel bank) +; fxs_tx: Transmit audio/PTT on an FXS loopstart interface (FXO at the +; channel bank) +; fxo_rx: Receive audio/COR on an FXO loopstart interface (FXS at the +; channel bank) +; fxo_tx: Transmit audio/PTT on an FXO groundstart interface (FXS at +; the channel bank) +; em_rx: Receive audio/COR on an E&M interface (1-way) +; em_tx: Transmit audio/PTT on an E&M interface (1-way) +; em_txrx: Receive audio/COR AND Transmit audio/PTT on an E&M interface +; (2-way) +; em_rxtx: Same as em_txrx (for our dyslexic friends) +; sf_rx: Receive audio/COR on an SF interface (1-way) +; sf_tx: Transmit audio/PTT on an SF interface (1-way) +; sf_txrx: Receive audio/COR AND Transmit audio/PTT on an SF interface +; (2-way) +; sf_rxtx: Same as sf_txrx (for our dyslexic friends) +; ss7: Signalling System 7 +; +; signalling of a channel can not be changed on a reload. +; +;signalling=fxo_ls +; +; If you have an outbound signalling format that is different from format +; specified above (but compatible), you can specify outbound signalling format, +; (see below). The 'signalling' format specified will be the inbound signalling +; format. If you only specify 'signalling', then it will be the format for +; both inbound and outbound. +; +; outsignalling can only be one of: +; em, em_e1, em_w, sf, sf_w, sf_featd, sf_featdmf, sf_featb, featd, +; featdmf, featdmf_ta, e911, fgccama, fgccamamf +; +; outsignalling cannot be changed on a reload. +; +;signalling=featdmf +; +;outsignalling=featb +; +; For Feature Group D Tandem access, to set the default CIC and OZZ use these +; parameters (Will not be updated on reload): +; +;defaultozz=0000 +;defaultcic=303 +; +; A variety of timing parameters can be specified as well +; The default values for those are "-1", which is to use the +; compile-time defaults of the DAHDI kernel modules. The timing +; parameters, (with the standard default from DAHDI): +; +; prewink: Pre-wink time (default 50ms) +; preflash: Pre-flash time (default 50ms) +; wink: Wink time (default 150ms) +; flash: Flash time (default 750ms) +; start: Start time (default 1500ms) +; rxwink: Receiver wink time (default 300ms) +; rxflash: Receiver flashtime (default 1250ms) +; debounce: Debounce timing (default 600ms) +; +; None of them will update on a reload. +; +; How long generated tones (DTMF and MF) will be played on the channel +; (in milliseconds). +; +; This is a global, rather than a per-channel setting. It will not be +; updated on a reload. +; +;toneduration=100 +; +; Whether or not to do distinctive ring detection on FXO lines: +; +;usedistinctiveringdetection=yes +; +; enable dring detection after caller ID for those countries like Australia +; where the ring cadence is changed *after* the caller ID spill: +; +;distinctiveringaftercid=yes +; +; Whether or not to use caller ID: +; +usecallerid=yes +; +; 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): +; Support flash-hook call transfer (requires three way calling) +; Also enables call parking (overrides the 'canpark' parameter) +; +; For digital ports using ISDN PRI protocols: +; Support switch-side transfer (called 2BCT, RLT or other names) +; This setting must be enabled on both ports involved, and the +; 'facilityenable' setting must also be enabled to allow sending +; the transfer to the ISDN switch, since it sent in a FACILITY +; message. +; +transfer=yes +; +; Allow call parking +; ('canpark=no' is overridden by 'transfer=yes') +; +canpark=yes +; +; Support call forward variable +; +cancallforward=yes +; +; Whether or not to support Call Return (*69, if your dialplan doesn't +; catch this first) +; +callreturn=yes +; +; Stutter dialtone support: If a mailbox is specified without a voicemail +; context, then when voicemail is received in a mailbox in the default +; voicemail context in voicemail.conf, taking the phone off hook will cause a +; stutter dialtone instead of a normal one. +; +; If a mailbox is specified *with* a voicemail context, the same will result +; if voicemail received in mailbox in the specified voicemail context. +; +; for default voicemail context, the example below is fine: +; +;mailbox=1234 +; +; for any other voicemail context, the following will produce the stutter tone: +; +;mailbox=1234@context +; +; Enable echo cancellation +; Use either "yes", "no", or a power of two from 32 to 256 if you wish to +; actually set the number of taps of cancellation. +; +; Note that when setting the number of taps, the number 256 does not translate +; to 256 ms of echo cancellation. echocancel=256 means 256 / 8 = 32 ms. +; +; Note that if any of your DAHDI cards have hardware echo cancellers, +; then this setting only turns them on and off; numeric settings will +; be treated as "yes". There are no special settings required for +; hardware echo cancellers; when present and enabled in their kernel +; modules, they take precedence over the software echo canceller compiled +; into DAHDI automatically. +; +; +echocancel=yes +; +; As of Zaptel 1.4.8, some DAHDI echo cancellers (software and hardware) +; support adjustable parameters; these parameters can be supplied as +; additional options to the 'echocancel' setting. Note that Asterisk +; does not attempt to validate the parameters or their values, so if you +; supply an invalid parameter you will not know the specific reason it +; failed without checking the kernel message log for the error(s) +; put there by DAHDI. +; +;echocancel=128,param1=32,param2=0,param3=14 +; +; Generally, it is not necessary (and in fact undesirable) to echo cancel when +; the circuit path is entirely TDM. You may, however, change this behavior +; by enabling the echo canceller during pure TDM bridging below. +; +echocancelwhenbridged=yes +; +; In some cases, the echo canceller doesn't train quickly enough and there +; is echo at the beginning of the call. Enabling echo training will cause +; DAHDI to briefly mute the channel, send an impulse, and use the impulse +; response to pre-train the echo canceller so it can start out with a much +; closer idea of the actual echo. Value may be "yes", "no", or a number of +; milliseconds to delay before training (default = 400) +; +; WARNING: In some cases this option can make echo worse! If you are +; trying to debug an echo problem, it is worth checking to see if your echo +; is better with the option set to yes or no. Use whatever setting gives +; the best results. +; +; Note that these parameters do not apply to hardware echo cancellers. +; +;echotraining=yes +;echotraining=800 +; +; If you are having trouble with DTMF detection, you can relax the DTMF +; detection parameters. Relaxing them may make the DTMF detector more likely +; to have "talkoff" where DTMF is detected when it shouldn't be. +; +;relaxdtmf=yes +; +; You may also set the default receive and transmit gains (in dB) +; +; Gain Settings: increasing / decreasing the volume level on a channel. +; The values are in db (decibells). A positive number +; increases the volume level on a channel, and a +; negavive value decreases volume level. +; +; There are several independent gain settings: +; rxgain: gain for the rx (receive - into Asterisk) channel. Default: 0.0 +; txgain: gain for the tx (transmit - out of Asterisk Asterisk) channel. +; Default: 0.0 +; cid_rxgain: set the gain just for the caller ID sounds Asterisk +; emits. Default: 5.0 . + +;rxgain=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 + +; +; Specify whether the channel should be answered immediately or if the simple +; switch should provide dialtone, read digits, etc. +; Note: If immediate=yes the dialplan execution will always start at extension +; 's' priority 1 regardless of the dialed number! +; +;immediate=yes +; +; Specify whether flash-hook transfers to 'busy' channels should complete or +; return to the caller performing the transfer (default is yes). +; +;transfertobusy=no +; +; caller ID can be set to "asreceived" or a specific number if you want to +; override it. Note that "asreceived" only applies to trunk interfaces. +; fullname sets just the +; +; fullname: sets just the name part. +; cid_number: sets just the number part: +; +;callerid = 123456 +; +;callerid = My Name <2564286000> +; Which can also be written as: +;cid_number = 2564286000 +;fullname = My Name +; +;callerid = asreceived +; +; should we use the caller ID from incoming call on DAHDI transfer? +; +;useincomingcalleridondahditransfer = yes +; +; AMA flags affects the recording of Call Detail Records. If specified +; it may be 'default', 'omit', 'billing', or 'documentation'. +; +;amaflags=default +; +; Channels may be associated with an account code to ease +; billing +; +;accountcode=lss0101 +; +; ADSI (Analog Display Services Interface) can be enabled on a per-channel +; basis if you have (or may have) ADSI compatible CPE equipment +; +;adsi=yes +; +; SMDI (Simplified Message Desk Interface) can be enabled on a per-channel +; basis if you would like that channel to behave like an SMDI message desk. +; The SMDI port specified should have already been defined in smdi.conf. The +; default port is /dev/ttyS0. +; +;usesmdi=yes +;smdiport=/dev/ttyS0 +; +; On trunk interfaces (FXS) and E&M interfaces (E&M, Wink, Feature Group D +; etc, it can be useful to perform busy detection either in an effort to +; detect hangup or for detecting busies. This enables listening for +; the beep-beep busy pattern. +; +;busydetect=yes +; +; If busydetect is enabled, it is also possible to specify how many busy tones +; to wait for before hanging up. The default is 3, but it might be +; safer to set to 6 or even 8. Mind that the higher the number, the more +; time that will be needed to hangup a channel, but lowers the probability +; that you will get random hangups. +; +;busycount=6 +; +; If busydetect is enabled, it is also possible to specify the cadence of your +; busy signal. In many countries, it is 500msec on, 500msec off. Without +; busypattern specified, we'll accept any regular sound-silence pattern that +; repeats times as a busy signal. If you specify busypattern, +; then we'll further check the length of the sound (tone) and silence, which +; will further reduce the chance of a false positive. +; +;busypattern=500,500 +; +; NOTE: In make menuselect, you'll find further options to tweak the busy +; detector. If your country has a busy tone with the same length tone and +; silence (as many countries do), consider enabling the +; BUSYDETECT_COMPARE_TONE_AND_SILENCE option. +; +; To further detect which hangup tone your telco provider is sending, it is +; useful to use the ztmonitor utility to record the audio that main/dsp.c +; is receiving after the caller hangs up. +; +; Use a polarity reversal to mark when a outgoing call is answered by the +; remote party. +; +;answeronpolarityswitch=yes +; +; In some countries, a polarity reversal is used to signal the disconnect of a +; phone line. If the hanguponpolarityswitch option is selected, the call will +; be considered "hung up" on a polarity reversal. +; +;hanguponpolarityswitch=yes +; +; polarityonanswerdelay: minimal time period (ms) between the answer +; polarity switch and hangup polarity switch. +; (default: 600ms) +; +; On trunk interfaces (FXS) it can be useful to attempt to follow the progress +; of a call through RINGING, BUSY, and ANSWERING. If turned on, call +; progress attempts to determine answer, busy, and ringing on phone lines. +; This feature is HIGHLY EXPERIMENTAL and can easily detect false answers, +; so don't count on it being very accurate. +; +; Few zones are supported at the time of this writing, but may be selected +; with "progzone". +; +; progzone also affects the pattern used for buzydetect (unless +; busypattern is set explicitly). The possible values are: +; us (default) +; ca (alias for 'us') +; cr (Costa Rica) +; br (Brazil, alias for 'cr') +; uk +; +; This feature can also easily detect false hangups. The symptoms of this is +; being disconnected in the middle of a call for no reason. +; +;callprogress=yes +;progzone=uk +; +; Set the tonezone. Equivalent of the defaultzone settings in +; /etc/dahdi.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. +;tonezone = 0 ; 0 is US +; +; FXO (FXS signalled) devices must have a timeout to determine if there was a +; hangup before the line was answered. This value can be tweaked to shorten +; how long it takes before DAHDI considers a non-ringing line to have hungup. +; +; ringtimeout will not update on a reload. +; +;ringtimeout=8000 +; +; For FXO (FXS signalled) devices, whether to use pulse dial instead of DTMF +; Pulse digits from phones (FXS devices, FXO signalling) are always +; detected. +; +;pulsedial=yes +; +; For fax detection, uncomment one of the following lines. The default is *OFF* +; +;faxdetect=both +;faxdetect=incoming +;faxdetect=outgoing +;faxdetect=no +; +; This option specifies a preference for which music on hold class this channel +; should listen to when put on hold if the music class has not been set on the +; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer +; channel putting this one on hold did not suggest a music class. +; +; If this option is set to "passthrough", then the hold message will always be +; passed through as signalling instead of generating hold music locally. This +; setting is only valid when used on a channel that uses digital signalling. +; +;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 +; pri_cpe or pri_net for CPE or Network termination, and generally you will +; want to create a single "group" for all channels of the PRI. +; +; switchtype cannot be changed on a reload. +; +; switchtype = national +; signalling = pri_cpe +; group = 2 +; channel => 1-23 + +; + +; Used for distinctive ring support for x100p. +; You can see the dringX patterns is to set any one of the dringXcontext fields +; and they will be printed on the console when an inbound call comes in. +; +; dringXrange is used to change the acceptable ranges for "tone offsets". Defaults to 10. +; Note: a range of 0 is NOT what you might expect - it instead forces it to the default. +; A range of -1 will force it to always match. +; Anything lower than -1 would presumably cause it to never match. +; +;dring1=95,0,0 +;dring1context=internal1 +;dring1range=10 +;dring2=325,95,0 +;dring2context=internal2 +;dring2range=10 +; If no pattern is matched here is where we go. +;context=default +;channel => 1 + +; ---------------- Options for use with signalling=ss7 ----------------- +; None of them can be changed by a reload. +; +; Variant of SS7 signalling: +; Options are itu and ansi +;ss7type = itu + +; SS7 Called Nature of Address Indicator +; +; unknown: Unknown +; subscriber: Subscriber +; national: National +; international: International +; dynamic: Dynamically selects the appropriate dialplan +; +;ss7_called_nai=dynamic +; +; SS7 Calling Nature of Address Indicator +; +; unknown: Unknown +; subscriber: Subscriber +; national: National +; international: International +; dynamic: Dynamically selects the appropriate dialplan +; +;ss7_calling_nai=dynamic +; +; +; sample 1 for Germany +;ss7_internationalprefix = 00 +;ss7_nationalprefix = 0 +;ss7_subscriberprefix = +;ss7_unknownprefix = +; + +; 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 + +; 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 +; [channels]: +; +;echocancel = 64 +;group = 1 +;channel => 1-8 +; +; When starting a new section almost all of the configuration values are +; copied from their values at the end of the section [channels] in +; 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 diff --git a/configs/zapata.conf.sample b/configs/zapata.conf.sample deleted file mode 100644 index f08bca9ab..000000000 --- a/configs/zapata.conf.sample +++ /dev/null @@ -1,981 +0,0 @@ -; -; DAHDI telephony -; -; Configuration file -; -; You need to restart Asterisk to re-configure the DAHDI channel -; CLI> reload chan_dahdi.so -; will reload the configuration file, -; but not all configuration options are -; re-configured during a reload (signalling, as well as -; PRI and SS7-related settings cannot be changed on a -; reload. -; -; This file documents many configuration variables. Normally unless you -; know what a variable means or that it should be changed, there's no -; reason to unrem lines. -; -; remmed-out examples below (those lines that begin with a ';' but no -; space afterwards) typically show a value that is not the defauult value, -; but would make sense under cetain circumstances. The default values -; are usually sane. Thus you should typically not touch them unless you -; know what they mean or you know you should change them. - - -[trunkgroups] -; -; Trunk groups are used for NFAS or GR-303 connections. -; -; Group: Defines a trunk group. -; trunkgroup => ,[,...] -; -; trunkgroup 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 - -[channels] -; -; Default language -; -;language=en -; -; Context for calls. Defaults to 'default' -; -;context=incoming -; -; Switchtype: Only used for PRI. -; -; national: National ISDN 2 (default) -; dms100: Nortel DMS100 -; 4ess: AT&T 4ESS -; 5ess: Lucent 5ESS -; euroisdn: EuroISDN (common in Europe) -; ni1: Old National ISDN 1 -; qsig: Q.SIG -; -;switchtype=euroisdn -; -; Some switches (AT&T especially) require network specific facility IE -; supported values are currently 'none', 'sdn', 'megacom', 'tollfreemegacom', 'accunet' -; -; nsf cannot be changed on a reload. -; -;nsf=none -; -; PRI Dialplan: The ISDN-level Type Of Number (TON) or numbering plan, used for -; the dialed number. For most installations, leaving this as 'unknown' (the -; default) works in the most cases. In some very unusual circumstances, you -; may need to set this to 'dynamic' or 'redundant'. Note that if you set one -; of the others, you will be unable to dial another class of numbers. For -; example, if you set 'national', you will be unable to dial local or -; international numbers. -; -; PRI Local Dialplan: Only RARELY used for PRI (sets the calling number's -; numbering plan). In North America, the typical use is sending the 10 digit -; callerID number and setting the prilocaldialplan to 'national' (the default). -; Only VERY rarely will you need to change this. -; -; Neither pridialplan nor prilocaldialplan can be changed on reload. -; -; unknown: Unknown -; private: Private ISDN -; local: Local ISDN -; national: National ISDN -; international: International ISDN -; dynamic: Dynamically selects the appropriate dialplan -; redundant: Same as dynamic, except that the underlying number is not -; changed (not common) -; -;pridialplan=unknown -;prilocaldialplan=national -; -; pridialplan may be also set at dialtime, by prefixing the dialled number with -; one of the following letters: -; U - Unknown -; I - International -; N - National -; L - Local (Net Specific) -; S - Subscriber -; V - Abbreviated -; R - Reserved (should probably never be used but is included for completeness) -; -; Additionally, you may also set the following NPI bits (also by prefixing the -; dialled string with one of the following letters): -; u - Unknown -; e - E.163/E.164 (ISDN/telephony) -; x - X.121 (Data) -; f - F.69 (Telex) -; n - National -; p - Private -; r - Reserved (should probably never be used but is included for completeness) -; -; You may also set the prilocaldialplan in the same way, but by prefixing the -; Caller*ID Number, rather than the dialled number. Please note that telcos -; which require this kind of additional manipulation of the TON/NPI are *rare*. -; Most telco PRIs will work fine simply by setting pridialplan to unknown or -; dynamic. -; -; -; PRI caller ID prefixes based on the given TON/NPI (dialplan) -; This is especially needed for EuroISDN E1-PRIs -; -; None of the prefix settings can be changed on reload. -; -; sample 1 for Germany -;internationalprefix = 00 -;nationalprefix = 0 -;localprefix = 0711 -;privateprefix = 07115678 -;unknownprefix = -; -; sample 2 for Germany -;internationalprefix = + -;nationalprefix = +49 -;localprefix = +49711 -;privateprefix = +497115678 -;unknownprefix = -; -; PRI resetinterval: sets the time in seconds between restart of unused -; B channels; defaults to 'never'. -; -;resetinterval = 3600 -; -; Overlap dialing mode (sending overlap digits) -; Cannot be changed on a reload. -; -;overlapdial=yes -; -; PRI Out of band indications. -; Enable this to report Busy and Congestion on a PRI using out-of-band -; notification. Inband indication, as used by Asterisk doesn't seem to work -; with all telcos. -; -; outofband: Signal Busy/Congestion out of band with RELEASE/DISCONNECT -; inband: Signal Busy/Congestion using in-band tones (default) -; -; priindication cannot be changed on a reload. -; -;priindication = outofband -; -; If you need to override the existing channels selection routine and force all -; PRI channels to be marked as exclusively selected, set this to yes. -; -; priexclusive cannot be changed on a reload. -; -;priexclusive = yes -; -; ISDN Timers -; All of the ISDN timers and counters that are used are configurable. Specify -; the timer name, and its value (in ms for timers). -; K: Layer 2 max number of outstanding unacknowledged I frames (default 7) -; N200: Layer 2 max number of retransmissions of a frame (default 3) -; T200: Layer 2 max time before retransmission of a frame (default 1000 ms) -; T203: Layer 2 max time without frames being exchanged (default 10000 ms) -; T305: Wait for DISCONNECT acknowledge (default 30000 ms) -; T308: Wait for RELEASE acknowledge (default 4000 ms) -; T309: Maintain active calls on Layer 2 disconnection (default -1, -; Asterisk clears calls) -; EuroISDN: 6000 to 12000 ms, according to (N200 + 1) x T200 + 2s -; May vary in other ISDN standards (Q.931 1993 : 90000 ms) -; T313: Wait for CONNECT acknowledge, CPE side only (default 3000 ms) -; -;pritimer => t200,1000 -;pritimer => t313,4000 -; -; To enable transmission of facility-based ISDN supplementary services (such -; as caller name from CPE over facility), enable this option. -; Cannot be changed on a reload. -; -;facilityenable = yes -; -; pritimer cannot be changed on a reload. -; -; Signalling method. The default is "auto". Valid values: -; auto: Use the current value from DAHDI. -; em: E & M -; em_e1: E & M E1 -; em_w: E & M Wink -; featd: Feature Group D (The fake, Adtran style, DTMF) -; featdmf: Feature Group D (The real thing, MF (domestic, US)) -; featdmf_ta: Feature Group D (The real thing, MF (domestic, US)) through -; a Tandem Access point -; featb: Feature Group B (MF (domestic, US)) -; fgccama Feature Group C-CAMA (DP DNIS, MF ANI) -; fgccamamf Feature Group C-CAMA MF (MF DNIS, MF ANI) -; fxs_ls: FXS (Loop Start) -; fxs_gs: FXS (Ground Start) -; fxs_ks: FXS (Kewl Start) -; fxo_ls: FXO (Loop Start) -; fxo_gs: FXO (Ground Start) -; fxo_ks: FXO (Kewl Start) -; pri_cpe: PRI signalling, CPE side -; pri_net: PRI signalling, Network side -; gr303fxoks_net: GR-303 Signalling, FXO Loopstart, Network side -; gr303fxsks_cpe: GR-303 Signalling, FXS Loopstart, CPE side -; sf: SF (Inband Tone) Signalling -; sf_w: SF Wink -; sf_featd: SF Feature Group D (The fake, Adtran style, DTMF) -; sf_featdmf: SF Feature Group D (The real thing, MF (domestic, US)) -; sf_featb: SF Feature Group B (MF (domestic, US)) -; e911: E911 (MF) style signalling -; ss7: Signalling System 7 -; -; The following are used for Radio interfaces: -; fxs_rx: Receive audio/COR on an FXS kewlstart interface (FXO at the -; channel bank) -; fxs_tx: Transmit audio/PTT on an FXS loopstart interface (FXO at the -; channel bank) -; fxo_rx: Receive audio/COR on an FXO loopstart interface (FXS at the -; channel bank) -; fxo_tx: Transmit audio/PTT on an FXO groundstart interface (FXS at -; the channel bank) -; em_rx: Receive audio/COR on an E&M interface (1-way) -; em_tx: Transmit audio/PTT on an E&M interface (1-way) -; em_txrx: Receive audio/COR AND Transmit audio/PTT on an E&M interface -; (2-way) -; em_rxtx: Same as em_txrx (for our dyslexic friends) -; sf_rx: Receive audio/COR on an SF interface (1-way) -; sf_tx: Transmit audio/PTT on an SF interface (1-way) -; sf_txrx: Receive audio/COR AND Transmit audio/PTT on an SF interface -; (2-way) -; sf_rxtx: Same as sf_txrx (for our dyslexic friends) -; ss7: Signalling System 7 -; -; signalling of a channel can not be changed on a reload. -; -;signalling=fxo_ls -; -; If you have an outbound signalling format that is different from format -; specified above (but compatible), you can specify outbound signalling format, -; (see below). The 'signalling' format specified will be the inbound signalling -; format. If you only specify 'signalling', then it will be the format for -; both inbound and outbound. -; -; outsignalling can only be one of: -; em, em_e1, em_w, sf, sf_w, sf_featd, sf_featdmf, sf_featb, featd, -; featdmf, featdmf_ta, e911, fgccama, fgccamamf -; -; outsignalling cannot be changed on a reload. -; -;signalling=featdmf -; -;outsignalling=featb -; -; For Feature Group D Tandem access, to set the default CIC and OZZ use these -; parameters (Will not be updated on reload): -; -;defaultozz=0000 -;defaultcic=303 -; -; A variety of timing parameters can be specified as well -; The default values for those are "-1", which is to use the -; compile-time defaults of the DAHDI kernel modules. The timing -; parameters, (with the standard default from DAHDI): -; -; prewink: Pre-wink time (default 50ms) -; preflash: Pre-flash time (default 50ms) -; wink: Wink time (default 150ms) -; flash: Flash time (default 750ms) -; start: Start time (default 1500ms) -; rxwink: Receiver wink time (default 300ms) -; rxflash: Receiver flashtime (default 1250ms) -; debounce: Debounce timing (default 600ms) -; -; None of them will update on a reload. -; -; How long generated tones (DTMF and MF) will be played on the channel -; (in milliseconds). -; -; This is a global, rather than a per-channel setting. It will not be -; updated on a reload. -; -;toneduration=100 -; -; Whether or not to do distinctive ring detection on FXO lines: -; -;usedistinctiveringdetection=yes -; -; enable dring detection after caller ID for those countries like Australia -; where the ring cadence is changed *after* the caller ID spill: -; -;distinctiveringaftercid=yes -; -; Whether or not to use caller ID: -; -usecallerid=yes -; -; 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): -; Support flash-hook call transfer (requires three way calling) -; Also enables call parking (overrides the 'canpark' parameter) -; -; For digital ports using ISDN PRI protocols: -; Support switch-side transfer (called 2BCT, RLT or other names) -; This setting must be enabled on both ports involved, and the -; 'facilityenable' setting must also be enabled to allow sending -; the transfer to the ISDN switch, since it sent in a FACILITY -; message. -; -transfer=yes -; -; Allow call parking -; ('canpark=no' is overridden by 'transfer=yes') -; -canpark=yes -; -; Support call forward variable -; -cancallforward=yes -; -; Whether or not to support Call Return (*69, if your dialplan doesn't -; catch this first) -; -callreturn=yes -; -; Stutter dialtone support: If a mailbox is specified without a voicemail -; context, then when voicemail is received in a mailbox in the default -; voicemail context in voicemail.conf, taking the phone off hook will cause a -; stutter dialtone instead of a normal one. -; -; If a mailbox is specified *with* a voicemail context, the same will result -; if voicemail received in mailbox in the specified voicemail context. -; -; for default voicemail context, the example below is fine: -; -;mailbox=1234 -; -; for any other voicemail context, the following will produce the stutter tone: -; -;mailbox=1234@context -; -; Enable echo cancellation -; Use either "yes", "no", or a power of two from 32 to 256 if you wish to -; actually set the number of taps of cancellation. -; -; Note that when setting the number of taps, the number 256 does not translate -; to 256 ms of echo cancellation. echocancel=256 means 256 / 8 = 32 ms. -; -; Note that if any of your DAHDI cards have hardware echo cancellers, -; then this setting only turns them on and off; numeric settings will -; be treated as "yes". There are no special settings required for -; hardware echo cancellers; when present and enabled in their kernel -; modules, they take precedence over the software echo canceller compiled -; into DAHDI automatically. -; -; -echocancel=yes -; -; As of Zaptel 1.4.8, some DAHDI echo cancellers (software and hardware) -; support adjustable parameters; these parameters can be supplied as -; additional options to the 'echocancel' setting. Note that Asterisk -; does not attempt to validate the parameters or their values, so if you -; supply an invalid parameter you will not know the specific reason it -; failed without checking the kernel message log for the error(s) -; put there by DAHDI. -; -;echocancel=128,param1=32,param2=0,param3=14 -; -; Generally, it is not necessary (and in fact undesirable) to echo cancel when -; the circuit path is entirely TDM. You may, however, change this behavior -; by enabling the echo canceller during pure TDM bridging below. -; -echocancelwhenbridged=yes -; -; In some cases, the echo canceller doesn't train quickly enough and there -; is echo at the beginning of the call. Enabling echo training will cause -; DAHDI to briefly mute the channel, send an impulse, and use the impulse -; response to pre-train the echo canceller so it can start out with a much -; closer idea of the actual echo. Value may be "yes", "no", or a number of -; milliseconds to delay before training (default = 400) -; -; WARNING: In some cases this option can make echo worse! If you are -; trying to debug an echo problem, it is worth checking to see if your echo -; is better with the option set to yes or no. Use whatever setting gives -; the best results. -; -; Note that these parameters do not apply to hardware echo cancellers. -; -;echotraining=yes -;echotraining=800 -; -; If you are having trouble with DTMF detection, you can relax the DTMF -; detection parameters. Relaxing them may make the DTMF detector more likely -; to have "talkoff" where DTMF is detected when it shouldn't be. -; -;relaxdtmf=yes -; -; You may also set the default receive and transmit gains (in dB) -; -; Gain Settings: increasing / decreasing the volume level on a channel. -; The values are in db (decibells). A positive number -; increases the volume level on a channel, and a -; negavive value decreases volume level. -; -; There are several independent gain settings: -; rxgain: gain for the rx (receive - into Asterisk) channel. Default: 0.0 -; txgain: gain for the tx (transmit - out of Asterisk Asterisk) channel. -; Default: 0.0 -; cid_rxgain: set the gain just for the caller ID sounds Asterisk -; emits. Default: 5.0 . - -;rxgain=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 - -; -; Specify whether the channel should be answered immediately or if the simple -; switch should provide dialtone, read digits, etc. -; Note: If immediate=yes the dialplan execution will always start at extension -; 's' priority 1 regardless of the dialed number! -; -;immediate=yes -; -; Specify whether flash-hook transfers to 'busy' channels should complete or -; return to the caller performing the transfer (default is yes). -; -;transfertobusy=no -; -; caller ID can be set to "asreceived" or a specific number if you want to -; override it. Note that "asreceived" only applies to trunk interfaces. -; fullname sets just the -; -; fullname: sets just the name part. -; cid_number: sets just the number part: -; -;callerid = 123456 -; -;callerid = My Name <2564286000> -; Which can also be written as: -;cid_number = 2564286000 -;fullname = My Name -; -;callerid = asreceived -; -; should we use the caller ID from incoming call on DAHDI transfer? -; -;useincomingcalleridondahditransfer = yes -; -; AMA flags affects the recording of Call Detail Records. If specified -; it may be 'default', 'omit', 'billing', or 'documentation'. -; -;amaflags=default -; -; Channels may be associated with an account code to ease -; billing -; -;accountcode=lss0101 -; -; ADSI (Analog Display Services Interface) can be enabled on a per-channel -; basis if you have (or may have) ADSI compatible CPE equipment -; -;adsi=yes -; -; SMDI (Simplified Message Desk Interface) can be enabled on a per-channel -; basis if you would like that channel to behave like an SMDI message desk. -; The SMDI port specified should have already been defined in smdi.conf. The -; default port is /dev/ttyS0. -; -;usesmdi=yes -;smdiport=/dev/ttyS0 -; -; On trunk interfaces (FXS) and E&M interfaces (E&M, Wink, Feature Group D -; etc, it can be useful to perform busy detection either in an effort to -; detect hangup or for detecting busies. This enables listening for -; the beep-beep busy pattern. -; -;busydetect=yes -; -; If busydetect is enabled, it is also possible to specify how many busy tones -; to wait for before hanging up. The default is 3, but it might be -; safer to set to 6 or even 8. Mind that the higher the number, the more -; time that will be needed to hangup a channel, but lowers the probability -; that you will get random hangups. -; -;busycount=6 -; -; If busydetect is enabled, it is also possible to specify the cadence of your -; busy signal. In many countries, it is 500msec on, 500msec off. Without -; busypattern specified, we'll accept any regular sound-silence pattern that -; repeats times as a busy signal. If you specify busypattern, -; then we'll further check the length of the sound (tone) and silence, which -; will further reduce the chance of a false positive. -; -;busypattern=500,500 -; -; NOTE: In make menuselect, you'll find further options to tweak the busy -; detector. If your country has a busy tone with the same length tone and -; silence (as many countries do), consider enabling the -; BUSYDETECT_COMPARE_TONE_AND_SILENCE option. -; -; To further detect which hangup tone your telco provider is sending, it is -; useful to use the ztmonitor utility to record the audio that main/dsp.c -; is receiving after the caller hangs up. -; -; Use a polarity reversal to mark when a outgoing call is answered by the -; remote party. -; -;answeronpolarityswitch=yes -; -; In some countries, a polarity reversal is used to signal the disconnect of a -; phone line. If the hanguponpolarityswitch option is selected, the call will -; be considered "hung up" on a polarity reversal. -; -;hanguponpolarityswitch=yes -; -; polarityonanswerdelay: minimal time period (ms) between the answer -; polarity switch and hangup polarity switch. -; (default: 600ms) -; -; On trunk interfaces (FXS) it can be useful to attempt to follow the progress -; of a call through RINGING, BUSY, and ANSWERING. If turned on, call -; progress attempts to determine answer, busy, and ringing on phone lines. -; This feature is HIGHLY EXPERIMENTAL and can easily detect false answers, -; so don't count on it being very accurate. -; -; Few zones are supported at the time of this writing, but may be selected -; with "progzone". -; -; progzone also affects the pattern used for buzydetect (unless -; busypattern is set explicitly). The possible values are: -; us (default) -; ca (alias for 'us') -; cr (Costa Rica) -; br (Brazil, alias for 'cr') -; uk -; -; This feature can also easily detect false hangups. The symptoms of this is -; being disconnected in the middle of a call for no reason. -; -;callprogress=yes -;progzone=uk -; -; Set the tonezone. Equivalent of the defaultzone settings in -; /etc/dahdi.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. -;tonezone = 0 ; 0 is US -; -; FXO (FXS signalled) devices must have a timeout to determine if there was a -; hangup before the line was answered. This value can be tweaked to shorten -; how long it takes before DAHDI considers a non-ringing line to have hungup. -; -; ringtimeout will not update on a reload. -; -;ringtimeout=8000 -; -; For FXO (FXS signalled) devices, whether to use pulse dial instead of DTMF -; Pulse digits from phones (FXS devices, FXO signalling) are always -; detected. -; -;pulsedial=yes -; -; For fax detection, uncomment one of the following lines. The default is *OFF* -; -;faxdetect=both -;faxdetect=incoming -;faxdetect=outgoing -;faxdetect=no -; -; This option specifies a preference for which music on hold class this channel -; should listen to when put on hold if the music class has not been set on the -; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer -; channel putting this one on hold did not suggest a music class. -; -; If this option is set to "passthrough", then the hold message will always be -; passed through as signalling instead of generating hold music locally. This -; setting is only valid when used on a channel that uses digital signalling. -; -;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 -; pri_cpe or pri_net for CPE or Network termination, and generally you will -; want to create a single "group" for all channels of the PRI. -; -; switchtype cannot be changed on a reload. -; -; switchtype = national -; signalling = pri_cpe -; group = 2 -; channel => 1-23 - -; - -; Used for distinctive ring support for x100p. -; You can see the dringX patterns is to set any one of the dringXcontext fields -; and they will be printed on the console when an inbound call comes in. -; -; dringXrange is used to change the acceptable ranges for "tone offsets". Defaults to 10. -; Note: a range of 0 is NOT what you might expect - it instead forces it to the default. -; A range of -1 will force it to always match. -; Anything lower than -1 would presumably cause it to never match. -; -;dring1=95,0,0 -;dring1context=internal1 -;dring1range=10 -;dring2=325,95,0 -;dring2context=internal2 -;dring2range=10 -; If no pattern is matched here is where we go. -;context=default -;channel => 1 - -; ---------------- Options for use with signalling=ss7 ----------------- -; None of them can be changed by a reload. -; -; Variant of SS7 signalling: -; Options are itu and ansi -;ss7type = itu - -; SS7 Called Nature of Address Indicator -; -; unknown: Unknown -; subscriber: Subscriber -; national: National -; international: International -; dynamic: Dynamically selects the appropriate dialplan -; -;ss7_called_nai=dynamic -; -; SS7 Calling Nature of Address Indicator -; -; unknown: Unknown -; subscriber: Subscriber -; national: National -; international: International -; dynamic: Dynamically selects the appropriate dialplan -; -;ss7_calling_nai=dynamic -; -; -; sample 1 for Germany -;ss7_internationalprefix = 00 -;ss7_nationalprefix = 0 -;ss7_subscriberprefix = -;ss7_unknownprefix = -; - -; 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 - -; 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 -; [channels]: -; -;echocancel = 64 -;group = 1 -;channel => 1-8 -; -; When starting a new section almost all of the configuration values are -; copied from their values at the end of the section [channels] in -; 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 -- cgit v1.2.3