; ; 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 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=SIERRA_Q2687RD ; ; Some switches (AT&T especially) require network specific facility IE ; supported values are currently 'none', 'sdn', 'megacom', 'tollfreemegacom', 'accunet' ; ; nsf cannot be changed on a reload. ; ;nsf=none ; ; PRI Dialplan: The ISDN-level Type Of Number (TON) or numbering plan, used for ; the dialed number. For most installations, leaving this as 'unknown' (the ; default) works in the most cases. In some very unusual circumstances, you ; may need to set this to 'dynamic' or 'redundant'. Note that if you set one ; of the others, you will be unable to dial another class of numbers. For ; example, if you set 'national', you will be unable to dial local or ; international numbers. ; ; PRI Local Dialplan: Only RARELY used for PRI (sets the calling number's ; numbering plan). In North America, the typical use is sending the 10 digit ; callerID number and setting the prilocaldialplan to 'national' (the default). ; Only VERY rarely will you need to change this. ; ; Neither pridialplan nor prilocaldialplan can be changed on reload. ; ; unknown: Unknown ; private: Private ISDN ; local: Local ISDN ; national: National ISDN ; international: International ISDN ; dynamic: Dynamically selects the appropriate dialplan ; redundant: Same as dynamic, except that the underlying number is not ; changed (not common) ; ;pridialplan=unknown ;prilocaldialplan=national ; ; pridialplan may be also set at dialtime, by prefixing the dialled number with ; one of the following letters: ; U - Unknown ; I - International ; N - National ; L - Local (Net Specific) ; S - Subscriber ; V - Abbreviated ; R - Reserved (should probably never be used but is included for completeness) ; ; Additionally, you may also set the following NPI bits (also by prefixing the ; dialled string with one of the following letters): ; u - Unknown ; e - E.163/E.164 (ISDN/telephony) ; x - X.121 (Data) ; f - F.69 (Telex) ; n - National ; p - Private ; r - Reserved (should probably never be used but is included for completeness) ; ; You may also set the prilocaldialplan in the same way, but by prefixing the ; Caller*ID Number, rather than the dialled number. Please note that telcos ; which require this kind of additional manipulation of the TON/NPI are *rare*. ; Most telco PRIs will work fine simply by setting pridialplan to unknown or ; dynamic. ; ; ; PRI caller ID prefixes based on the given TON/NPI (dialplan) ; This is especially needed for EuroISDN E1-PRIs ; ; None of the prefix settings can be changed on reload. ; ; sample 1 for Germany ;internationalprefix = 00 ;nationalprefix = 0 ;localprefix = 0711 ;privateprefix = 07115678 ;unknownprefix = ; ; sample 2 for Germany ;internationalprefix = + ;nationalprefix = +49 ;localprefix = +49711 ;privateprefix = +497115678 ;unknownprefix = ; ; PRI resetinterval: sets the time in seconds between restart of unused ; B channels; defaults to 'never'. ; ;resetinterval = 3600 ; ; Overlap dialing mode (sending overlap digits) ; Cannot be changed on a reload. ; ; incoming: incoming direction only ; outgoing: outgoing direction only ; no: neither direction ; yes or both: both directions ; ;overlapdial=yes ; ; Allow inband audio (progress) when a call is RELEASEd by the far end of a PRI ; ;inbanddisconnect=yes ; ; PRI Out of band indications. ; Enable this to report Busy and Congestion on a PRI using out-of-band ; notification. Inband indication, as used by Asterisk doesn't seem to work ; with all telcos. ; ; outofband: Signal Busy/Congestion out of band with RELEASE/DISCONNECT ; inband: Signal Busy/Congestion using in-band tones (default) ; ; priindication cannot be changed on a reload. ; ;priindication = outofband ; ; If you need to override the existing channels selection routine and force all ; PRI channels to be marked as exclusively selected, set this to yes. ; ; priexclusive cannot be changed on a reload. ; ;priexclusive = yes ; ; ISDN Timers ; All of the ISDN timers and counters that are used are configurable. Specify ; the timer name, and its value (in ms for timers). ; K: Layer 2 max number of outstanding unacknowledged I frames (default 7) ; N200: Layer 2 max number of retransmissions of a frame (default 3) ; T200: Layer 2 max time before retransmission of a frame (default 1000 ms) ; T203: Layer 2 max time without frames being exchanged (default 10000 ms) ; T305: Wait for DISCONNECT acknowledge (default 30000 ms) ; T308: Wait for RELEASE acknowledge (default 4000 ms) ; T309: Maintain active calls on Layer 2 disconnection (default -1, ; Asterisk clears calls) ; EuroISDN: 6000 to 12000 ms, according to (N200 + 1) x T200 + 2s ; May vary in other ISDN standards (Q.931 1993 : 90000 ms) ; T313: Wait for CONNECT acknowledge, CPE side only (default 3000 ms) ; ;pritimer => t200,1000 ;pritimer => t313,4000 ; ; To enable transmission of facility-based ISDN supplementary services (such ; as caller name from CPE over facility), enable this option. ; Cannot be changed on a reload. ; ;facilityenable = yes ; ; pritimer cannot be changed on a reload. ; ; Signalling method. The default is "auto". Valid values: ; auto: Use the current value from DAHDI. ; em: E & M ; em_e1: E & M E1 ; em_w: E & M Wink ; featd: Feature Group D (The fake, Adtran style, DTMF) ; featdmf: Feature Group D (The real thing, MF (domestic, US)) ; featdmf_ta: Feature Group D (The real thing, MF (domestic, US)) through ; a Tandem Access point ; featb: Feature Group B (MF (domestic, US)) ; fgccama Feature Group C-CAMA (DP DNIS, MF ANI) ; fgccamamf Feature Group C-CAMA MF (MF DNIS, MF ANI) ; fxs_ls: FXS (Loop Start) ; fxs_gs: FXS (Ground Start) ; fxs_ks: FXS (Kewl Start) ; fxo_ls: FXO (Loop Start) ; fxo_gs: FXO (Ground Start) ; fxo_ks: FXO (Kewl Start) ; pri_cpe: PRI signalling, CPE side ; pri_net: PRI signalling, Network side ; gr303fxoks_net: GR-303 Signalling, FXO Loopstart, Network side ; gr303fxsks_cpe: GR-303 Signalling, FXS Loopstart, CPE side ; sf: SF (Inband Tone) Signalling ; sf_w: SF Wink ; sf_featd: SF Feature Group D (The fake, Adtran style, DTMF) ; sf_featdmf: SF Feature Group D (The real thing, MF (domestic, US)) ; sf_featb: SF Feature Group B (MF (domestic, US)) ; e911: E911 (MF) style signalling ; ss7: Signalling System 7 ; ; The following are used for Radio interfaces: ; fxs_rx: Receive audio/COR on an FXS kewlstart interface (FXO at the ; channel bank) ; fxs_tx: Transmit audio/PTT on an FXS loopstart interface (FXO at the ; channel bank) ; fxo_rx: Receive audio/COR on an FXO loopstart interface (FXS at the ; channel bank) ; fxo_tx: Transmit audio/PTT on an FXO groundstart interface (FXS at ; the channel bank) ; em_rx: Receive audio/COR on an E&M interface (1-way) ; em_tx: Transmit audio/PTT on an E&M interface (1-way) ; em_txrx: Receive audio/COR AND Transmit audio/PTT on an E&M interface ; (2-way) ; em_rxtx: Same as em_txrx (for our dyslexic friends) ; sf_rx: Receive audio/COR on an SF interface (1-way) ; sf_tx: Transmit audio/PTT on an SF interface (1-way) ; sf_txrx: Receive audio/COR AND Transmit audio/PTT on an SF interface ; (2-way) ; sf_rxtx: Same as sf_txrx (for our dyslexic friends) ; ss7: Signalling System 7 ; ; signalling of a channel can not be changed on a reload. ; ;signalling=fxo_ls ; ; If you have an outbound signalling format that is different from format ; specified above (but compatible), you can specify outbound signalling format, ; (see below). The 'signalling' format specified will be the inbound signalling ; format. If you only specify 'signalling', then it will be the format for ; both inbound and outbound. ; ; outsignalling can only be one of: ; em, em_e1, em_w, sf, sf_w, sf_featd, sf_featdmf, sf_featb, featd, ; featdmf, featdmf_ta, e911, fgccama, fgccamamf ; ; outsignalling cannot be changed on a reload. ; ;signalling=featdmf ; ;outsignalling=featb ; ; For Feature Group D Tandem access, to set the default CIC and OZZ use these ; parameters (Will not be updated on reload): ; ;defaultozz=0000 ;defaultcic=303 ; ; A variety of timing parameters can be specified as well ; The default values for those are "-1", which is to use the ; compile-time defaults of the DAHDI kernel modules. The timing ; parameters, (with the standard default from DAHDI): ; ; prewink: Pre-wink time (default 50ms) ; preflash: Pre-flash time (default 50ms) ; wink: Wink time (default 150ms) ; flash: Flash time (default 750ms) ; start: Start time (default 1500ms) ; rxwink: Receiver wink time (default 300ms) ; rxflash: Receiver flashtime (default 1250ms) ; debounce: Debounce timing (default 600ms) ; ; None of them will update on a reload. ; ; How long generated tones (DTMF and MF) will be played on the channel ; (in milliseconds). ; ; This is a global, rather than a per-channel setting. It will not be ; updated on a reload. ; ;toneduration=100 ; ; Whether or not to do distinctive ring detection on FXO lines: ; ;usedistinctiveringdetection=yes ; ; enable dring detection after caller ID for those countries like Australia ; where the ring cadence is changed *after* the caller ID spill: ; ;distinctiveringaftercid=yes ; ; Whether or not to use caller ID: ; usecallerid=yes ; ; 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. ; The mwimonitor can take the following values ; no - No mwimonitoring occurs. (default) ; yes - The same as specifying fsk ; fsk - the FXO line is monitored for MWI FSK spills ; fsk,rpas - the FXO line is monitored for MWI FSK spills preceded ; by a ring pulse alert signal. ; neon - The fxo line is monitored for the presence of NEON pulses ; indicating MWI. ; When detected, an internal Asterisk MWI event is generated so that any other ; part of Asterisk that cares about MWI state changes is notified, just as if ; the state change came from app_voicemail. ; For FSK MWI Spills, the energy level that must be seen before starting the ; MWI detection process can be set with 'mwilevel'. ; ;mwimonitor=no ;mwilevel=512 ; ; This option is used in conjunction with mwimonitor. This will get executed ; when incoming MWI state changes. The script is passed 2 arguments. The ; first is the corresponding mailbox, and the second is 1 or 0, indicating if ; there are messages waiting or not. ; ;mwimonitornotify=/usr/local/bin/dahdinotify.sh ; ; Whether or not to enable call waiting on internal extensions ; With this set to 'yes', busy extensions will hear the call-waiting ; tone, and can use hook-flash to switch between callers. The Dial() ; app will not return the "BUSY" result for extensions. ; callwaiting=yes ; ; Whether or not restrict outgoing caller ID (will be sent as ANI only, not ; available for the user) ; Mostly use with FXS ports ; Does nothing. Use hidecallerid instead. ; ;restrictcid=no ; ; Whether or not to use the caller ID presentation from the Asterisk channel ; for outgoing calls. ; See dialplan function CALLERID(pres) for more information. ; Only applies to PRI and SS7 channels. ; usecallingpres=yes ; ; Some countries (UK) have ring tones with different ring tones (ring-ring), ; which means the caller ID needs to be set later on, and not just after ; the first ring, as per the default (1). ; ;sendcalleridafter = 2 ; ; ; Support caller ID on Call Waiting ; callwaitingcallerid=yes ; ; Support three-way calling ; threewaycalling=yes ; ; For FXS ports (either direct analog or over T1/E1): ; Support flash-hook call transfer (requires three way calling) ; Also enables call parking (overrides the 'canpark' parameter) ; ; For digital ports using ISDN PRI protocols: ; Support switch-side transfer (called 2BCT, RLT or other names) ; This setting must be enabled on both ports involved, and the ; 'facilityenable' setting must also be enabled to allow sending ; the transfer to the ISDN switch, since it sent in a FACILITY ; message. ; transfer=yes ; ; Allow call parking ; ('canpark=no' is overridden by 'transfer=yes') ; canpark=yes ; ; Support call forward variable ; cancallforward=yes ; ; Whether or not to support Call Return (*69, if your dialplan doesn't ; catch this first) ; callreturn=yes ; ; Stutter dialtone support: If a mailbox is specified without a voicemail ; context, then when voicemail is received in a mailbox in the default ; voicemail context in voicemail.conf, taking the phone off hook will cause a ; stutter dialtone instead of a normal one. ; ; If a mailbox is specified *with* a voicemail context, the same will result ; if voicemail received in mailbox in the specified voicemail context. ; ; for default voicemail context, the example below is fine: ; ;mailbox=1234 ; ; for any other voicemail context, the following will produce the stutter tone: ; ;mailbox=1234@context ; ; Enable echo cancellation ; Use either "yes", "no", or a power of two from 32 to 256 if you wish to ; actually set the number of taps of cancellation. ; ; Note that when setting the number of taps, the number 256 does not translate ; to 256 ms of echo cancellation. echocancel=256 means 256 / 8 = 32 ms. ; ; Note that if any of your DAHDI cards have hardware echo cancellers, ; then this setting only turns them on and off; numeric settings will ; be treated as "yes". There are no special settings required for ; hardware echo cancellers; when present and enabled in their kernel ; modules, they take precedence over the software echo canceller compiled ; into DAHDI automatically. ; ; echocancel=yes ;channel=1-8 ; ; 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 ;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! ; ;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/system.conf. This sets the tone zone by number. ; Note that you'd still need to load tonezones (loadzone in ; /etc/dahdi/system.conf). ; The default is -1: not to set anything. ;tonezone = 0 ; 0 is US ; ; FXO (FXS signalled) devices must have a timeout to determine if there was a ; hangup before the line was answered. This value can be tweaked to shorten ; how long it takes before DAHDI considers a non-ringing line to have hungup. ; ; ringtimeout will not update on a reload. ; ;ringtimeout=8000 ; ; For FXO (FXS signalled) devices, whether to use pulse dial instead of DTMF ; Pulse digits from phones (FXS devices, FXO signalling) are always ; detected. ; ;pulsedial=yes ; ; For fax detection, uncomment one of the following lines. The default is *OFF* ; ;faxdetect=both ;faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ; ; This option specifies a preference for which music on hold class this channel ; should listen to when put on hold if the music class has not been set on the ; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer ; channel putting this one on hold did not suggest a music class. ; ; If this option is set to "passthrough", then the hold message will always be ; passed through as signalling instead of generating hold music locally. This ; setting is only valid when used on a channel that uses digital signalling. ; ; This option may be set globally or on a per-channel basis. ; ;mohinterpret=default ; ; This option specifies which music on hold class to suggest to the peer channel ; when this channel places the peer on hold. This option may be set globally, ; or on a per-channel basis. ; ;mohsuggest=default ; ; PRI channels can have an idle extension and a minunused number. So long as ; at least "minunused" channels are idle, chan_dahdi will try to call "idledial" ; on them, and then dump them into the PBX in the "idleext" extension (which ; is of the form exten@context). When channels are needed the "idle" calls ; are disconnected (so long as there are at least "minidle" calls still ; running, of course) to make more channels available. The primary use of ; this is to create a dynamic service, where idle channels are bundled through ; multilink PPP, thus more efficiently utilizing combined voice/data services ; than conventional fixed mappings/muxings. ; ; Those settings cannot be changed on reload. ; ;idledial=6999 ;idleext=6999@dialout ;minunused=2 ;minidle=1 ; ; Configure jitter buffers in DAHDI (each one is 20ms, default is 4) ; This is set globally, rather than per-channel. ; ;jitterbuffers=4 ; ;------------------------------ JITTER BUFFER CONFIGURATION -------------------------- ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a ; DAHDI channel. Defaults to "no". An enabled jitterbuffer will ; be used only if the sending side can create and the receiving ; side can not accept jitter. The DAHDI channel can't accept jitter, ; thus an enabled jitterbuffer on the receive DAHDI side will always ; be used if the sending side can create jitter. ; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds. ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is ; resynchronized. Useful to improve the quality of the voice, with ; big jumps in/broken timestamps, usually sent from exotic devices ; and programs. Defaults to 1000. ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a DAHDI ; channel. Two implementations are currently available - "fixed" ; (with size always equals to jbmax-size) and "adaptive" (with ; variable size, actually the new jb of IAX2). Defaults to fixed. ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no". ;----------------------------------------------------------------------------------- ; ; You can define your own custom ring cadences here. You can define up to 8 ; pairs. If the silence is negative, it indicates where the caller ID spill is ; to be placed. Also, if you define any custom cadences, the default cadences ; will be turned off. ; ; This setting is global, rather than per-channel. It will not update on ; a reload. ; ; Syntax is: cadence=ring,silence[,ring,silence[...]] ; ; These are the default cadences: ; ;cadence=125,125,2000,-4000 ;cadence=250,250,500,1000,250,250,500,-4000 ;cadence=125,125,125,125,125,-4000 ;cadence=1000,500,2500,-5000 ; ; Each channel consists of the channel number or range. It inherits the ; parameters that were specified above its declaration. ; ; For GR-303, CRV's are created like channels except they must start with the ; trunk group followed by a colon, e.g.: ; ; crv => 1:1 ; crv => 2:1-2,5-8 ; ; ;callerid="Green Phone"<(256) 428-6121> ;channel => 1 ;callerid="Black Phone"<(256) 428-6122> ;channel => 2 ;callerid="CallerID Phone" <(630) 372-1564> ;channel => 3 ;callerid="Pac Tel Phone" <(256) 428-6124> ;channel => 4 ;callerid="Uniden Dead" <(256) 428-6125> ;channel => 5 ;callerid="Cortelco 2500" <(256) 428-6126> ;channel => 6 ;callerid="Main TA 750" <(256) 428-6127> ;channel => 44 ; ; For example, maybe we have some other channels which start out in a ; different context and use E & M signalling instead. ; ;context=remote ;signaling=em ;channel => 15 ;channel => 16 ;signalling=em_w ; ; All those in group 0 I'll use for outgoing calls ; ; Strip most significant digit (9) before sending ; ;stripmsd=1 ;callerid=asreceived ;group=0 ;signalling=fxs_ls ;channel => 45 ;signalling=fxo_ls ;group=1 ;callerid="Joe Schmoe" <(256) 428-6131> ;channel => 25 ;callerid="Megan May" <(256) 428-6132> ;channel => 26 ;callerid="Suzy Queue" <(256) 428-6233> ;channel => 27 ;callerid="Larry Moe" <(256) 428-6234> ;channel => 28 ; ; Sample PRI (CPE) config: Specify the switchtype, the signalling as either ; pri_cpe or pri_net for CPE or Network termination, and generally you will ; want to create a single "group" for all channels of the PRI. ; ; switchtype cannot be changed on a reload. ; ; switchtype = national ; signalling = pri_cpe ; group = 2 ; channel => 1-23 ; ; Used for distinctive ring support for x100p. ; You can see the dringX patterns is to set any one of the dringXcontext fields ; and they will be printed on the console when an inbound call comes in. ; ; dringXrange is used to change the acceptable ranges for "tone offsets". Defaults to 10. ; Note: a range of 0 is NOT what you might expect - it instead forces it to the default. ; A range of -1 will force it to always match. ; Anything lower than -1 would presumably cause it to never match. ; ;dring1=95,0,0 ;dring1context=internal1 ;dring1range=10 ;dring2=325,95,0 ;dring2context=internal2 ;dring2range=10 ; If no pattern is matched here is where we go. ;context=default ;channel => 1 ; ---------------- Options for use with signalling=ss7 ----------------- ; None of them can be changed by a reload. ; ; Variant of SS7 signalling: ; Options are itu and ansi ;ss7type = itu ; SS7 Called Nature of Address Indicator ; ; unknown: Unknown ; subscriber: Subscriber ; national: National ; international: International ; dynamic: Dynamically selects the appropriate dialplan ; ;ss7_called_nai=dynamic ; ; SS7 Calling Nature of Address Indicator ; ; unknown: Unknown ; subscriber: Subscriber ; national: National ; international: International ; dynamic: Dynamically selects the appropriate dialplan ; ;ss7_calling_nai=dynamic ; ; ; sample 1 for Germany ;ss7_internationalprefix = 00 ;ss7_nationalprefix = 0 ;ss7_subscriberprefix = ;ss7_unknownprefix = ; ; This option is used to disable automatic sending of ACM when the call is started ; in the dialplan. If you do use this option, you will need to use the Proceeding() ; application in the dialplan to send ACM. ;ss7_explictacm=yes ; All settings apply to linkset 1 ;linkset = 1 ; Point code of the linkset. For ITU, this is the decimal number ; format of the point code. For ANSI, this can either be in decimal ; number format or in the xxx-xxx-xxx format ;pointcode = 1 ; Point code of node adjacent to this signalling link (Possibly the STP between you and ; your destination). Point code format follows the same rules as above. ;adjpointcode = 2 ; Default point code that you would like to assign to outgoing messages (in case of ; routing through STPs, or using A links). Point code format follows the same rules ; as above. ;defaultdpc = 3 ; Begin CIC (Circuit indication codes) count with this number ;cicbeginswith = 1 ; What the MTP3 network indicator bits should be set to. Choices are ; national, national_spare, international, international_spare ;networkindicator=international ; First signalling channel ;sigchan = 48 ; Additional signalling channel for this linkset (So you can have a linkset ; with two signalling links in it). It seems like a silly way to do it, but ; for linksets with multiple signalling links, you add an additional sigchan ; line for every additional signalling link on the linkset. ;sigchan = 96 ; Channels to associate with CICs on this linkset ;channel = 25-47 ; ; For more information on setting up SS7, see the README file in libss7 or ; the doc/ss7.txt file in the Asterisk source tree. ; ----------------- SS7 Options ---------------------------------------- ; Configuration Sections ; ~~~~~~~~~~~~~~~~~~~~~~ ; You can also configure channels in a separate chan_dahdi.conf section. In ; this case the keyword 'channel' is not used. Instead the keyword ; 'dahdichan' is used (as in users.conf) - configuration is only processed ; in a section where the keyword dahdichan is used. It will only be ; processed in the end of the section. Thus the following section: ; ;[phones] ;echocancel = 64 ;dahdichan = 1-8 ;group = 1 ; ; Is somewhat equivalent to the following snippet in the section ; [channels]: ; ;echocancel = 64 ;group = 1 ;channel => 1-8 ; ; When starting a new section almost all of the configuration values are ; copied from their values at the end of the section [channels] in ; chan_dahdi.conf and [general] in users.conf - one section's configuration ; does not affect another one's. ; ; Instead of letting common configuration values "slide through" you can ; use configuration templates to easily keep the common part in one ; place and override where needed. ; ;[phones](!) ;echocancel = yes ;group = 0,4 ;callgroup = 3 ;pickupgroup = 3 ;threewaycalling = yes ;transfer = yes ;context = phones ;faxdetect = incoming ; ;[phone-1](phones) ;dahdichan = 1 ;callerid = My Name <501> ;mailbox = 501@mailboxes ; ; ;[fax](phones) ;dahdichan = 2 ;faxdetect = no ;context = fax ; ;[phone-3](phones) ;dahdichan = 3 ;pickupgroup = 3,4 #include allogsm-channels.conf