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diff --git a/trunk/configs/zapata.conf.sample b/trunk/configs/zapata.conf.sample new file mode 100644 index 000000000..b2146b6b5 --- /dev/null +++ b/trunk/configs/zapata.conf.sample @@ -0,0 +1,909 @@ +; +; Zapata telephony interface +; +; Configuration file +; +; You need to restart Asterisk to re-configure the Zap channel +; CLI> reload chan_zap.so +; will reload the configuration file, +; but not all configuration options are +; re-configured during a reload (signalling, as well as +; PRI and SS7-related settings cannot be changed on a +; reload. +; +; This file documents many configuration variables. Normally unless you +; know what a variable means or that it should be changed, there's no +; reason to unrem lines. +; +; remmed-out examples below (those lines that begin with a ';' but no +; space afterwards) typically show a value that is not the defauult value, +; but would make sense under cetain circumstances. The default values +; are usually sane. Thus you should typically not touch them unless you +; know what they mean or you know you should change them. + + +[trunkgroups] +; +; Trunk groups are used for NFAS or GR-303 connections. +; +; Group: Defines a trunk group. +; trunkgroup => <trunkgroup>,<dchannel>[,<backup1>...] +; +; trunkgroup is the numerical trunk group to create +; dchannel is the zap 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 => <zapspan>,<trunkgroup>[,<logicalspan>] +; +; zapspan is the zap 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: Only RARELY used for PRI. +; PRI Local Dialplan: Only RARELY used for PRI (sets the calling number's +; numbering plan) +; pridialplan and prilocaldialplan cannot be changed on a 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=national +;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 Zaptel. +; 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 Zaptel kernel modules. The timing +; parameters, (with the standard default from Zaptel): +; +; 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/zapnotify.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 +; +; Support flash-hook call transfer (requires three way calling) +; Also enables call parking (overrides the 'canpark' parameter) +; +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 Zaptel 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 Zaptel automatically. +; +; +echocancel=yes +; +; As of Zaptel 1.4.8, some Zaptel 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 Zaptel. +; +;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 +; Zaptel 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 zap transfer? +; +;useincomingcalleridonzaptransfer = yes +; +; AMA flags affects the recording of Call Detail Records. If specified +; it may be 'default', 'omit', 'billing', or 'documentation'. +; +;amaflags=default +; +; Channels may be associated with an account code to ease +; billing +; +;accountcode=lss0101 +; +; ADSI (Analog Display Services Interface) can be enabled on a per-channel +; basis if you have (or may have) ADSI compatible CPE equipment +; +;adsi=yes +; +; SMDI (Simplified Message Desk Interface) can be enabled on a per-channel +; basis if you would like that channel to behave like an SMDI message desk. +; The SMDI port specified should have already been defined in smdi.conf. The +; default port is /dev/ttyS0. +; +;usesmdi=yes +;smdiport=/dev/ttyS0 +; +; On trunk interfaces (FXS) and E&M interfaces (E&M, Wink, Feature Group D +; etc, it can be useful to perform busy detection either in an effort to +; detect hangup or for detecting busies. This enables listening for +; the beep-beep busy pattern. +; +;busydetect=yes +; +; If busydetect is enabled, it is also possible to specify how many busy tones +; to wait for before hanging up. The default is 3, but it might be +; safer to set to 6 or even 8. Mind that the higher the number, the more +; time that will be needed to hangup a channel, but lowers the probability +; that you will get random hangups. +; +;busycount=6 +; +; If busydetect is enabled, it is also possible to specify the cadence of your +; busy signal. In many countries, it is 500msec on, 500msec off. Without +; busypattern specified, we'll accept any regular sound-silence pattern that +; repeats <busycount> times as a busy signal. If you specify busypattern, +; then we'll further check the length of the sound (tone) and silence, which +; will further reduce the chance of a false positive. +; +;busypattern=500,500 +; +; NOTE: In make menuselect, you'll find further options to tweak the busy +; detector. If your country has a busy tone with the same length tone and +; silence (as many countries do), consider enabling the +; BUSYDETECT_COMPARE_TONE_AND_SILENCE option. +; +; To further detect which hangup tone your telco provider is sending, it is +; useful to use the ztmonitor utility to record the audio that main/dsp.c +; is receiving after the caller hangs up. +; +; Use a polarity reversal to mark when a outgoing call is answered by the +; remote party. +; +;answeronpolarityswitch=yes +; +; In some countries, a polarity reversal is used to signal the disconnect of a +; phone line. If the hanguponpolarityswitch option is selected, the call will +; be considered "hung up" on a polarity reversal. +; +;hanguponpolarityswitch=yes +; +; polarityonanswerdelay: minimal time period (ms) between the answer +; polarity switch and hangup polarity switch. +; (default: 600ms) +; +; On trunk interfaces (FXS) it can be useful to attempt to follow the progress +; of a call through RINGING, BUSY, and ANSWERING. If turned on, call +; progress attempts to determine answer, busy, and ringing on phone lines. +; This feature is HIGHLY EXPERIMENTAL and can easily detect false answers, +; so don't count on it being very accurate. +; +; Few zones are supported at the time of this writing, but may be selected +; with "progzone". +; +; progzone also affects the pattern used for buzydetect (unless +; busypattern is set explicitly). The possible values are: +; us (default) +; ca (alias for 'us') +; cr (Costa Rica) +; br (Brazil, alias for 'cr') +; uk +; +; This feature can also easily detect false hangups. The symptoms of this is +; being disconnected in the middle of a call for no reason. +; +;callprogress=yes +;progzone=uk +; +; Set the tonezone. Equivalent of the defaultzone settings in +; /etc/zaptel.conf . This sets the tone zone by number. +; Note that you'd still need to load tonezones (loadzone in zaptel.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 Zap 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_zap 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 Zapata (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 + ; ZAP 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 ZAP channel can't accept jitter, + ; thus an enabled jitterbuffer on the receive ZAP 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 ZAP + ; 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 ---------------------------------------- |