git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
640 lines
22 KiB
Text
640 lines
22 KiB
Text
;
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; Zapata telephony interface
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;
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; Configuration file
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;
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; You need to restart Asterisk to re-configure the Zap channel
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; CLI> reload chan_zap.so
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; will reload the configuration file,
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; but not all configuration options are
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; re-configured during a reload.
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[trunkgroups]
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;
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; Trunk groups are used for NFAS or GR-303 connections.
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;
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; Group: Defines a trunk group.
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; group => <trunkgroup>,<dchannel>[,<backup1>...]
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;
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; trunkgroup is the numerical trunk group to create
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; dchannel is the zap channel which will have the
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; d-channel for the trunk.
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; backup1 is an optional list of backup d-channels.
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;
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;trunkgroup => 1,24,48
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;trunkgroup => 1,24
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;
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; Spanmap: Associates a span with a trunk group
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; spanmap => <zapspan>,<trunkgroup>[,<logicalspan>]
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;
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; zapspan is the zap span number to associate
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; trunkgroup is the trunkgroup (specified above) for the mapping
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; logicalspan is the logical span number within the trunk group to use.
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; if unspecified, no logical span number is used.
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;
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;spanmap => 1,1,1
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;spanmap => 2,1,2
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;spanmap => 3,1,3
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;spanmap => 4,1,4
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[channels]
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;
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; Default language
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;
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;language=en
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;
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; Default context
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;
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context=default
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;
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; Switchtype: Only used for PRI.
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;
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; national: National ISDN 2 (default)
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; dms100: Nortel DMS100
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; 4ess: AT&T 4ESS
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; 5ess: Lucent 5ESS
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; euroisdn: EuroISDN
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; ni1: Old National ISDN 1
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; qsig: Q.SIG
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;
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switchtype=national
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;
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; Some switches (AT&T especially) require network specific facility IE
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; supported values are currently 'none', 'sdn', 'megacom', 'tollfreemegacom', 'accunet'
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;
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;nsf=none
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;
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; PRI Dialplan: Only RARELY used for PRI.
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;
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; unknown: Unknown
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; private: Private ISDN
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; local: Local ISDN
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; national: National ISDN
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; international: International ISDN
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;
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;pridialplan=national
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;
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; PRI Local Dialplan: Only RARELY used for PRI (sets the calling number's numbering plan)
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;
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; unknown: Unknown
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; private: Private ISDN
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; local: Local ISDN
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; national: National ISDN
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; international: International ISDN
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;
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;prilocaldialplan=national
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;
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; PRI callerid prefixes based on the given TON/NPI (dialplan)
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; This is especially needed for euroisdn E1-PRIs
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;
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; sample 1 for Germany
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;internationalprefix = 00
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;nationalprefix = 0
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;localprefix = 0711
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;privateprefix = 07115678
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;unknownprefix =
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;
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; sample 2 for Germany
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;internationalprefix = +
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;nationalprefix = +49
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;localprefix = +49711
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;privateprefix = +497115678
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;unknownprefix =
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;
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; PRI resetinterval: sets the time in seconds between restart of unused
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; channels, defaults to 3600; minimum 60 seconds. Some PBXs don't like
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; channel restarts. so set the interval to a very long interval e.g. 100000000
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; or 'never' to disable *entirely*.
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;
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;resetinterval = 3600
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;
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; Overlap dialing mode (sending overlap digits)
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;
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;overlapdial=yes
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;
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; PRI Out of band indications.
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; Enable this to report Busy and Congestion on a PRI using out-of-band
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; notification. Inband indication, as used by Asterisk doesn't seem to work
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; with all telcos.
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;
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; outofband: Signal Busy/Congestion out of band with RELEASE/DISCONNECT
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; inband: Signal Busy/Congestion using in-band tones
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;
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; priindication = outofband
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;
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; If you need to override the existing channels selection routine and force all
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; PRI channels to be marked as exclusively selected, set this to yes.
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; priexclusive = yes
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;
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; ISDN Timers
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; All of the ISDN timers and counters that are used are configurable. Specify
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; the timer name, and its value (in ms for timers).
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; K: Layer 2 max number of outstanding unacknowledged I frames (default 7)
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; N200: Layer 2 max number of retransmissions of a frame (default 3)
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; T200: Layer 2 max time before retransmission of a frame (default 1000 ms)
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; T203: Layer 2 max time without frames being exchanged (default 10000 ms)
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; T305: Wait for DISCONNECT acknowledge (default 30000 ms)
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; T308: Wait for RELEASE acknowledge (default 4000 ms)
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; T309: Maintain active calls on Layer 2 disconnection (default -1, Asterisk clears calls)
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; EuroISDN: 6000 to 12000 ms, according to (N200 + 1) × T200 + 2s
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; May vary in other ISDN standards (Q.931 1993 : 90000 ms)
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; T313: Wait for CONNECT acknowledge, CPE side only (default 3000 ms)
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;
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; pritimer => t200,1000
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; pritimer => t313,4000
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;
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; To enable transmission of facility-based ISDN supplementary services (such
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; as caller name from CPE over facility), enable this option.
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; facilityenable = yes
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;
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;
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; Signalling method (default is fxs). Valid values:
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; em: E & M
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; em_w: E & M Wink
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; featd: Feature Group D (The fake, Adtran style, DTMF)
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; featdmf: Feature Group D (The real thing, MF (domestic, US))
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; featdmf_ta: Feature Group D (The real thing, MF (domestic, US)) through
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; a Tandem Access point
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; featb: Feature Group B (MF (domestic, US))
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; fgccama Feature Group C-CAMA (DP DNIS, MF ANI)
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; fgccamamf Feature Group C-CAMA MF (MF DNIS, MF ANI)
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; fxs_ls: FXS (Loop Start)
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; fxs_gs: FXS (Ground Start)
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; fxs_ks: FXS (Kewl Start)
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; fxo_ls: FXO (Loop Start)
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; fxo_gs: FXO (Ground Start)
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; fxo_ks: FXO (Kewl Start)
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; pri_cpe: PRI signalling, CPE side
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; pri_net: PRI signalling, Network side
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; gr303fxoks_net: GR-303 Signalling, FXO Loopstart, Network side
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; gr303fxsks_cpe: GR-303 Signalling, FXS Loopstart, CPE side
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; sf: SF (Inband Tone) Signalling
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; sf_w: SF Wink
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; sf_featd: SF Feature Group D (The fake, Adtran style, DTMF)
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; sf_featdmf: SF Feature Group D (The real thing, MF (domestic, US))
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; sf_featb: SF Feature Group B (MF (domestic, US))
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; e911: E911 (MF) style signalling
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;
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; The following are used for Radio interfaces:
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; fxs_rx: Receive audio/COR on an FXS kewlstart interface (FXO at the
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; channel bank)
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; fxs_tx: Transmit audio/PTT on an FXS loopstart interface (FXO at the
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; channel bank)
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; fxo_rx: Receive audio/COR on an FXO loopstart interface (FXS at the
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; channel bank)
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; fxo_tx: Transmit audio/PTT on an FXO groundstart interface (FXS at
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; the channel bank)
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; em_rx: Receive audio/COR on an E&M interface (1-way)
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; em_tx: Transmit audio/PTT on an E&M interface (1-way)
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; em_txrx: Receive audio/COR AND Transmit audio/PTT on an E&M interface
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; (2-way)
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; em_rxtx: Same as em_txrx (for our dyslexic friends)
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; sf_rx: Receive audio/COR on an SF interface (1-way)
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; sf_tx: Transmit audio/PTT on an SF interface (1-way)
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; sf_txrx: Receive audio/COR AND Transmit audio/PTT on an SF interface
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; (2-way)
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; sf_rxtx: Same as sf_txrx (for our dyslexic friends)
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;
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signalling=fxo_ls
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;
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; If you have an outbound signalling format that is different from format
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; specified above (but compatible), you can specify outbound signalling format,
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; (see below). The 'signalling' format specified will be the inbound signalling
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; format. If you only specify 'signalling', then it will be the format for
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; both inbound and outbound.
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;
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; signaling=featdmf
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; outsignalling=featb
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;
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; For Feature Group D Tandem access, to set the default CIC and OZZ use these
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; parameters:
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;defaultozz=0000
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;defaultcic=303
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;
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; A variety of timing parameters can be specified as well
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; Including:
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; prewink: Pre-wink time (default 50ms)
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; preflash: Pre-flash time (default 50ms)
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; wink: Wink time (default 150ms)
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; flash: Flash time (default 750ms)
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; start: Start time (default 1500ms)
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; rxwink: Receiver wink time (default 300ms)
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; rxflash: Receiver flashtime (default 1250ms)
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; debounce: Debounce timing (default 600ms)
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;
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rxwink=300 ; Atlas seems to use long (250ms) winks
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;
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; How long generated tones (DTMF and MF) will be played on the channel
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; (in miliseconds)
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;toneduration=100
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;
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; Whether or not to do distinctive ring detection on FXO lines
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;
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;usedistinctiveringdetection=yes
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;distinctiveringaftercid=yes ; enable dring detection after callerid for those countries like Australia
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; where the ring cadence is changed *after* the callerid spill.
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;
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; Whether or not to use caller ID
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;
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usecallerid=yes
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;
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; Type of caller ID signalling in use
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; bell = bell202 as used in US
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; v23 = v23 as used in the UK
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; v23_jp = v23 as used in Japan
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; dtmf = DTMF as used in Denmark, Sweden and Netherlands
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; smdi = Use SMDI for callerid. Requires SMDI to be enabled (usesmdi).
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;
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;cidsignalling=bell
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;
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; What signals the start of caller ID
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; ring = a ring signals the start
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; polarity = polarity reversal signals the start
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;
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;cidstart=ring
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;
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; Whether or not to hide outgoing caller ID (Override with *67 or *82)
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;
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hidecallerid=no
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;
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; Whether or not to enable call waiting on FXO lines
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;
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callwaiting=yes
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;
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; Whether or not restrict outgoing caller ID (will be sent as ANI only, not
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; available for the user)
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; Mostly use with FXS ports
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;
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;restrictcid=no
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;
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; Whether or not use the caller ID presentation for the outgoing call that the
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; calling switch is sending.
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; See README.callingpres
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;
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usecallingpres=yes
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;
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; Some countries (UK) have ring tones with different ring tones (ring-ring),
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; which means the callerid needs to be set later on, and not just after
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; the first ring, as per the default.
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;
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;sendcalleridafter=1
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;
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;
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; Support Caller*ID on Call Waiting
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;
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callwaitingcallerid=yes
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;
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; Support three-way calling
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;
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threewaycalling=yes
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;
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; Support flash-hook call transfer (requires three way calling)
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; Also enables call parking (overrides the 'canpark' parameter)
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;
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transfer=yes
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;
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; Allow call parking
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; ('canpark=no' is overridden by 'transfer=yes')
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;
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canpark=yes
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;
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; Support call forward variable
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;
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cancallforward=yes
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;
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; Whether or not to support Call Return (*69)
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;
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callreturn=yes
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;
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; Stutter dialtone support: If a mailbox is specified without a voicemail
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; context, then when voicemail is received in a mailbox in the default
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; voicemail context in voicemail.conf, taking the phone off hook will cause a
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; stutter dialtone instead of a normal one.
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;
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; If a mailbox is specified *with* a voicemail context, the same will result
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; if voicemail recieved in mailbox in the specified voicemail context.
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;
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; for default voicemail context, the example below is fine:
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;
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;mailbox=1234
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;
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; for any other voicemail context, the following will produce the stutter tone:
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;
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;mailbox=1234@context
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;
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; Enable echo cancellation
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; Use either "yes", "no", or a power of two from 32 to 256 if you wish to
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; actually set the number of taps of cancellation.
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;
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echocancel=yes
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;
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; Generally, it is not necessary (and in fact undesirable) to echo cancel when
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; the circuit path is entirely TDM. You may, however, reverse this behavior
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; by enabling the echo cancel during pure TDM bridging below.
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;
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echocancelwhenbridged=yes
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;
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; In some cases, the echo canceller doesn't train quickly enough and there
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; is echo at the beginning of the call. Enabling echo training will cause
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; asterisk to briefly mute the channel, send an impulse, and use the impulse
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; response to pre-train the echo canceller so it can start out with a much
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; closer idea of the actual echo. Value may be "yes", "no", or a number of
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; milliseconds to delay before training (default = 400)
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;
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;echotraining=yes
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;echotraining=800
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;
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; If you are having trouble with DTMF detection, you can relax the DTMF
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; detection parameters. Relaxing them may make the DTMF detector more likely
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; to have "talkoff" where DTMF is detected when it shouldn't be.
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;
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;relaxdtmf=yes
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;
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; You may also set the default receive and transmit gains (in dB)
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;
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rxgain=0.0
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txgain=0.0
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;
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; Logical groups can be assigned to allow outgoing rollover. Groups range
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; from 0 to 63, and multiple groups can be specified.
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;
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group=1
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;
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; Ring groups (a.k.a. call groups) and pickup groups. If a phone is ringing
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; and it is a member of a group which is one of your pickup groups, then
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; you can answer it by picking up and dialing *8#. For simple offices, just
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; make these both the same. Groups range from 0 to 63.
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;
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callgroup=1
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pickupgroup=1
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;
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; Specify whether the channel should be answered immediately or if the simple
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; switch should provide dialtone, read digits, etc.
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;
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immediate=no
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;
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; Specify whether flash-hook transfers to 'busy' channels should complete or
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; return to the caller performing the transfer (default is yes).
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;
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;transfertobusy=no
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;
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; CallerID can be set to "asreceived" or a specific number if you want to
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; override it. Note that "asreceived" only applies to trunk interfaces.
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;
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;callerid=2564286000
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;
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; AMA flags affects the recording of Call Detail Records. If specified
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; it may be 'default', 'omit', 'billing', or 'documentation'.
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;
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;amaflags=default
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;
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; Channels may be associated with an account code to ease
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; billing
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;
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;accountcode=lss0101
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;
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; ADSI (Analog Display Services Interface) can be enabled on a per-channel
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; basis if you have (or may have) ADSI compatible CPE equipment
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;
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;adsi=yes
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;
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; SMDI (Simplified Message Desk Interface) can be enabled on a per-channel
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; basis if you would like that channel to behave like an SMDI message desk.
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; The SMDI port specfied should have already been defined in smdi.conf. The
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; default port is /dev/ttyS0.
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;
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;usesmdi=yes
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;smdiport=/dev/ttyS0
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;
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; On trunk interfaces (FXS) and E&M interfaces (E&M, Wink, Feature Group D
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; etc, it can be useful to perform busy detection either in an effort to
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; detect hangup or for detecting busies. This enables listening for
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; the beep-beep busy pattern.
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;
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;busydetect=yes
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;
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; If busydetect is enabled, it is also possible to specify how many busy tones
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; to wait for before hanging up. The default is 4, but better results can be
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; achieved if set to 6 or even 8. Mind that the higher the number, the more
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; time that will be needed to hangup a channel, but lowers the probability
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; that you will get random hangups.
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;
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;busycount=4
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;
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; If busydetect is enabled, it is also possible to specify the cadence of your
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; busy signal. In many countries, it is 500msec on, 500msec off. Without
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; busypattern specified, we'll accept any regular sound-silence pattern that
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; repeats <busycount> times as a busy signal. If you specify busypattern,
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; then we'll further check the length of the sound (tone) and silence, which
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; will further reduce the chance of a false positive.
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;
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;busypattern=500,500
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;
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; NOTE: In the Asterisk Makefile you'll find further options to tweak the busy
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; detector. If your country has a busy tone with the same length tone and
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; silence (as many countries do), consider defining the
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; -DBUSYDETECT_COMPARE_TONE_AND_SILENCE option.
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;
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; Use a polarity reversal to mark when a outgoing call is answered by the
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; remote party.
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;
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;answeronpolarityswitch=yes
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;
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; In some countries, a polarity reversal is used to signal the disconnect of a
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; phone line. If the hanguponpolarityswitch option is selected, the call will
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; be considered "hung up" on a polarity reversal.
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;
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;hanguponpolarityswitch=yes
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;
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; On trunk interfaces (FXS) it can be useful to attempt to follow the progress
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; of a call through RINGING, BUSY, and ANSWERING. If turned on, call
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; progress attempts to determine answer, busy, and ringing on phone lines.
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; This feature is HIGHLY EXPERIMENTAL and can easily detect false answers,
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; so don't count on it being very accurate.
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;
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; Few zones are supported at the time of this writing, but may be selected
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; with "progzone"
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;
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; This feature can also easily detect false hangups. The symptoms of this is
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; being disconnected in the middle of a call for no reason.
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;
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;callprogress=yes
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;progzone=us
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;
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; FXO (FXS signalled) devices must have a timeout to determine if there was a
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; hangup before the line was answered. This value can be tweaked to shorten
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; how long it takes before Zap considers a non-ringing line to have hungup.
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;
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;ringtimeout=8000
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;
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; For FXO (FXS signalled) devices, whether to use pulse dial instead of DTMF
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;
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;pulsedial=yes
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;
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; For fax detection, uncomment one of the following lines. The default is *OFF*
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;
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;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 specified globally, or on a per-user or per-peer 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. It may be specified globally or on
|
||
; a per-user or per-peer basis.
|
||
;
|
||
;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.
|
||
;
|
||
;idledial=6999
|
||
;idleext=6999@dialout
|
||
;minunused=2
|
||
;minidle=1
|
||
;
|
||
; Configure jitter buffers in zapata (each one is 20ms, default is 4)
|
||
;
|
||
;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, usualy 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 currenlty 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 callerid spill is
|
||
; to be placed. Also, if you define any custom cadences, the default cadences
|
||
; will be turned off.
|
||
;
|
||
; 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" <(256) 428-6123>
|
||
;callerid="CallerID Phone" <(630) 372-1564>
|
||
;callerid="CallerID Phone" <(256) 704-4666>
|
||
;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 = national
|
||
; signalling = pri_cpe
|
||
; group = 2
|
||
; channel => 1-23
|
||
|
||
;
|
||
|
||
; Used for distintive 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.
|
||
;
|
||
;dring1=95,0,0
|
||
;dring1context=internal1
|
||
;dring2=325,95,0
|
||
;dring2context=internal2
|
||
; If no pattern is matched here is where we go.
|
||
;context=default
|
||
;channel => 1
|
||
|