Improved comments and organization for zapata.conf (#10904)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86572 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Matthew Fredrickson
parent
d30564fe86
commit
c5bb538818
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@ -7,8 +7,19 @@
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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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; re-configured during a reload (signalling, as well as
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; PRI and SS7-related settings cannot be changed on a
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; reload.
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;
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; This file documents many configuration variables. Normally unless you
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; know what a variable means or that it should be changed, there's no
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; reason to unrem lines.
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;
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; remmed-out examples below (those lines that begin with a ';' but no
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; space afterwards) typically show a value that is not the defauult value,
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; but would make sense under cetain circumstances. The default values
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; are usually sane. Thus you should typically not touch them unless you
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; know what they mean or you know you should change them.
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[trunkgroups]
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@ -45,9 +56,9 @@
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;
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;language=en
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;
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; Default context
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; Context for calls. Defaults to 'default'
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;
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context=default
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;context=incoming
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;
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; Switchtype: Only used for PRI.
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;
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@ -55,18 +66,23 @@ context=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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; euroisdn: EuroISDN (common in Europe)
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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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;switchtype=euroisdn
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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 cannot be changed on a reload.
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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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; PRI Local Dialplan: Only RARELY used for PRI (sets the calling number's
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; numbering plan)
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; pridialplan and prilocaldialplan cannot be changed on a reload.
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;
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; unknown: Unknown
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; private: Private ISDN
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@ -74,25 +90,17 @@ switchtype=national
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; national: National ISDN
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; international: International ISDN
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; dynamic: Dynamically selects the appropriate dialplan
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; redundant: Same as dynamic, except that the underlying number is not changed (not common)
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; redundant: Same as dynamic, except that the underlying number is not
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; changed (not common)
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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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; dynamic: Dynamically selects the appropriate dialplan
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; redundant: Same as dynamic, except that the underlying number is not changed (not common)
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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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; PRI caller ID 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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; None of the prefix settings can be changed on reload.
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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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@ -115,6 +123,7 @@ switchtype=national
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;resetinterval = 3600
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;
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; Overlap dialing mode (sending overlap digits)
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; Cannot be changed on a reload.
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;
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;overlapdial=yes
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;
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@ -124,13 +133,18 @@ switchtype=national
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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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; inband: Signal Busy/Congestion using in-band tones (default)
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;
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; priindication = outofband
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; priindication cannot be changed on a reload.
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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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; priexclusive cannot be changed on a reload.
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;
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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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@ -141,21 +155,26 @@ switchtype=national
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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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; T309: Maintain active calls on Layer 2 disconnection (default -1,
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Asterisk clears calls)
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; EuroISDN: 6000 to 12000 ms, according to (N200 + 1) x 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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;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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; Cannot be changed on a reload.
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;
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;facilityenable = yes
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;
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; pritimer cannot be changed on a reload.
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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_e1: E & M E1
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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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@ -180,6 +199,7 @@ switchtype=national
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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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; ss7: Signalling System 7
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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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@ -202,6 +222,8 @@ switchtype=national
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; sf_rxtx: Same as sf_txrx (for our dyslexic friends)
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; ss7: Signalling System 7
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;
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; signalling of a channel can not be changed on a reload.
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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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@ -210,16 +232,27 @@ signalling=fxo_ls
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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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; signalling=featdmf
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; outsignalling=featb
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; outsignalling can only be one of:
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; em, em_e1, em_w, sf, sf_w, sf_featd, sf_featdmf, sf_featb, featd,
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; featdmf, featdmf_ta, e911, fgccama, fgccamamf
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;
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; outsignalling cannot be changed on a reload.
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;
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;signalling=featdmf
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;
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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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; parameters (Will not be updated on reload):
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;
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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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; The default values for those are "-1", which is to use the
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; compile-time defaults of the Zaptel kernel modules. The timing
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; parameters, (with the standard default from Zaptel):
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;
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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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@ -229,43 +262,55 @@ signalling=fxo_ls
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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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; None of them will update on a reload.
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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 milliseconds)
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; (in milliseconds).
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;
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; This is a global, rather than a per-channel setting. It will not be
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; updated on a reload.
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;
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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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; 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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; enable dring detection after caller ID for those countries like Australia
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; where the ring cadence is changed *after* the caller ID spill:
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;
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;distinctiveringaftercid=yes
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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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; Hide the name part and leave just the number part of the caller ID
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; string. Only applies to PRI channels.
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;hidecalleridname=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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; bell = bell202 as used in US (default)
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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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; smdi = Use SMDI for caller ID. Requires SMDI to be enabled (usesmdi).
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;
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;cidsignalling=bell
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;cidsignalling=v23
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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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; ring = a ring signals the start (default)
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; polarity = polarity reversal signals the start
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; polarity_IN = polarity reversal signals the start, for India,
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; for dtmf dialtone detection; using DTMF.
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; (see doc/India-CID.txt)
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; for dtmf dialtone detection; using DTMF.
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; (see doc/India-CID.txt)
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;
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;cidstart=ring
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;cidstart=polarity
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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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; (If your dialplan doesn't catch it)
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;
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hidecallerid=no
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;hidecallerid=yes
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;
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; Whether or not to enable call waiting on internal extensions
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; With this set to 'yes', busy extensions will hear the call-waiting
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@ -282,18 +327,18 @@ callwaiting=yes
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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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; See README.callingpres. FIXME: file no longer exists.
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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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; which means the caller ID needs to be set later on, and not just after
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; the first ring, as per the default (1).
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;
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;sendcalleridafter=1
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;sendcalleridafter = 2
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;
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;
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; Support Caller*ID on Call Waiting
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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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@ -315,7 +360,8 @@ canpark=yes
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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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; Whether or not to support Call Return (*69, if your dialplan doesn't
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; catch this first)
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;
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callreturn=yes
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;
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@ -382,39 +428,72 @@ echocancelwhenbridged=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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; Gain Settings: increasing / decreasing the volume level on a channel.
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; The values are in db (decibells). A positive number
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; increases the volume level on a channel, and a
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; negavive value decreases volume level.
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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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; There are several independent gain settings:
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; rxgain: gain for the rx (receive - into Asterisk) channel. Default: 0.0
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; txgain: gain for the tx (transmit - out of Asterisk Asterisk) channel.
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; Default: 0.0
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; cid_rxgain: set the gain just for the caller ID sounds Asterisk
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; emits. Default: 5.0 .
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;rxgain=2.0
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;txgain=3.0
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;
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; Logical groups can be assigned to allow outgoing roll-over. Groups range
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; from 0 to 63, and multiple groups can be specified. By default the
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; channel is not a member of any group.
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;
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; Note that an explicit empty value for 'group' is invalid, and will not
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; override a previous non-empty one. The same applies to callgroup and
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; pickupgroup as well.
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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 dialling *8#. For simple offices, just
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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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;setvar=CHANNEL=42 ; Channel variable to be set for all calls from this channel
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; Channel variable to be set for all calls from this channel
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;setvar=CHANNEL=42
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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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;immediate=yes
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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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; caller ID 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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; fullname sets just the
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;
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;callerid=2564286000
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; fullname: sets just the name part.
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; cid_number: sets just the number part:
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;
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;callerid = 123456
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;
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;callerid = My Name <2564286000>
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; Which can also be written as:
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;cid_number = 2564286000
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;fullname = My Name
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;
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;callerid = asreceived
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;
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; should we use the caller ID from incoming call on zap transfer?
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;
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;useincomingcalleridonzaptransfer = yes
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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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@ -447,12 +526,12 @@ immediate=no
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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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; to wait for before hanging up. The default is 3, but it might be
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; safer to 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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;busycount=6
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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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@ -479,6 +558,10 @@ immediate=no
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;
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;hanguponpolarityswitch=yes
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;
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; polarityonanswerdelay: minimal time period (ms) between the answer
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; polarity switch and hangup polarity switch.
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; (default: 600ms)
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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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|
@ -486,21 +569,39 @@ immediate=no
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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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; with "progzone".
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;
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; progzone also affects the pattern used for buzydetect (unless
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; busypattern is set explicitly). The possible values are:
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; us (default)
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; ca (alias for 'us')
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; cr (Costa Rica)
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; br (Brazil, alias for 'cr')
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; uk
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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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;progzone=uk
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;
|
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; Set the tonezone. Equivalent of the defaultzone settings in
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; /etc/zaptel.conf . This sets the tone zone by number.
|
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; Note that you'd still need to load tonezones (loadzone in zaptel.conf).
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; The default is -1: not to set anything.
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;tonezone = 0 ; 0 is US
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;
|
||||
; 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
|
||||
;
|
||||
|
|
@ -520,13 +621,10 @@ immediate=no
|
|||
; 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.
|
||||
; when this channel places the peer on hold.
|
||||
;
|
||||
;mohsuggest=default
|
||||
;
|
||||
|
|
@ -540,12 +638,15 @@ immediate=no
|
|||
; 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)
|
||||
; Configure jitter buffers in Zapata (each one is 20ms, default is 4)
|
||||
; This is set globally, rather than per-channel.
|
||||
;
|
||||
;jitterbuffers=4
|
||||
;
|
||||
|
|
@ -573,10 +674,13 @@ immediate=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
|
||||
; 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:
|
||||
|
|
@ -600,9 +704,7 @@ immediate=no
|
|||
;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
|
||||
|
|
@ -648,6 +750,8 @@ immediate=no
|
|||
; 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
|
||||
|
|
@ -675,6 +779,8 @@ immediate=no
|
|||
;channel => 1
|
||||
|
||||
; ---------------- Options for use with signalling=ss7 -----------------
|
||||
; None of them can be changed by a reload.
|
||||
;
|
||||
; Variant of SS7 signaling:
|
||||
; Options are itu and ansi
|
||||
;ss7type = itu
|
||||
|
|
@ -736,4 +842,7 @@ immediate=no
|
|||
|
||||
; 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 ----------------------------------------
|
||||
|
|
|
|||
Loading…
Reference in New Issue