f15be28fb0
This introduces two new options for zapata.conf: mwimonitor and mwimonitornotify. The mwimonitor option enables MWI monitoring. When the MWI state on a line changes, then the script specified by mwimonitornotify will be executed for custom handling of the state change, similar to the externnotify option of voicemail.conf. Also, when the MWI state on an FXO line changes, an internal Asterisk event is generated to indicate the new state of the associated mailbox. That may, any module that cares about MWI information will get notified and can handle it just as if app_voicemail had sent this notification. (BE-253, original patch from markster, with some minor modifications by me to add comments, documentation, and internal event support) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90949 65c4cc65-6c06-0410-ace0-fbb531ad65f3
887 lines
30 KiB
Text
887 lines
30 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 (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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;
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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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; trunkgroup => <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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; Context for calls. Defaults to 'default'
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;
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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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; 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 (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=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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; 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
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; changed (not common)
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;
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;pridialplan=national
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;prilocaldialplan=national
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;
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; pridialplan may be also set at dialtime, by prefixing the dialled number with
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; one of the following letters:
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; U - Unknown
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; P - Private
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; L - Local
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; N - National
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; I - International
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; D - Dynamic
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; R - Redundant
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;
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; Additionally, you may also set the following NPI bits (also by prefixing the
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; dialled string with one of the following letters):
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; u - Unknown
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; e - E.163/E.164
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; x - X.121
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; f - F.69
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; n - National
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; p - Private
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; r - Reserved
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;
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; You may also set the prilocaldialplan in the same way, but by prefixing the
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; Caller*ID Number, rather than the dialled number. Please note that telcos
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; which require this kind of additional manipulation of the TON/NPI are *rare*.
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; Most telco PRIs will work fine simply by setting pridialplan to unknown or
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; dynamic.
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;
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;
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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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;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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; B channels; defaults to 'never'.
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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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; Cannot be changed on a reload.
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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 (default)
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;
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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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;
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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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; 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,
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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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;
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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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; 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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; 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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; 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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; 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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; 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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; 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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; 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 (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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; 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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; 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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; 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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;
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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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;
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;usedistinctiveringdetection=yes
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;
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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 (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 caller ID. Requires SMDI to be enabled (usesmdi).
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;
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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 (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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;
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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=yes
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;
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; The following option enables receiving MWI on FXO lines. The default
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; value is no. When this is enabled, and MWI notification indicates on or off,
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; the script specified by the mwimonitornotify option is executed.
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;
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;mwimonitor=no
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;
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; This option is used in conjunction with mwimonitor. This will get executed
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; when incoming MWI state changes. The script is passed 2 arguments. The
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; first is the corresponding mailbox, and the second is 1 or 0, indicating if
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; there are messages waiting or not.
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;
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;mwimonitornotify=/usr/local/bin/zapnotify.sh
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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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; tone, and can use hook-flash to switch between callers. The Dial()
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; app will not return the "BUSY" result for extensions.
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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. 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 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 = 2
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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, 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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; 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 received 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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; Note that when setting the number of taps, the number 256 does not translate
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; to 256 ms of echo cancellation. echocancel=256 means 256 / 8 = 32 ms.
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;
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; Note that if any of your Zaptel cards have hardware echo cancellers,
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; then this setting only turns them on and off; numeric settings will
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; be treated as "yes". There are no special settings required for
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; hardware echo cancellers; when present and enabled in their kernel
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; modules, they take precedence over the software echo canceller compiled
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; into Zaptel automatically.
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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, change 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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; WARNING: In some cases this option can make echo worse! If you are
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; trying to debug an echo problem, it is worth checking to see if your echo
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; is better with the option set to yes or no. Use whatever setting gives
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; the best results.
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;
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; Note that these parameters do not apply to hardware echo cancellers.
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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
|
|
;
|
|
; 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.
|
|
;
|
|
;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 the Asterisk Makefile 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 defining the
|
|
; -DBUSYDETECT_COMPARE_TONE_AND_SILENCE option.
|
|
;
|
|
; 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 signaling:
|
|
; 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 signaling 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 signaling 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 ----------------------------------------
|