1737 lines
68 KiB
Plaintext
1737 lines
68 KiB
Plaintext
;
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; DAHDI Telephony Configuration file
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;
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; You need to restart Asterisk to re-configure the DAHDI channel
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; CLI> module reload chan_dahdi.so
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; will reload the configuration file, but not all configuration options
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; are re-configured during a reload (signalling, as well as PRI and
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; SS7-related settings cannot be changed on a reload).
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;
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; This file documents many configuration variables. Normally unless you know
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; what a variable means or that it should be changed, there's no reason to
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; un-comment those lines.
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;
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; Examples below that are commented out (those lines that begin with a ';' but
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; no space afterwards) typically show a value that is not the default value,
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; but would make sense under certain circumstances. The default values are
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; usually sane. Thus you should typically not touch them unless you know what
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; 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 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 DAHDI 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 => <dahdispan>,<trunkgroup>[,<logicalspan>]
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;
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; dahdispan is the DAHDI 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 incoming calls. Defaults to 'default'
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;
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context=public
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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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; MSNs for ISDN spans. Asterisk will listen for the listed numbers on
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; incoming calls and ignore any calls not listed.
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; Here you can give a comma separated list of numbers or dialplan extension
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; patterns. An empty list disables MSN matching to allow any incoming call.
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; Only set on PTMP CPE side of ISDN span if needed.
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; The default is an empty list.
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;msn=
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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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;service_message_support=yes
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; Enable service message support for channel. Must be set after switchtype.
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;
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; Dialing options for ISDN (i.e., Dial(DAHDI/g1/exten/options)):
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; R Reverse Charge Indication
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; Indicate to the called party that the call will be reverse charged.
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; K(n) Keypad digits n
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; Send out the specified digits as keypad digits.
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;
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; PRI Dialplan: The ISDN-level Type Of Number (TON) or numbering plan, used for
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; the dialed number. Leaving this as 'unknown' (the default) works for most
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; cases. In some very unusual circumstances, you may need to set this to
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; 'dynamic' or 'redundant'.
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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 using the
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; prefix settings.
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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 cannot be changed on reload.
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;pridialplan=unknown
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;
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; PRI Local Dialplan: Only RARELY used for PRI (sets the calling number's
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; numbering plan). In North America, the typical use is sending the 10 digit
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; callerID number and setting the prilocaldialplan to 'national' (the default).
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; Only VERY rarely will you need to change this.
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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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; from_channel: Use the CALLERID(ton) value from the channel.
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; dynamic: Dynamically selects the appropriate dialplan using the
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; prefix settings.
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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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; prilocaldialplan cannot be changed on reload.
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;prilocaldialplan=national
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;
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; PRI Connected Line Dialplan: Sets the connected party 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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; from_channel: Use the CONNECTEDLINE(ton) value from the channel.
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; dynamic: Dynamically selects the appropriate dialplan using the
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; prefix settings.
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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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; pricpndialplan cannot be changed on reload.
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;pricpndialplan=from_channel
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;
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; pridialplan may be also set at dialtime, by prefixing the dialed number with
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; one of the following letters:
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; U - Unknown
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; I - International
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; N - National
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; L - Local (Net Specific)
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; S - Subscriber
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; V - Abbreviated
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; R - Reserved (should probably never be used but is included for completeness)
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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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; dialed string with one of the following letters):
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; u - Unknown
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; e - E.163/E.164 (ISDN/telephony)
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; x - X.121 (Data)
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; f - F.69 (Telex)
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; n - National
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; p - Private
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; r - Reserved (should probably never be used but is included for completeness)
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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 dialed number.
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; Please note that telcos which require this kind of additional manipulation
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; of the TON/NPI are *rare*. Most telco PRIs will work fine simply by
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; setting pridialplan to unknown or 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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; Enable per ISDN span to force a RESTART on a channel that returns a cause
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; code of PRI_CAUSE_REQUESTED_CHAN_UNAVAIL(44). If this option is enabled
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; and the reason the peer rejected the call with cause 44 was that the
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; channel is stuck in an unavailable state on the peer, then this might
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; help release the channel. It is worth noting that the next outgoing call
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; Asterisk makes will likely try the same channel again.
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;
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; NOTE: Sending a RESTART in response to a cause 44 is not required
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; (nor prohibited) by the standards and is likely a primitive chan_dahdi
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; response to call collisions (glare) and buggy peers. However, there
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; are telco switches out there that ignore the RESTART and continue to
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; send calls to the channel in the restarting state.
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; Default no.
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;
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;force_restart_unavailable_chans=yes
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;
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; Assume inband audio may be present when a SETUP ACK message is received.
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; Q.931 Section 5.1.3 says that in scenarios with overlap dialing, when a
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; dialtone is sent from the network side, progress indicator 8 "Inband info
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; now available" MAY be sent to the CPE if no digits were received with
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; the SETUP. It is thus implied that the ie is mandatory if digits came
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; with the SETUP and dialtone is needed.
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; This option should be enabled, when the network sends dialtone and you
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; want to hear it, but the network doesn't send the progress indicator when
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; needed.
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;
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; NOTE: For Q.SIG setups this option should be enabled when outgoing overlap
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; dialing is also enabled because Q.SIG does not send the progress indicator
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; with the SETUP ACK.
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; Default no.
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;
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;inband_on_setup_ack=yes
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;
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; Assume inband audio may be present when a PROCEEDING message is received.
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; Q.931 Section 5.1.2 says the network cannot assume that the CPE side has
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; attached to the B channel at this time without explicitly sending the
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; progress indicator ie informing the CPE side to attach to the B channel
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; for audio. However, some non-compliant ISDN switches send a PROCEEDING
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; without the progress indicator ie indicating inband audio is available and
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; assume that the CPE device has connected the media path for listening to
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; ringback and other messages.
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; Default no.
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;
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;inband_on_proceeding=yes
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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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; incoming: incoming direction only
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; outgoing: outgoing direction only
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; no: neither direction
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; yes or both: both directions
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;
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;overlapdial=yes
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; Send/receive ISDN display IE options. The display options are a comma separated
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; list of the following options:
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;
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; block: Do not pass display text data.
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; Q.SIG: Default for send/receive.
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; ETSI CPE: Default for send.
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; name_initial: Use display text in SETUP/CONNECT messages as the party name.
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; Default for all other modes.
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; name_update: Use display text in other messages (NOTIFY/FACILITY) for COLP name
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; update.
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; name: Combined name_initial and name_update options.
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; text: Pass any unused display text data as an arbitrary display message
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; during a call. Sent text goes out in an INFORMATION message.
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;
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; * Default is an empty string for legacy behavior.
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; * The name options are not recommended for Q.SIG since Q.SIG already
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; supports names.
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; * The send block is the only recommended setting for CPE mode since Q.931 uses
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; the display IE only in the network to user direction.
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;
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; display_send and display_receive cannot be changed on reload.
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;
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;display_send=
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;display_receive=
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; Allow sending an ISDN Malicious Caller ID (MCID) request on this span.
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; Default disabled
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;
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;mcid_send=yes
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; Send ISDN date/time IE in CONNECT message option. Only valid on NT spans.
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;
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; no: Do not send date/time IE in CONNECT message.
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; date: Send date only.
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; date_hh Send date and hour.
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; date_hhmm Send date, hour, and minute.
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; date_hhmmss Send date, hour, minute, and second.
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;
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; Default is an empty string which lets libpri pick the default
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; date/time IE send policy.
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;
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;datetime_send=
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; Send ISDN conected line information.
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;
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; block: Do not send any connected line information.
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; connect: Send connected line information on initial connect.
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; update: Same as connect but also send any updates during a call.
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; Updates happen if the call is transferred. (Default)
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;
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;colp_send=update
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; Allow inband audio (progress) when a call is DISCONNECTed by the far end of a PRI
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;
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;inbanddisconnect=yes
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;
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; Allow a held call to be transferred to the active call on disconnect.
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; This is useful on BRI PTMP NT lines where an ISDN phone can simulate the
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; transfer feature of an analog phone.
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; The default is no.
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;hold_disconnect_transfer=yes
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; BRI PTMP layer 1 presence.
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; You should normally not need to set this option.
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; You may need to set this option if your telco brings layer 1 down when
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; the line is idle.
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; required: Layer 1 presence required for outgoing calls. (default)
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; ignore: Ignore alarms from DAHDI about this span.
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; (Layer 1 and 2 will be brought back up for an outgoing call.)
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; NOTE: You will not be able to detect physical line problems
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; until an outgoing call is attempted and fails.
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;
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;layer1_presence=ignore
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; BRI PTMP layer 2 persistence.
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; You should normally not need to set this option.
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; You may need to set this option if your telco brings layer 1 down when
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; the line is idle.
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; <blank>: Use libpri default.
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; keep_up: Bring layer 2 back up if peer takes it down.
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; leave_down: Leave layer 2 down if peer takes it down. (Libpri default)
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; (Layer 2 will be brought back up for an outgoing call.)
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;
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;layer2_persistence=leave_down
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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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;
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; If you need to use the logical channel mapping with your Q.SIG PRI instead
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; of the physical mapping you must use the qsigchannelmapping option.
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;
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; logical: Use the logical channel mapping
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; physical: Use physical channel mapping (default)
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;
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;qsigchannelmapping=logical
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;
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; If you wish to ignore remote hold indications (and use MOH that is supplied over
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; the B channel) enable this option.
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;
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;discardremoteholdretrieval=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 6000 ms)
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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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; T-RESPONSE: Maximum time to wait for a typical APDU response. (default 4000 ms)
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; This is an implementation timer when the standard does not specify one.
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; T-ACTIVATE: Request supervision timeout. (default 10000 ms)
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; T-RETENTION: Maximum time to wait for user A to activate call-completion. (default 30000 ms)
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; Used by ETSI PTP, ETSI PTMP, and Q.SIG as the cc_offer_timer.
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; T-CCBS1: T-STATUS timer equivalent for CC user A status. (default 4000 ms)
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; T-CCBS2: Maximum time the CCBS service will be active (default 45 min in ms)
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; T-CCBS3: Maximum time to wait for user A to respond to user B availability. (default 20000 ms)
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; T-CCBS5: Network B CCBS supervision timeout. (default 60 min in ms)
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; T-CCBS6: Network A CCBS supervision timeout. (default 60 min in ms)
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; T-CCNR2: Maximum time the CCNR service will be active (default 180 min in ms)
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; T-CCNR5: Network B CCNR supervision timeout. (default 195 min in ms)
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; T-CCNR6: Network A CCNR supervision timeout. (default 195 min in ms)
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; CC-T1: Q.SIG CC request supervision timeout. (default 30000 ms)
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; CCBS-T2: Q.SIG CCBS supervision timeout. (default 60 min in ms)
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; CCNR-T2: Q.SIG CCNR supervision timeout. (default 195 min in ms)
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; CC-T3: Q.SIG CC Maximum time to wait for user A to respond to user B availability. (default 30000 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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; CC PTMP recall mode:
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; specific - Only the CC original party A can participate in the CC callback
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; global - Other compatible endpoints on the PTMP line can be party A in the CC callback
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;
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; cc_ptmp_recall_mode cannot be changed on a reload.
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;
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;cc_ptmp_recall_mode = specific
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;
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; CC Q.SIG Party A (requester) retain signaling link option
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; retain Require that the signaling link be retained.
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; release Request that the signaling link be released.
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; do_not_care The responder is free to choose if the signaling link will be retained.
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;
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;cc_qsig_signaling_link_req = retain
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;
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; CC Q.SIG Party B (responder) retain signaling link option
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; retain Prefer that the signaling link be retained.
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; release Prefer that the signaling link be released.
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;
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;cc_qsig_signaling_link_rsp = retain
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;
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; See ccss.conf.sample for more options. The timers described by ccss.conf.sample
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; are not used by ISDN for the native protocol since they are defined by the
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; standards and set by pritimer above.
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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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; This option enables Advice of Charge pass-through between the ISDN PRI and
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; Asterisk. This option can be set to any combination of 's', 'd', and 'e' which
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; represent the different variants of Advice of Charge, AOC-S, AOC-D, and AOC-E.
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; Advice of Charge pass-through is currently only supported for ETSI. Since most
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; AOC messages are sent on facility messages, the 'facilityenable' option must
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; also be enabled to fully support AOC pass-through.
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;
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;aoc_enable=s,d,e
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;
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; When this option is enabled, a hangup initiated by the ISDN PRI side of the
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; asterisk channel will result in the channel delaying its hangup in an
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; attempt to receive the final AOC-E message from its bridge. The delay
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; period is configured as one half the T305 timer length. If the channel
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; is not bridged the hangup will occur immediatly without delay.
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;
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;aoce_delayhangup=yes
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; pritimer cannot be changed on a reload.
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;
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; Signalling method. The default is "auto". Valid values:
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; auto: Use the current value from DAHDI.
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; em: E & M
|
|
; em_e1: E & M E1
|
|
; em_w: E & M Wink
|
|
; featd: Feature Group D (The fake, Adtran style, DTMF)
|
|
; featdmf: Feature Group D (The real thing, MF (domestic, US))
|
|
; featdmf_ta: Feature Group D (The real thing, MF (domestic, US)) through
|
|
; a Tandem Access point
|
|
; featb: Feature Group B (MF (domestic, US))
|
|
; fgccama: Feature Group C-CAMA (DP DNIS, MF ANI)
|
|
; fgccamamf: Feature Group C-CAMA MF (MF DNIS, MF ANI)
|
|
; fxs_ls: FXS (Loop Start)
|
|
; fxs_gs: FXS (Ground Start)
|
|
; fxs_ks: FXS (Kewl Start)
|
|
; fxo_ls: FXO (Loop Start)
|
|
; fxo_gs: FXO (Ground Start)
|
|
; fxo_ks: FXO (Kewl Start)
|
|
; pri_cpe: PRI signalling, CPE side
|
|
; pri_net: PRI signalling, Network side
|
|
; bri_cpe: BRI PTP signalling, CPE side
|
|
; bri_net: BRI PTP signalling, Network side
|
|
; bri_cpe_ptmp: BRI PTMP signalling, CPE side
|
|
; bri_net_ptmp: BRI PTMP signalling, Network side
|
|
; sf: SF (Inband Tone) Signalling
|
|
; sf_w: SF Wink
|
|
; sf_featd: SF Feature Group D (The fake, Adtran style, DTMF)
|
|
; sf_featdmf: SF Feature Group D (The real thing, MF (domestic, US))
|
|
; sf_featb: SF Feature Group B (MF (domestic, US))
|
|
; e911: E911 (MF) style signalling
|
|
; ss7: Signalling System 7
|
|
; mfcr2: MFC/R2 Signalling. To specify the country variant see 'mfcr2_variant'
|
|
;
|
|
; The following are used for Radio interfaces:
|
|
; fxs_rx: Receive audio/COR on an FXS kewlstart interface (FXO at the
|
|
; channel bank)
|
|
; fxs_tx: Transmit audio/PTT on an FXS loopstart interface (FXO at the
|
|
; channel bank)
|
|
; fxo_rx: Receive audio/COR on an FXO loopstart interface (FXS at the
|
|
; channel bank)
|
|
; fxo_tx: Transmit audio/PTT on an FXO groundstart interface (FXS at
|
|
; the channel bank)
|
|
; em_rx: Receive audio/COR on an E&M interface (1-way)
|
|
; em_tx: Transmit audio/PTT on an E&M interface (1-way)
|
|
; em_txrx: Receive audio/COR AND Transmit audio/PTT on an E&M interface
|
|
; (2-way)
|
|
; em_rxtx: Same as em_txrx (for our dyslexic friends)
|
|
; sf_rx: Receive audio/COR on an SF interface (1-way)
|
|
; sf_tx: Transmit audio/PTT on an SF interface (1-way)
|
|
; sf_txrx: Receive audio/COR AND Transmit audio/PTT on an SF interface
|
|
; (2-way)
|
|
; sf_rxtx: Same as sf_txrx (for our dyslexic friends)
|
|
; ss7: Signalling System 7
|
|
;
|
|
; signalling of a channel can not be changed on a reload.
|
|
;
|
|
;signalling=fxo_ls
|
|
;
|
|
; If you have an outbound signalling format that is different from format
|
|
; specified above (but compatible), you can specify outbound signalling format,
|
|
; (see below). The 'signalling' format specified will be the inbound signalling
|
|
; format. If you only specify 'signalling', then it will be the format for
|
|
; both inbound and outbound.
|
|
;
|
|
; outsignalling can only be one of:
|
|
; em, em_e1, em_w, sf, sf_w, sf_featd, sf_featdmf, sf_featb, featd,
|
|
; featdmf, featdmf_ta, e911, fgccama, fgccamamf
|
|
;
|
|
; outsignalling cannot be changed on a reload.
|
|
;
|
|
;signalling=featdmf
|
|
;
|
|
;outsignalling=featb
|
|
;
|
|
; For Feature Group D Tandem access, to set the default CIC and OZZ use these
|
|
; parameters (Will not be updated on reload):
|
|
;
|
|
;defaultozz=0000
|
|
;defaultcic=303
|
|
;
|
|
; A variety of timing parameters can be specified as well
|
|
; The default values for those are "-1", which is to use the
|
|
; compile-time defaults of the DAHDI kernel modules. The timing
|
|
; parameters, (with the standard default from DAHDI):
|
|
;
|
|
; prewink: Pre-wink time (default 50ms)
|
|
; preflash: Pre-flash time (default 50ms)
|
|
; wink: Wink time (default 150ms)
|
|
; flash: Flash time (default 750ms)
|
|
; start: Start time (default 1500ms)
|
|
; rxwink: Receiver wink time (default 300ms)
|
|
; rxflash: Receiver flashtime (default 1250ms)
|
|
; debounce: Debounce timing (default 600ms)
|
|
;
|
|
; None of them will update on a reload.
|
|
;
|
|
; How long generated tones (DTMF and MF) will be played on the channel
|
|
; (in milliseconds).
|
|
;
|
|
; This is a global, rather than a per-channel setting. It will not be
|
|
; updated on a reload.
|
|
;
|
|
;toneduration=100
|
|
;
|
|
; Whether or not to do distinctive ring detection on FXO lines:
|
|
;
|
|
;usedistinctiveringdetection=yes
|
|
;
|
|
; enable dring detection after caller ID for those countries like Australia
|
|
; where the ring cadence is changed *after* the caller ID spill:
|
|
;
|
|
;distinctiveringaftercid=yes
|
|
;
|
|
; Whether or not to use caller ID:
|
|
;
|
|
usecallerid=yes
|
|
;
|
|
; Type of caller ID signalling in use
|
|
; bell = bell202 as used in US (default)
|
|
; v23 = v23 as used in the UK
|
|
; v23_jp = v23 as used in Japan
|
|
; dtmf = DTMF as used in Denmark, Sweden and Netherlands
|
|
; smdi = Use SMDI for caller ID. Requires SMDI to be enabled (usesmdi).
|
|
;
|
|
;cidsignalling=v23
|
|
;
|
|
; What signals the start of caller ID
|
|
; ring = a ring signals the start (default)
|
|
; polarity = polarity reversal signals the start
|
|
; polarity_IN = polarity reversal signals the start, for India,
|
|
; for dtmf dialtone detection; using DTMF.
|
|
; (see https://wiki.asterisk.org/wiki/display/AST/Caller+ID+in+India)
|
|
; dtmf = causes monitor loop to look for dtmf energy on the
|
|
; incoming channel to initate cid acquisition
|
|
;
|
|
;cidstart=polarity
|
|
;
|
|
; When cidstart=dtmf, the energy level on the line used to trigger dtmf cid
|
|
; acquisition. This number is compared to the average over a packet of audio
|
|
; of the absolute values of 16 bit signed linear samples. The default is set
|
|
; to 256. The choice of 256 is arbitrary. The value you should select should
|
|
; be high enough to prevent false detections while low enough to insure that
|
|
; no dtmf spills are missed.
|
|
;
|
|
;dtmfcidlevel=256
|
|
;
|
|
; Whether or not to hide outgoing caller ID (Override with *67 or *82)
|
|
; (If your dialplan doesn't catch it)
|
|
;
|
|
;hidecallerid=yes
|
|
;
|
|
; Enable if you need to hide just the name and not the number for legacy PBX use.
|
|
; Only applies to PRI channels.
|
|
;hidecalleridname=yes
|
|
;
|
|
; On UK analog lines, the caller hanging up determines the end of calls. So
|
|
; Asterisk hanging up the line may or may not end a call (DAHDI could just as
|
|
; easily be re-attaching to a prior incoming call that was not yet hung up).
|
|
; This option changes the hangup to wait for a dialtone on the line, before
|
|
; marking the line as once again available for use with outgoing calls.
|
|
; Specified in milliseconds, not set by default.
|
|
;waitfordialtone=1000
|
|
;
|
|
; For analog lines, enables Asterisk to use dialtone detection per channel
|
|
; if an incoming call was hung up before it was answered. If dialtone is
|
|
; detected, the call is hung up.
|
|
; no: Disabled. (Default)
|
|
; yes: Look for dialtone for 10000 ms after answer.
|
|
; <number>: Look for dialtone for the specified number of ms after answer.
|
|
; always: Look for dialtone for the entire call. Dialtone may return
|
|
; if the far end hangs up first.
|
|
;
|
|
;dialtone_detect=no
|
|
;
|
|
; The following option enables receiving MWI on FXO lines. The default
|
|
; value is no.
|
|
; The mwimonitor can take the following values
|
|
; no - No mwimonitoring occurs. (default)
|
|
; yes - The same as specifying fsk
|
|
; fsk - the FXO line is monitored for MWI FSK spills
|
|
; fsk,rpas - the FXO line is monitored for MWI FSK spills preceded
|
|
; by a ring pulse alert signal.
|
|
; neon - The fxo line is monitored for the presence of NEON pulses
|
|
; indicating MWI.
|
|
; When detected, an internal Asterisk MWI event is generated so that any other
|
|
; part of Asterisk that cares about MWI state changes is notified, just as if
|
|
; the state change came from app_voicemail.
|
|
; For FSK MWI Spills, the energy level that must be seen before starting the
|
|
; MWI detection process can be set with 'mwilevel'.
|
|
;
|
|
;mwimonitor=no
|
|
;mwilevel=512
|
|
;
|
|
; This option is used in conjunction with mwimonitor. This will get executed
|
|
; when incoming MWI state changes. The script is passed 2 arguments. The
|
|
; first is the corresponding configured mailbox, and the second is 1 or 0,
|
|
; indicating if there are messages waiting or not.
|
|
; Note: app_voicemail mailboxes are in the form of mailbox@context.
|
|
;
|
|
; /usr/local/bin/dahdinotify.sh 501@mailboxes 1
|
|
;
|
|
;mwimonitornotify=/usr/local/bin/dahdinotify.sh
|
|
;
|
|
; The following keyword 'mwisendtype' enables various VMWI methods on FXS lines (if supported).
|
|
; The default is to send FSK only.
|
|
; The following options are available;
|
|
; 'rpas' Ring Pulse Alert Signal, alerts intelligent phones that a FSK message is about to be sent.
|
|
; 'lrev' Line reversed to indicate messages waiting.
|
|
; 'hvdc' 90Vdc OnHook DC voltage to indicate messages waiting.
|
|
; 'hvac' or 'neon' 90Vac OnHook AC voltage to light Neon bulb.
|
|
; 'nofsk' Disables FSK MWI spills from being sent out.
|
|
; It is feasible that multiple options can be enabled.
|
|
;mwisendtype=rpas,lrev
|
|
;
|
|
; Whether or not to enable call waiting on internal extensions
|
|
; With this set to 'yes', busy extensions will hear the call-waiting
|
|
; tone, and can use hook-flash to switch between callers. The Dial()
|
|
; app will not return the "BUSY" result for extensions.
|
|
;
|
|
callwaiting=yes
|
|
;
|
|
; Configure the number of outstanding call waiting calls for internal ISDN
|
|
; endpoints before bouncing the calls as busy. This option is equivalent to
|
|
; the callwaiting option for analog ports.
|
|
; A call waiting call is a SETUP message with no B channel selected.
|
|
; The default is zero to disable call waiting for ISDN endpoints.
|
|
;max_call_waiting_calls=0
|
|
;
|
|
; Allow incoming ISDN call waiting calls.
|
|
; A call waiting call is a SETUP message with no B channel selected.
|
|
;allow_call_waiting_calls=no
|
|
|
|
; Configure the ISDN span to indicate MWI for the list of mailboxes.
|
|
; You can give a comma separated list of up to 8 mailboxes per span.
|
|
; An empty list disables MWI.
|
|
;
|
|
; The default is an empty list.
|
|
;mwi_mailboxes=vm-mailbox{,vm-mailbox}
|
|
; vm-mailbox = Internal voicemail mailbox identifier.
|
|
; Note: app_voicemail mailboxes must be in the form of mailbox@context.
|
|
;mwi_mailboxes=501@mailboxes,502@mailboxes
|
|
|
|
; Configure the ISDN mailbox number sent over the span for MWI mailboxes.
|
|
; The position of the number in the list corresponds to the position in
|
|
; mwi_mailboxes. If either position in mwi_mailboxes or mwi_vm_boxes is
|
|
; empty then that position is disabled.
|
|
;
|
|
; The default is an empty list.
|
|
;mwi_vm_boxes=mailbox_number{,mailbox_number}
|
|
;mwi_vm_boxes=501,502
|
|
|
|
; Configure the ISDN span voicemail controlling numbers for MWI mailboxes.
|
|
; What number to call for a user to retrieve voicemail messages.
|
|
;
|
|
; You can give a comma separated list of numbers. The position of the number
|
|
; corresponds to the position in mwi_mailboxes. If a position is empty then
|
|
; the last number is reused.
|
|
;
|
|
; For example:
|
|
; mwi_vm_numbers=700,,800,,900
|
|
; is equivalent to:
|
|
; mwi_vm_numbers=700,700,800,800,900,900,900,900
|
|
;
|
|
; The default is no number.
|
|
;mwi_vm_numbers=
|
|
|
|
; Whether or not restrict outgoing caller ID (will be sent as ANI only, not
|
|
; available for the user)
|
|
; Mostly use with FXS ports
|
|
; Does nothing. Use hidecallerid instead.
|
|
;
|
|
;restrictcid=no
|
|
;
|
|
; Whether or not to use the caller ID presentation from the Asterisk channel
|
|
; for outgoing calls.
|
|
; See dialplan function CALLERID(pres) for more information.
|
|
; Only applies to PRI and SS7 channels.
|
|
;
|
|
usecallingpres=yes
|
|
;
|
|
; Some countries (UK) have ring tones with different ring tones (ring-ring),
|
|
; which means the caller ID needs to be set later on, and not just after
|
|
; the first ring, as per the default (1).
|
|
;
|
|
;sendcalleridafter = 2
|
|
;
|
|
;
|
|
; Support caller ID on Call Waiting
|
|
;
|
|
callwaitingcallerid=yes
|
|
;
|
|
; Support three-way calling
|
|
;
|
|
threewaycalling=yes
|
|
;
|
|
; For FXS ports (either direct analog or over T1/E1):
|
|
; Support flash-hook call transfer (requires three way calling)
|
|
; Also enables call parking (overrides the 'canpark' parameter)
|
|
;
|
|
; For digital ports using ISDN PRI protocols:
|
|
; Support switch-side transfer (called 2BCT, RLT or other names)
|
|
; This setting must be enabled on both ports involved, and the
|
|
; 'facilityenable' setting must also be enabled to allow sending
|
|
; the transfer to the ISDN switch, since it sent in a FACILITY
|
|
; message.
|
|
; NOTE: This should be disabled for NT PTMP mode. Phones cannot
|
|
; have tromboned calls pushed down to them.
|
|
;
|
|
transfer=yes
|
|
;
|
|
; Allow call parking
|
|
; ('canpark=no' is overridden by 'transfer=yes')
|
|
;
|
|
canpark=yes
|
|
|
|
; Sets the default parking lot for call parking.
|
|
; This is setable per channel.
|
|
; Parkinglots are configured in features.conf
|
|
;
|
|
;parkinglot=plaza
|
|
|
|
;
|
|
; Support call forward variable
|
|
;
|
|
cancallforward=yes
|
|
;
|
|
; Whether or not to support Call Return (*69, if your dialplan doesn't
|
|
; catch this first)
|
|
;
|
|
callreturn=yes
|
|
;
|
|
; Stutter dialtone support: If voicemail is received in the mailbox then
|
|
; taking the phone off hook will cause a stutter dialtone instead of a
|
|
; normal one.
|
|
;
|
|
; Note: app_voicemail mailboxes must be in the form of mailbox@context.
|
|
;
|
|
;mailbox=1234@context
|
|
;
|
|
; Enable echo cancellation
|
|
; Use either "yes", "no", or a power of two from 32 to 256 if you wish to
|
|
; actually set the number of taps of cancellation.
|
|
;
|
|
; Note that when setting the number of taps, the number 256 does not translate
|
|
; to 256 ms of echo cancellation. echocancel=256 means 256 / 8 = 32 ms.
|
|
;
|
|
; Note that if any of your DAHDI cards have hardware echo cancellers,
|
|
; then this setting only turns them on and off; numeric settings will
|
|
; be treated as "yes". There are no special settings required for
|
|
; hardware echo cancellers; when present and enabled in their kernel
|
|
; modules, they take precedence over the software echo canceller compiled
|
|
; into DAHDI automatically.
|
|
;
|
|
;
|
|
echocancel=yes
|
|
;
|
|
; Some DAHDI echo cancellers (software and hardware) support adjustable
|
|
; parameters; these parameters can be supplied as additional options to
|
|
; the 'echocancel' setting. Note that Asterisk does not attempt to
|
|
; validate the parameters or their values, so if you supply an invalid
|
|
; parameter you will not know the specific reason it failed without
|
|
; checking the kernel message log for the error(s) put there by DAHDI.
|
|
;
|
|
;echocancel=128,param1=32,param2=0,param3=14
|
|
;
|
|
; Generally, it is not necessary (and in fact undesirable) to echo cancel when
|
|
; the circuit path is entirely TDM. You may, however, change this behavior
|
|
; by enabling the echo canceller during pure TDM bridging below.
|
|
;
|
|
echocancelwhenbridged=yes
|
|
;
|
|
; In some cases, the echo canceller doesn't train quickly enough and there
|
|
; is echo at the beginning of the call. Enabling echo training will cause
|
|
; DAHDI to briefly mute the channel, send an impulse, and use the impulse
|
|
; response to pre-train the echo canceller so it can start out with a much
|
|
; closer idea of the actual echo. Value may be "yes", "no", or a number of
|
|
; milliseconds to delay before training (default = 400)
|
|
;
|
|
; WARNING: In some cases this option can make echo worse! If you are
|
|
; trying to debug an echo problem, it is worth checking to see if your echo
|
|
; is better with the option set to yes or no. Use whatever setting gives
|
|
; the best results.
|
|
;
|
|
; Note that these parameters do not apply to hardware echo cancellers.
|
|
;
|
|
;echotraining=yes
|
|
;echotraining=800
|
|
;
|
|
; If you are having trouble with DTMF detection, you can relax the DTMF
|
|
; detection parameters. Relaxing them may make the DTMF detector more likely
|
|
; to have "talkoff" where DTMF is detected when it shouldn't be.
|
|
;
|
|
;relaxdtmf=yes
|
|
;
|
|
; Hardware gain settings increase/decrease the analog volume level on a channel.
|
|
; The values are in db (decibels) and can be adjusted in 0.1 dB increments.
|
|
; A positive number increases the volume level on a channel, and a negavive
|
|
; value decreases volume level.
|
|
;
|
|
; Hardware gain settings are only possible on hardware with analog ports
|
|
; because the gain is done on the analog side of the analog/digital conversion.
|
|
;
|
|
; When hardware gains are disabled, Asterisk will NOT touch the gain setting
|
|
; already configured in hardware.
|
|
;
|
|
; hwrxgain: Hardware receive gain for the channel (into Asterisk).
|
|
; Default: disabled
|
|
; hwtxgain: Hardware transmit gain for the channel (out of Asterisk).
|
|
; Default: disabled
|
|
;
|
|
;hwrxgain=disabled
|
|
;hwtxgain=disabled
|
|
;hwrxgain=2.0
|
|
;hwtxgain=3.0
|
|
;
|
|
; Software gain settings digitally increase/decrease the volume level on a channel.
|
|
; The values are in db (decibels). A positive number increases the volume
|
|
; level on a channel, and a negavive value decreases volume level.
|
|
;
|
|
; Software gains work on the digital side of the analog/digital conversion
|
|
; and thus can also work with T1/E1 cards.
|
|
;
|
|
; rxgain: Software receive gain for the channel (into Asterisk). Default: 0.0
|
|
; txgain: Software transmit gain for the channel (out of Asterisk).
|
|
; Default: 0.0
|
|
;
|
|
; cid_rxgain: Add this gain to rxgain when Asterisk expects to receive
|
|
; a Caller ID stream.
|
|
; Default: 5.0 .
|
|
;
|
|
;rxgain=2.0
|
|
;txgain=3.0
|
|
;
|
|
; Dynamic Range Compression: You can also enable dynamic range compression
|
|
; on a channel. This will digitally amplify quiet sounds while leaving louder
|
|
; sounds untouched. This is useful in situations where a linear gain setting
|
|
; would cause clipping. Acceptable values are in the range of 0.0 to around
|
|
; 6.0 with higher values causing more compression to be done.
|
|
;
|
|
; rxdrc: dynamic range compression for the rx channel. Default: 0.0
|
|
; txdrc: dynamic range compression for the tx channel. Default: 0.0
|
|
;
|
|
;rxdrc=1.0
|
|
;txdrc=4.0
|
|
;
|
|
; 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
|
|
;
|
|
; Named ring groups (a.k.a. named call groups) and named pickup groups.
|
|
; If a phone is ringing and it is a member of a group which is one of your
|
|
; named pickup groups, then you can answer it by picking up and dialing *8#.
|
|
; For simple offices, just make these both the same.
|
|
; The number of named groups is not limited.
|
|
;
|
|
;namedcallgroup=engineering,sales,netgroup,protgroup
|
|
;namedpickupgroup=sales
|
|
|
|
; Channel variables to be set for all calls from this channel
|
|
;setvar=CHANNEL=42
|
|
;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
|
|
; cause the given audio file to
|
|
; be played upon completion of
|
|
; an attended transfer to the
|
|
; target of the transfer.
|
|
|
|
;
|
|
; Specify whether the channel should be answered immediately or if the simple
|
|
; switch should provide dialtone, read digits, etc.
|
|
; Note: If immediate=yes the dialplan execution will always start at extension
|
|
; 's' priority 1 regardless of the dialed number!
|
|
;
|
|
;immediate=yes
|
|
;
|
|
; Specify whether flash-hook transfers to 'busy' channels should complete or
|
|
; return to the caller performing the transfer (default is yes).
|
|
;
|
|
;transfertobusy=no
|
|
|
|
; Calls will have the party id user tag set to this string value.
|
|
;
|
|
;cid_tag=
|
|
|
|
; With this set, you can automatically append the MSN of a party
|
|
; to the cid_tag. An '_' is used to separate the tag from the MSN.
|
|
; Applies to ISDN spans.
|
|
; Default is no.
|
|
;
|
|
; Table of what number is appended:
|
|
; outgoing incoming
|
|
; net dialed caller
|
|
; cpe caller dialed
|
|
;
|
|
;append_msn_to_cid_tag=no
|
|
|
|
; caller ID can be set to "asreceived" or a specific number if you want to
|
|
; override it. Note that "asreceived" only applies to trunk interfaces.
|
|
; fullname sets just the
|
|
;
|
|
; fullname: sets just the name part.
|
|
; cid_number: sets just the number part:
|
|
;
|
|
;callerid = 123456
|
|
;
|
|
;callerid = My Name <2564286000>
|
|
; Which can also be written as:
|
|
;cid_number = 2564286000
|
|
;fullname = My Name
|
|
;
|
|
;callerid = asreceived
|
|
;
|
|
; should we use the caller ID from incoming call on DAHDI transfer?
|
|
;
|
|
;useincomingcalleridondahditransfer = yes
|
|
;
|
|
; Add a description for the channel which can be shown through the Asterisk
|
|
; console when executing the 'dahdi show channels' command is run.
|
|
;
|
|
;description=Phone located in lobby
|
|
;
|
|
; AMA flags affects the recording of Call Detail Records. If specified
|
|
; it may be 'default', 'omit', 'billing', or 'documentation'.
|
|
;
|
|
;amaflags=default
|
|
;
|
|
; Channels may be associated with an account code to ease
|
|
; billing
|
|
;
|
|
;accountcode=lss0101
|
|
;
|
|
; ADSI (Analog Display Services Interface) can be enabled on a per-channel
|
|
; basis if you have (or may have) ADSI compatible CPE equipment
|
|
;
|
|
;adsi=yes
|
|
;
|
|
; SMDI (Simplified Message Desk Interface) can be enabled on a per-channel
|
|
; basis if you would like that channel to behave like an SMDI message desk.
|
|
; The SMDI port specified should have already been defined in smdi.conf. The
|
|
; default port is /dev/ttyS0.
|
|
;
|
|
;usesmdi=yes
|
|
;smdiport=/dev/ttyS0
|
|
;
|
|
; On trunk interfaces (FXS) and E&M interfaces (E&M, Wink, Feature Group D
|
|
; etc, it can be useful to perform busy detection either in an effort to
|
|
; detect hangup or for detecting busies. This enables listening for
|
|
; the beep-beep busy pattern.
|
|
;
|
|
;busydetect=yes
|
|
;
|
|
; If busydetect is enabled, it is also possible to specify how many busy tones
|
|
; to wait for before hanging up. The default is 3, but it might be
|
|
; safer to set to 6 or even 8. Mind that the higher the number, the more
|
|
; time that will be needed to hangup a channel, but lowers the probability
|
|
; that you will get random hangups.
|
|
;
|
|
;busycount=6
|
|
;
|
|
; If busydetect is enabled, it is also possible to specify the cadence of your
|
|
; busy signal. In many countries, it is 500msec on, 500msec off. Without
|
|
; busypattern specified, we'll accept any regular sound-silence pattern that
|
|
; repeats <busycount> times as a busy signal. If you specify busypattern,
|
|
; then we'll further check the length of the sound (tone) and silence, which
|
|
; will further reduce the chance of a false positive.
|
|
;
|
|
;busypattern=500,500
|
|
;
|
|
; NOTE: In make menuselect, you'll find further options to tweak the busy
|
|
; detector. If your country has a busy tone with the same length tone and
|
|
; silence (as many countries do), consider enabling the
|
|
; BUSYDETECT_COMPARE_TONE_AND_SILENCE option.
|
|
;
|
|
; To further detect which hangup tone your telco provider is sending, it is
|
|
; useful to use the dahdi_monitor utility to record the audio that main/dsp.c
|
|
; is receiving after the caller hangs up.
|
|
;
|
|
; For FXS (FXO signalled) ports
|
|
; switch the line polarity to signal the connected PBX that an outgoing
|
|
; call was answered by the remote party.
|
|
; For FXO (FXS signalled) ports
|
|
; watch for a polarity reversal to mark when a outgoing call is
|
|
; answered by the remote party.
|
|
;
|
|
;answeronpolarityswitch=yes
|
|
;
|
|
; For FXS (FXO signalled) ports
|
|
; switch the line polarity to signal the connected PBX that the current
|
|
; call was "hung up" by the remote party
|
|
; For FXO (FXS signalled) ports
|
|
; 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 busydetect (unless
|
|
; busypattern is set explicitly). The possible values are:
|
|
; us (default)
|
|
; ca (alias for 'us')
|
|
; cr (Costa Rica)
|
|
; br (Brazil, alias for 'cr')
|
|
; uk
|
|
;
|
|
; This feature can also easily detect false hangups. The symptoms of this is
|
|
; being disconnected in the middle of a call for no reason.
|
|
;
|
|
;callprogress=yes
|
|
;progzone=uk
|
|
;
|
|
; Set the tonezone. Equivalent of the defaultzone settings in
|
|
; /etc/dahdi/system.conf. This sets the tone zone by number.
|
|
; Note that you'd still need to load tonezones (loadzone in
|
|
; /etc/dahdi/system.conf).
|
|
; The default is -1: not to set anything.
|
|
;tonezone = 0 ; 0 is US
|
|
;
|
|
; The number of ANI info digits to expect before the main ANI spill.
|
|
; Switches using ANI-B, -C, and -D will usually send 1 digit. Modern digital
|
|
; systems will send 2, following NANPA ANI II requirements.
|
|
;
|
|
;ani_info_digits=2
|
|
;
|
|
; Time in ms to wait before asterisk sends wink to start ANI spill. Can be
|
|
; shortened if your switch supports it.
|
|
;
|
|
;ani_wink_time=1000
|
|
;
|
|
; Time in ms to wait for each digit in the spill including the ST pulse.
|
|
; This value can affect how long it takes to recognize ANI failures that do
|
|
; not send a ST pulse. If ANI failures take too long to recognize, you can
|
|
; lower this value.
|
|
;
|
|
;ani_timeout=10000
|
|
;
|
|
; FXO (FXS signalled) devices must have a timeout to determine if there was a
|
|
; hangup before the line was answered. This value can be tweaked to shorten
|
|
; how long it takes before DAHDI considers a non-ringing line to have hungup.
|
|
;
|
|
; ringtimeout will not update on a reload.
|
|
;
|
|
;ringtimeout=8000
|
|
;
|
|
; For FXO (FXS signalled) devices, whether to use pulse dial instead of DTMF
|
|
; Pulse digits from phones (FXS devices, FXO signalling) are always
|
|
; detected.
|
|
;
|
|
;pulsedial=yes
|
|
;
|
|
; For fax detection, uncomment one of the following lines. The default is *OFF*
|
|
;
|
|
;faxdetect=both
|
|
;faxdetect=incoming
|
|
;faxdetect=outgoing
|
|
;faxdetect=no
|
|
;
|
|
; When 'faxdetect' is enabled, one could use 'faxdetect_timeout' to disable fax
|
|
; detection after the specified number of seconds into a call. Be aware that
|
|
; outgoing analog channels may consider the channel is answered immediately
|
|
; when dialing completes. Analog does not have a reliable method of detecting
|
|
; when the far end answers. Zero disables the timeout.
|
|
; Default is 0 to disable the timeout.
|
|
;
|
|
;faxdetect_timeout=30
|
|
;
|
|
; When 'faxdetect' is used, one could use 'faxbuffers' to configure the DAHDI
|
|
; transmit buffer policy. The default is *OFF*. When this configuration
|
|
; option is used, the faxbuffer policy will be used for the life of the call
|
|
; after a fax tone is detected. The faxbuffer policy is reverted after the
|
|
; call is torn down. The sample below will result in 6 buffers and a full
|
|
; buffer policy.
|
|
;
|
|
;faxbuffers=>6,full
|
|
;
|
|
; When FXO signalling (FXS device, e.g. analog phone) is used, overlap dialing
|
|
; is typically used. Asterisk has several configurable (per-channel) timeouts
|
|
; to know how long to wait for the next digit. All the values are in
|
|
; milliseconds.
|
|
; * firstdigit_timeout: a longer timeout before any digit is dialed.
|
|
; By default: 16 seconds.
|
|
; * interdigit_timeout: timeout for next digits, if the current number dialed
|
|
; does not match a number in the current context. Default: 8 seconds.
|
|
; * matchdigit_timeout: timeout for next digits, if the current number dialed
|
|
; matches a number in the current context. Default: 3 seconds.
|
|
;
|
|
;firstdigit_timeout=16000
|
|
;interdigit_timeout=8000
|
|
;matchdigit_timeout=3000
|
|
;
|
|
; Configure the default number of DAHDI buffers and the transmit policy to use.
|
|
; This can be used to eliminate data drops when scheduling jitter prevents
|
|
; Asterisk from writing to a DAHDI channel regularly. Most users will probably
|
|
; want "faxbuffers" instead of "buffers".
|
|
;
|
|
; The policies are:
|
|
; immediate - DAHDI will immediately start sending the data to the hardware after
|
|
; Asterisk writes to the channel. This is the default mode. It
|
|
; introduces the least amount of latency but has an increased chance for
|
|
; hardware under runs if Asterisk is not able to keep the DAHDI write
|
|
; queue from going empty.
|
|
; half - DAHDI will wait until half of the configured buffers are full before
|
|
; starting to transmit. This adds latency to the audio but reduces
|
|
; the chance of under runs. Essentially, this is like an in-kernel jitter
|
|
; buffer.
|
|
; full - DAHDI will not start transmitting until all buffers are full.
|
|
; Introduces the most amount of latency and is susceptible to over
|
|
; runs from the Asterisk process.
|
|
;
|
|
; The receive policy is never changed. DAHDI will always pass up audio as soon
|
|
; as possible.
|
|
;
|
|
; The default number of buffers is 4 (from jitterbuffers) and the default policy
|
|
; is immediate.
|
|
;
|
|
;buffers=4,immediate
|
|
;
|
|
; This option specifies what to do when the channel's bridged peer puts the
|
|
; ISDN channel on hold. Settable per logical ISDN span.
|
|
; moh: Generate music-on-hold to the remote party.
|
|
; notify: Send hold notification signaling to the remote party.
|
|
; For ETSI PTP and ETSI PTMP NT links.
|
|
; (The notify setting deprecates the mohinterpret=passthrough setting.)
|
|
; hold: Use HOLD/RETRIEVE signaling to release the B channel while on hold.
|
|
; For ETSI PTMP TE links.
|
|
;
|
|
;moh_signaling=moh
|
|
;
|
|
; 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.
|
|
;
|
|
; This option may be set globally or on a per-channel basis.
|
|
;
|
|
;mohinterpret=default
|
|
;
|
|
; This option specifies which music on hold class to suggest to the peer channel
|
|
; when this channel places the peer on hold. This option may be set globally,
|
|
; or on a per-channel basis.
|
|
;
|
|
;mohsuggest=default
|
|
;
|
|
; PRI channels can have an idle extension and a minunused number. So long as
|
|
; at least "minunused" channels are idle, chan_dahdi will try to call "idledial"
|
|
; on them, and then dump them into the PBX in the "idleext" extension (which
|
|
; is of the form exten@context). When channels are needed the "idle" calls
|
|
; are disconnected (so long as there are at least "minidle" calls still
|
|
; running, of course) to make more channels available. The primary use of
|
|
; this is to create a dynamic service, where idle channels are bundled through
|
|
; multilink PPP, thus more efficiently utilizing combined voice/data services
|
|
; than conventional fixed mappings/muxings.
|
|
;
|
|
; Those settings cannot be changed on reload.
|
|
;
|
|
;idledial=6999
|
|
;idleext=6999@dialout
|
|
;minunused=2
|
|
;minidle=1
|
|
;
|
|
;
|
|
; ignore_failed_channels: Continue even if some channels failed to configure.
|
|
; True by default. Disable this if you can guarantee that DAHDI starts before
|
|
; Asterisk and want to be sure chan_dahdi will not start with broken
|
|
; configuration.
|
|
;
|
|
;ignore_failed_channels = false
|
|
;
|
|
; Configure jitter buffers in DAHDI (each one is 20ms, default is 4)
|
|
; This is set globally, rather than per-channel.
|
|
;
|
|
;jitterbuffers=4
|
|
;
|
|
; ----------------------------- JITTER BUFFER CONFIGURATION --------------------------
|
|
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
|
|
; DAHDI channel. Defaults to "no". An enabled jitterbuffer will
|
|
; be used only if the sending side can create and the receiving
|
|
; side can not accept jitter. The DAHDI channel can't accept jitter,
|
|
; thus an enabled jitterbuffer on the receive DAHDI side will always
|
|
; be used if the sending side can create jitter.
|
|
|
|
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
|
|
|
|
; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
|
|
; resynchronized. Useful to improve the quality of the voice, with
|
|
; big jumps in/broken timestamps, usually sent from exotic devices
|
|
; and programs. Defaults to 1000.
|
|
|
|
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a DAHDI
|
|
; channel. Two implementations are currently available - "fixed"
|
|
; (with size always equals to jbmax-size) and "adaptive" (with
|
|
; variable size, actually the new jb of IAX2). Defaults to fixed.
|
|
|
|
; jbtargetextra = 40 ; This option only affects the jb when 'jbimpl = adaptive' is set.
|
|
; The option represents the number of milliseconds by which the new
|
|
; jitter buffer will pad its size. the default is 40, so without
|
|
; modification, the new jitter buffer will set its size to the jitter
|
|
; value plus 40 milliseconds. increasing this value may help if your
|
|
; network normally has low jitter, but occasionally has spikes.
|
|
|
|
; 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 (overwritten).
|
|
;
|
|
; This setting is global, rather than per-channel. It will not update on
|
|
; a reload, but new and modified cadences will update on dahdi restart.
|
|
; A maximum of 24 cadences may be specified.
|
|
;
|
|
; 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.
|
|
;
|
|
;
|
|
;callerid="Green Phone"<(256) 428-6121>
|
|
;description=Reception Phone ; add a description for 'dahdi show channels'
|
|
;channel => 1
|
|
;callerid="Black Phone"<(256) 428-6122>
|
|
;description=Courtesy Phone
|
|
;channel => 2
|
|
;callerid="CallerID Phone" <(630) 372-1564>
|
|
;description= ; reset the description for following channels
|
|
;channel => 3
|
|
;callerid="Pac Tel Phone" <(256) 428-6124>
|
|
;channel => 4
|
|
;callerid="Uniden Dead" <(256) 428-6125>
|
|
;channel => 5
|
|
;callerid="Cortelco 2500" <(256) 428-6126>
|
|
;channel => 6
|
|
;callerid="Main TA 750" <(256) 428-6127>
|
|
;channel => 44
|
|
;
|
|
; For example, maybe we have some other channels which start out in a
|
|
; different context and use E & M signalling instead.
|
|
;
|
|
;context=remote
|
|
;signaling=em
|
|
;channel => 15
|
|
;channel => 16
|
|
|
|
;signalling=em_w
|
|
;
|
|
; All those in group 0 I'll use for outgoing calls
|
|
;
|
|
; Strip most significant digit (9) before sending
|
|
;
|
|
;stripmsd=1
|
|
;callerid=asreceived
|
|
;group=0
|
|
;signalling=fxs_ls
|
|
;channel => 45
|
|
|
|
;signalling=fxo_ls
|
|
;group=1
|
|
;callerid="Joe Schmoe" <(256) 428-6131>
|
|
;channel => 25
|
|
;callerid="Megan May" <(256) 428-6132>
|
|
;channel => 26
|
|
;callerid="Suzy Queue" <(256) 428-6233>
|
|
;channel => 27
|
|
;callerid="Larry Moe" <(256) 428-6234>
|
|
;channel => 28
|
|
;
|
|
; Sample PRI (CPE) config: Specify the switchtype, the signalling as either
|
|
; pri_cpe or pri_net for CPE or Network termination, and generally you will
|
|
; want to create a single "group" for all channels of the PRI.
|
|
;
|
|
; switchtype cannot be changed on a reload.
|
|
;
|
|
; switchtype = national
|
|
; signalling = pri_cpe
|
|
; group = 2
|
|
; channel => 1-23
|
|
|
|
; Used for distinctive ring support for x100p.
|
|
; You can see the dringX patterns is to set any one of the dringXcontext fields
|
|
; and they will be printed on the console when an inbound call comes in.
|
|
;
|
|
; dringXrange is used to change the acceptable ranges for "tone offsets". Defaults to 10.
|
|
; Note: a range of 0 is NOT what you might expect - it instead forces it to the default.
|
|
; A range of -1 will force it to always match.
|
|
; Anything lower than -1 would presumably cause it to never match.
|
|
;
|
|
;dring1=95,0,0
|
|
;dring1context=internal1
|
|
;dring1range=10
|
|
;dring2=325,95,0
|
|
;dring2context=internal2
|
|
;dring2range=10
|
|
; If no pattern is matched here is where we go.
|
|
;context=default
|
|
;channel => 1
|
|
|
|
; AMI alarm event reporting
|
|
;reportalarms=channels
|
|
;Possible values are:
|
|
;channels - report each channel alarms (current behavior, default for backward compatibility)
|
|
;spans - report an "SpanAlarm" event when the span of any configured channel is alarmed
|
|
;all - report channel and span alarms (aggregated behavior)
|
|
;none - do not report any alarms.
|
|
|
|
; ---------------- Options for use with signalling=ss7 -----------------
|
|
; None of them can be changed by a reload.
|
|
;
|
|
; Variant of SS7 signalling:
|
|
; Options are itu and ansi
|
|
;ss7type = itu
|
|
|
|
; SS7 Called Nature of Address Indicator
|
|
;
|
|
; unknown: Unknown
|
|
; subscriber: Subscriber
|
|
; national: National
|
|
; international: International
|
|
; dynamic: Dynamically selects the appropriate dialplan
|
|
;
|
|
;ss7_called_nai=dynamic
|
|
;
|
|
; SS7 Calling Nature of Address Indicator
|
|
;
|
|
; unknown: Unknown
|
|
; subscriber: Subscriber
|
|
; national: National
|
|
; international: International
|
|
; dynamic: Dynamically selects the appropriate dialplan
|
|
;
|
|
;ss7_calling_nai=dynamic
|
|
;
|
|
;
|
|
; sample 1 for Germany
|
|
;ss7_internationalprefix = 00
|
|
;ss7_nationalprefix = 0
|
|
;ss7_subscriberprefix =
|
|
;ss7_unknownprefix =
|
|
;
|
|
|
|
; This option is used to disable automatic sending of ACM when the call is started
|
|
; in the dialplan. If you do use this option, you will need to use the Proceeding()
|
|
; application in the dialplan to send ACM or enable ss7_autoacm below.
|
|
;ss7_explicitacm=yes
|
|
|
|
; Use this option to automatically send ACM when the call rings or is answered and
|
|
; has not seen proceeding yet. If you use this option, you should disable ss7_explicitacm.
|
|
; You may still use Proceeding() to explicitly send an ACM from the dialplan.
|
|
;ss7_autoacm=yes
|
|
|
|
; Create the linkset with all CICs in hardware remotely blocked state.
|
|
;ss7_initialhwblo=yes
|
|
|
|
; This option is whether or not to trust the remote echo control indication. This means
|
|
; that in cases where echo control is reported by the remote end, we will trust them and
|
|
; not enable echo cancellation on the call.
|
|
;ss7_use_echocontrol=yes
|
|
|
|
; This option is to set what our echo control indication is to the other end. Set to
|
|
; yes to indicate that we are using echo cancellation or no if we are not.
|
|
;ss7_default_echocontrol=yes
|
|
|
|
; All settings apply to linkset 1
|
|
;linkset = 1
|
|
|
|
; Set the Signaling Link Code (SLC) for each sigchan.
|
|
; If you manually set any you need to manually set all.
|
|
; Should be defined before sigchan.
|
|
; The default SLC starts with zero and increases for each defined sigchan.
|
|
;slc=
|
|
|
|
; Point code of the linkset. For ITU, this is the decimal number
|
|
; format of the point code. For ANSI, this can either be in decimal
|
|
; number format or in the xxx-xxx-xxx format
|
|
;pointcode = 1
|
|
|
|
; Point code of node adjacent to this signalling link (Possibly the STP between you and
|
|
; your destination). Point code format follows the same rules as above.
|
|
;adjpointcode = 2
|
|
|
|
; Default point code that you would like to assign to outgoing messages (in case of
|
|
; routing through STPs, or using A links). Point code format follows the same rules
|
|
; as above.
|
|
;defaultdpc = 3
|
|
|
|
; Begin CIC (Circuit indication codes) count with this number
|
|
;cicbeginswith = 1
|
|
|
|
; What the MTP3 network indicator bits should be set to. Choices are
|
|
; national, national_spare, international, international_spare
|
|
;networkindicator=international
|
|
|
|
; First signalling channel
|
|
;sigchan = 48
|
|
|
|
; Additional signalling channel for this linkset (So you can have a linkset
|
|
; with two signalling links in it). It seems like a silly way to do it, but
|
|
; for linksets with multiple signalling links, you add an additional sigchan
|
|
; line for every additional signalling link on the linkset.
|
|
;sigchan = 96
|
|
|
|
; Channels to associate with CICs on this linkset
|
|
;channel = 25-47
|
|
;
|
|
|
|
; Set this option if you wish to send an Information Request Message (INR) request
|
|
; if no calling party number is specified. This will attempt to tell the other end
|
|
; to send it anyways. Should be defined after sigchan.
|
|
;inr_if_no_calling=yes
|
|
|
|
; Set this to set whether or not the originating access is (non) ISDN in the forward and
|
|
; backward call indicators. Should be defined after sigchan
|
|
;non_isdn_access=yes
|
|
|
|
; This sets the number of binary places to shift the CIC when doing load balancing between
|
|
; sigchans on a linkset. Should be defined after sigchan. Default 0
|
|
;sls_shift = 0
|
|
|
|
; Send custom cause_location value
|
|
; Should be defined after sigchan. Default 1 (private local)
|
|
;cause_location=1
|
|
|
|
; SS7 timers (ISUP and MTP3) should be explicitly defined for each linkset to be used.
|
|
; For a full list of supported timers and their default values (applicable for both ITU
|
|
; and ANSI) see ss7.timers
|
|
; Should be defined after sigchan
|
|
;#include ss7.timers
|
|
|
|
; For more information on setting up SS7, see the README file in libss7 or
|
|
; https://wiki.asterisk.org/wiki/display/AST/Signaling+System+Number+7
|
|
; ----------------- SS7 Options ----------------------------------------
|
|
|
|
; ---------------- Options for use with signalling=mfcr2 --------------
|
|
|
|
; MFC-R2 signaling has lots of variants from country to country and even sometimes
|
|
; minor variants inside the same country. The only mandatory parameters here are:
|
|
; mfcr2_variant, mfcr2_max_ani and mfcr2_max_dnis.
|
|
; IT IS RECOMMENDED that you leave the default values (leaving it commented) for the
|
|
; other parameters unless you have problems or you have been instructed to change some
|
|
; parameter. OpenR2 library uses the mfcr2_variant parameter to try to determine the
|
|
; best defaults for your country, also refer to the OpenR2 package directory
|
|
; doc/asterisk/ where you can find sample configurations for some countries. If you
|
|
; want to contribute your configs for a particular country send them to the e-mail
|
|
; of the primary OpenR2 developer that you can find in the AUTHORS file of the OpenR2 package
|
|
|
|
; MFC/R2 variant. This depends on the OpenR2 supported variants
|
|
; A list of values can be found by executing the openr2 command r2test -l
|
|
; some valid values are:
|
|
; ar (Argentina)
|
|
; br (Brazil)
|
|
; mx (Mexico)
|
|
; ph (Philippines)
|
|
; itu (per ITU spec)
|
|
; mfcr2_variant=mx
|
|
|
|
; Max amount of ANI to ask for
|
|
; mfcr2_max_ani=10
|
|
|
|
; Max amount of DNIS to ask for
|
|
; mfcr2_max_dnis=4
|
|
|
|
; whether or not to get the ANI before getting DNIS.
|
|
; some telcos require ANI first some others do not care
|
|
; if this go wrong, change this value
|
|
; mfcr2_get_ani_first=no
|
|
|
|
; Caller Category to send
|
|
; national_subscriber
|
|
; national_priority_subscriber
|
|
; international_subscriber
|
|
; international_priority_subscriber
|
|
; collect_call
|
|
; usually national_subscriber works just fine
|
|
; you can change this setting from the dialplan
|
|
; by setting the variable MFCR2_CATEGORY
|
|
; (remember to set _MFCR2_CATEGORY from originating channels)
|
|
; MFCR2_CATEGORY will also be a variable available in your context
|
|
; on incoming calls set to the value received from the far end
|
|
; mfcr2_category=national_subscriber
|
|
|
|
; Call logging is stored at the Asterisk
|
|
; logging directory specified in asterisk.conf
|
|
; plus mfcr2/<whatever you put here>
|
|
; if you specify 'span1' here and asterisk.conf has
|
|
; as logging directory /var/log/asterisk then the full
|
|
; path to your MFC/R2 call logs will be /var/log/asterisk/mfcr2/span1
|
|
; (the directory will be automatically created if not present already)
|
|
; remember to set mfcr2_call_files=yes
|
|
; mfcr2_logdir=span1
|
|
|
|
; whether or not to drop call files into mfcr2_logdir
|
|
; mfcr2_call_files=yes|no
|
|
|
|
; MFC/R2 valid logging values are: all,error,warning,debug,notice,cas,mf,stack,nothing
|
|
; error,warning,debug and notice are self-descriptive
|
|
; 'cas' is for logging ABCD CAS tx and rx
|
|
; 'mf' is for logging of the Multi Frequency tones
|
|
; 'stack' is for very verbose output of the channel and context call stack, only useful
|
|
; if you are debugging a crash or want to learn how the library works. The stack logging
|
|
; will be only enabled if the openr2 library was compiled with -DOR2_TRACE_STACKS
|
|
; You can mix up values, like: loglevel=error,debug,mf to log just error, debug and
|
|
; multi frequency messages
|
|
; 'all' is a special value to log all the activity
|
|
; 'nothing' is a clean-up value, in case you want to not log any activity for
|
|
; a channel or group of channels
|
|
; BE AWARE that the level of output logged will ALSO depend on
|
|
; the value you have in logger.conf, if you disable output in logger.conf
|
|
; then it does not matter you specify 'all' here, nothing will be logged
|
|
; so logger.conf has the last word on what is going to be logged
|
|
; mfcr2_logging=all
|
|
|
|
; MFC/R2 value in milliseconds for the MF timeout. Any negative value
|
|
; means 'default', smaller values than 500ms are not recommended
|
|
; and can cause malfunctioning. If you experience protocol error
|
|
; due to MF timeout try incrementing this value in 500ms steps
|
|
; mfcr2_mfback_timeout=-1
|
|
|
|
; MFC/R2 value in milliseconds for the metering pulse timeout.
|
|
; Metering pulses are sent by some telcos for some R2 variants
|
|
; during a call presumably for billing purposes to indicate costs,
|
|
; however this pulses use the same signal that is used to indicate
|
|
; call hangup, therefore a timeout is sometimes required to distinguish
|
|
; between a *real* hangup and a billing pulse that should not
|
|
; last more than 500ms, If you experience call drops after some
|
|
; minutes of being stablished try setting a value of some ms here,
|
|
; values greater than 500ms are not recommended.
|
|
; BE AWARE that choosing the proper protocol mfcr2_variant parameter
|
|
; implicitly sets a good recommended value for this timer, use this
|
|
; parameter only when you *really* want to override the default, otherwise
|
|
; just comment out this value or put a -1
|
|
; Any negative value means 'default'.
|
|
; mfcr2_metering_pulse_timeout=-1
|
|
|
|
; Brazil uses a special calling party category for collect calls (llamadas por cobrar)
|
|
; instead of using the operator (as in Mexico). The R2 spec in Brazil says a special GB tone
|
|
; should be used to reject collect calls. If you want to ALLOW collect calls specify 'yes',
|
|
; if you want to BLOCK collect calls then say 'no'. Default is to block collect calls.
|
|
; (see also 'mfcr2_double_answer')
|
|
; mfcr2_allow_collect_calls=no
|
|
|
|
; This feature is related but independent of mfcr2_allow_collect_calls
|
|
; Some PBX's require a double-answer process to block collect calls, if
|
|
; you ever have problems blocking collect calls using Group B signals (mfcr2_allow_collect_calls=no)
|
|
; then you may want to try with mfcr2_double_answer=yes, this will cause that every answer signal
|
|
; is changed by answer->clear back->answer (sort of a flash)
|
|
; (see also 'mfcr2_allow_collect_calls')
|
|
; mfcr2_double_answer=no
|
|
|
|
; This feature allows to skip the use of Group B/II signals and go directly
|
|
; to the accepted state for incoming calls
|
|
; mfcr2_immediate_accept=no
|
|
|
|
; You most likely dont need this feature. Default is yes.
|
|
; When this is set to yes, all calls that are offered (incoming calls) which
|
|
; DNIS is valid (exists in extensions.conf) and pass collect call validation
|
|
; will be accepted with a Group B tone (either call with charge or not, depending on mfcr2_charge_calls)
|
|
; with this set to 'no' then the call will NOT be accepted on offered, and the call will start its
|
|
; execution in extensions.conf without being accepted until the channel is answered (either with Answer() or
|
|
; any other application resulting in the channel being answered).
|
|
; This can be set to 'no' if your telco or PBX needs the hangup cause to be set accurately
|
|
; when this option is set to no you must explicitly accept the call with DAHDIAcceptR2Call
|
|
; or implicitly through the Answer() application.
|
|
; mfcr2_accept_on_offer=yes
|
|
|
|
; Skip request of calling party category and ANI
|
|
; you need openr2 >= 1.2.0 to use this feature
|
|
; mfcr2_skip_category=no
|
|
|
|
; WARNING: advanced users only! I really mean it
|
|
; this parameter is commented by default because
|
|
; YOU DON'T NEED IT UNLESS YOU REALLY GROK MFC/R2
|
|
; READ COMMENTS on doc/r2proto.conf in openr2 package
|
|
; for more info
|
|
; mfcr2_advanced_protocol_file=/path/to/r2proto.conf
|
|
|
|
; Brazil use a special signal to force the release of the line (hangup) from the
|
|
; backward perspective. When mfcr2_forced_release=no, the normal clear back signal
|
|
; will be sent on hangup, which is OK for all mfcr2 variants I know of, except for
|
|
; Brazilian variant, where the central will leave the line up for several seconds (30, 60)
|
|
; which sometimes is not what people really want. When mfcr2_forced_release=yes, a different
|
|
; signal will be sent to hangup the call indicating that the line should be released immediately
|
|
; mfcr2_forced_release=no
|
|
|
|
; Whether or not report to the other end 'accept call with charge'
|
|
; This setting has no effect with most telecos, usually is safe
|
|
; leave the default (yes), but once in a while when interconnecting with
|
|
; old PBXs this may be useful.
|
|
; Concretely this affects the Group B signal used to accept calls
|
|
; The application DAHDIAcceptR2Call can also be used to decide this
|
|
; in the dial plan in a per-call basis instead of doing it here for all calls
|
|
; mfcr2_charge_calls=yes
|
|
|
|
; ---------------- END of options to be used with signalling=mfcr2
|
|
|
|
; Configuration Sections
|
|
; ~~~~~~~~~~~~~~~~~~~~~~
|
|
; You can also configure channels in a separate chan_dahdi.conf section. In
|
|
; this case the keyword 'channel' is not used. Instead the keyword
|
|
; 'dahdichan' is used (as in users.conf) - configuration is only processed
|
|
; in a section where the keyword dahdichan is used. It will only be
|
|
; processed in the end of the section. Thus the following section:
|
|
;
|
|
;[phones]
|
|
;echocancel = 64
|
|
;dahdichan = 1-8
|
|
;group = 1
|
|
;
|
|
; Is somewhat equivalent to the following snippet in the section
|
|
; [channels]:
|
|
;
|
|
;echocancel = 64
|
|
;group = 1
|
|
;channel => 1-8
|
|
;
|
|
; When starting a new section almost all of the configuration values are
|
|
; copied from their values at the end of the section [channels] in
|
|
; chan_dahdi.conf and [general] in users.conf - one section's configuration
|
|
; does not affect another one's.
|
|
;
|
|
; Instead of letting common configuration values "slide through" you can
|
|
; use configuration templates to easily keep the common part in one
|
|
; place and override where needed.
|
|
;
|
|
;[phones](!)
|
|
;echocancel = yes
|
|
;group = 0,4
|
|
;callgroup = 3
|
|
;pickupgroup = 3
|
|
;threewaycalling = yes
|
|
;transfer = yes
|
|
;context = phones
|
|
;faxdetect = incoming
|
|
;
|
|
;[phone-1](phones)
|
|
;dahdichan = 1
|
|
;callerid = My Name <501>
|
|
;mailbox = 501@mailboxes
|
|
;
|
|
;
|
|
;[fax](phones)
|
|
;dahdichan = 2
|
|
;faxdetect = no
|
|
;context = fax
|
|
;
|
|
;[phone-3](phones)
|
|
;dahdichan = 3
|
|
;pickupgroup = 3,4
|