835 lines
30 KiB
Plaintext
835 lines
30 KiB
Plaintext
; extensions.conf - the Asterisk dial plan
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;
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; Static extension configuration file, used by
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; the pbx_config module. This is where you configure all your
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; inbound and outbound calls in Asterisk.
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;
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; This configuration file is reloaded
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; - With the "dialplan reload" command in the CLI
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; - With the "reload" command (that reloads everything) in the CLI
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;
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; The "General" category is for certain variables.
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;
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[general]
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;
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; If static is set to no, or omitted, then the pbx_config will rewrite
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; this file when extensions are modified. Remember that all comments
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; made in the file will be lost when that happens.
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;
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; XXX Not yet implemented XXX
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;
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static=yes
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;
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; if static=yes and writeprotect=no, you can save dialplan by
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; CLI command "dialplan save" too
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;
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writeprotect=no
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;
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; If autofallthrough is set, then if an extension runs out of
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; things to do, it will terminate the call with BUSY, CONGESTION
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; or HANGUP depending on Asterisk's best guess. This is the default.
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;
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; If autofallthrough is not set, then if an extension runs out of
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; things to do, Asterisk will wait for a new extension to be dialed
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; (this is the original behavior of Asterisk 1.0 and earlier).
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;
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;autofallthrough=no
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;
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;
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;
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; If extenpatternmatchnew is set (true, yes, etc), then a new algorithm that uses
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; a Trie to find the best matching pattern is used. In dialplans
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; with more than about 20-40 extensions in a single context, this
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; new algorithm can provide a noticeable speedup.
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; With 50 extensions, the speedup is 1.32x
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; with 88 extensions, the speedup is 2.23x
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; with 138 extensions, the speedup is 3.44x
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; with 238 extensions, the speedup is 5.8x
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; with 438 extensions, the speedup is 10.4x
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; With 1000 extensions, the speedup is ~25x
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; with 10,000 extensions, the speedup is 374x
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; Basically, the new algorithm provides a flat response
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; time, no matter the number of extensions.
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;
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; By default, the old pattern matcher is used.
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;
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; ****This is a new feature! *********************
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; The new pattern matcher is for the brave, the bold, and
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; the desperate. If you have large dialplans (more than about 50 extensions
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; in a context), and/or high call volume, you might consider setting
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; this value to "yes" !!
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; Please, if you try this out, and are forced to return to the
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; old pattern matcher, please report your reasons in a bug report
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; on bugs.digium.com. We have made good progress in providing something
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; compatible with the old matcher; help us finish the job!
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;
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; This value can be switched at runtime using the cli command "dialplan set extenpatternmatchnew true"
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; or "dialplan set extenpatternmatchnew false", so you can experiment to your hearts content.
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;
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;extenpatternmatchnew=no
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;
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; If clearglobalvars is set, global variables will be cleared
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; and reparsed on a dialplan reload, or Asterisk reload.
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;
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; If clearglobalvars is not set, then global variables will persist
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; through reloads, and even if deleted from the extensions.conf or
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; one of its included files, will remain set to the previous value.
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;
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; NOTE: A complication sets in, if you put your global variables into
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; the AEL file, instead of the extensions.conf file. With clearglobalvars
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; set, a "reload" will often leave the globals vars cleared, because it
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; is not unusual to have extensions.conf (which will have no globals)
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; load after the extensions.ael file (where the global vars are stored).
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; So, with "reload" in this particular situation, first the AEL file will
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; clear and then set all the global vars, then, later, when the extensions.conf
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; file is loaded, the global vars are all cleared, and then not set, because
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; they are not stored in the extensions.conf file.
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;
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clearglobalvars=no
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;
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; User context is where entries from users.conf are registered. The
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; default value is 'default'
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;
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;userscontext=default
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;
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; You can include other config files, use the #include command
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; (without the ';'). Note that this is different from the "include" command
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; that includes contexts within other contexts. The #include command works
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; in all asterisk configuration files.
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;#include "filename.conf"
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;#include <filename.conf>
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;#include filename.conf
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;
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; You can execute a program or script that produces config files, and they
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; will be inserted where you insert the #exec command. The #exec command
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; works on all asterisk configuration files. However, you will need to
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; activate them within asterisk.conf with the "execincludes" option. They
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; are otherwise considered a security risk.
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;#exec /opt/bin/build-extra-contexts.sh
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;#exec /opt/bin/build-extra-contexts.sh --foo="bar"
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;#exec </opt/bin/build-extra-contexts.sh --foo="bar">
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;#exec "/opt/bin/build-extra-contexts.sh --foo=\"bar\""
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;
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; The "Globals" category contains global variables that can be referenced
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; in the dialplan with the GLOBAL dialplan function:
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; ${GLOBAL(VARIABLE)}
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; ${${GLOBAL(VARIABLE)}} or ${text${GLOBAL(VARIABLE)}} or any hybrid
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; Unix/Linux environmental variables can be reached with the ENV dialplan
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; function: ${ENV(VARIABLE)}
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;
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[globals]
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CONSOLE=Console/dsp ; Console interface for demo
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;CONSOLE=DAHDI/1
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;CONSOLE=Phone/phone0
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IAXINFO=guest ; IAXtel username/password
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;IAXINFO=myuser:mypass
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TRUNK=DAHDI/G2 ; Trunk interface
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;
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; Note the 'G2' in the TRUNK variable above. It specifies which group (defined
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; in chan_dahdi.conf) to dial, i.e. group 2, and how to choose a channel to use
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; in the specified group. The four possible options are:
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;
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; g: select the lowest-numbered non-busy DAHDI channel
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; (aka. ascending sequential hunt group).
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; G: select the highest-numbered non-busy DAHDI channel
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; (aka. descending sequential hunt group).
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; r: use a round-robin search, starting at the next highest channel than last
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; time (aka. ascending rotary hunt group).
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; R: use a round-robin search, starting at the next lowest channel than last
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; time (aka. descending rotary hunt group).
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;
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TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
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;TRUNK=IAX2/user:pass@provider
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;FREENUMDOMAIN=mydomain.com ; domain to send on outbound
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; freenum calls (uses outbound-freenum
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; context)
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;
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; WARNING WARNING WARNING WARNING
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; If you load any other extension configuration engine, such as pbx_ael.so,
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; your global variables may be overridden by that file. Please take care to
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; use only one location to set global variables, and you will likely save
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; yourself a ton of grief.
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; WARNING WARNING WARNING WARNING
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;
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; Any category other than "General" and "Globals" represent
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; extension contexts, which are collections of extensions.
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;
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; Extension names may be numbers, letters, or combinations
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; thereof. If an extension name is prefixed by a '_'
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; character, it is interpreted as a pattern rather than a
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; literal. In patterns, some characters have special meanings:
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;
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; X - any digit from 0-9
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; Z - any digit from 1-9
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; N - any digit from 2-9
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; [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
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; . - wildcard, matches anything remaining (e.g. _9011. matches
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; anything starting with 9011 excluding 9011 itself)
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; ! - wildcard, causes the matching process to complete as soon as
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; it can unambiguously determine that no other matches are possible
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;
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; For example, the extension _NXXXXXX would match normal 7 digit dialings,
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; while _1NXXNXXXXXX would represent an area code plus phone number
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; preceded by a one.
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;
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; Each step of an extension is ordered by priority, which must always start
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; with 1 to be considered a valid extension. The priority "next" or "n" means
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; the previous priority plus one, regardless of whether the previous priority
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; was associated with the current extension or not. The priority "same" or "s"
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; means the same as the previously specified priority, again regardless of
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; whether the previous entry was for the same extension. Priorities may be
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; immediately followed by a plus sign and another integer to add that amount
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; (most useful with 's' or 'n'). Priorities may then also have an alias, or
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; label, in parentheses after their name which can be used in goto situations.
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;
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; Contexts contain several lines, one for each step of each extension. One may
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; include another context in the current one as well, optionally with a date
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; and time. Included contexts are included in the order they are listed.
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; Switches may also be included within a context. The order of matching within
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; a context is always exact extensions, pattern match extensions, includes, and
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; switches. Includes are always processed depth-first. So for example, if you
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; would like a switch "A" to match before context "B", simply put switch "A" in
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; an included context "C", where "C" is included in your original context
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; before "B".
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;
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;[context]
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;exten => someexten,{priority|label{+|-}offset}[(alias)],application(arg1,arg2,...)
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;
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; Timing list for includes is
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;
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; <time range>,<days of week>,<days of month>,<months>[,<timezone>]
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;
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; Note that ranges may be specified to wrap around the ends. Also, minutes are
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; fine-grained only down to the closest even minute.
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;
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;include => daytime,9:00-17:00,mon-fri,*,*
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;include => weekend,*,sat-sun,*,*
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;include => weeknights,17:02-8:58,mon-fri,*,*
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;
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; ignorepat can be used to instruct drivers to not cancel dialtone upon receipt
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; of a particular pattern. The most commonly used example is of course '9'
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; like this:
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;
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;ignorepat => 9
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;
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; so that dialtone remains even after dialing a 9. Please note that ignorepat
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; only works with channels which receive dialtone from the PBX, such as DAHDI,
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; Phone, and VPB. Other channels, such as SIP and MGCP, which generate their
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; own dialtone and converse with the PBX only after a number is complete, are
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; generally unaffected by ignorepat (unless DISA or another method is used to
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; generate a dialtone after answering the channel).
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;
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;
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; Sample entries for extensions.conf
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;
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;
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[dundi-e164-canonical]
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;include => stdexten
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;
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; List canonical entries here
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;
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;exten => 12564286000,1,Gosub(6000,stdexten(IAX2/foo))
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;exten => 12564286000,n,Goto(default,s,1) ; exited Voicemail
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;exten => _125642860XX,1,Dial(IAX2/otherbox/${EXTEN:7})
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[dundi-e164-customers]
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;
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; If you are an ITSP or Reseller, list your customers here.
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;
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;exten => _12564286000,1,Dial(SIP/customer1)
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;exten => _12564286001,1,Dial(IAX2/customer2)
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[dundi-e164-via-pstn]
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;
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; If you are freely delivering calls to the PSTN, list them here
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;
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;exten => _1256428XXXX,1,Dial(DAHDI/G2/${EXTEN:7}) ; Expose all of 256-428
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;exten => _1256325XXXX,1,Dial(DAHDI/G2/${EXTEN:7}) ; Ditto for 256-325
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[dundi-e164-local]
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;
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; Context to put your dundi IAX2 or SIP user in for
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; full access
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;
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include => dundi-e164-canonical
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include => dundi-e164-customers
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include => dundi-e164-via-pstn
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[dundi-e164-switch]
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;
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; Just a wrapper for the switch
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;
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switch => DUNDi/e164
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[dundi-e164-lookup]
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;
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; Locally to lookup, try looking for a local E.164 solution
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; then try DUNDi if we don't have one.
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;
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include => dundi-e164-local
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include => dundi-e164-switch
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;
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; DUNDi can also be implemented as a Macro instead of using
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; the Local channel driver.
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;
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[macro-dundi-e164]
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;
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; ARG1 is the extension to Dial
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;
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; Extension "s" is not a wildcard extension that matches "anything".
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; In macros, it is the start extension. In most other cases,
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; you have to goto "s" to execute that extension.
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;
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; For wildcard matches, see above - all pattern matches start with
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; an underscore.
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exten => s,1,Goto(${ARG1},1)
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include => dundi-e164-lookup
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;
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; Here are the entries you need to participate in the IAXTEL
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; call routing system. Most IAXTEL numbers begin with 1-700, but
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; there are exceptions. For more information, and to sign
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; up, please go to www.gnophone.com or www.iaxtel.com
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;
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[iaxtel700]
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exten => _91700XXXXXXX,1,Dial(IAX2/${GLOBAL(IAXINFO)}@iaxtel.com/${EXTEN:1}@iaxtel)
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;
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; The SWITCH statement permits a server to share the dialplan with
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; another server. Use with care: Reciprocal switch statements are not
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; allowed (e.g. both A -> B and B -> A), and the switched server needs
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; to be on-line or else dialing can be severly delayed.
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;
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[iaxprovider]
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;switch => IAX2/user:[key]@myserver/mycontext
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[trunkint]
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;
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; International long distance through trunk
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;
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exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
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exten => _9011.,n,Dial(${GLOBAL(TRUNK)}/${FILTER(0-9,${EXTEN:${GLOBAL(TRUNKMSD)}})})
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[trunkld]
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;
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; Long distance context accessed through trunk
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;
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exten => _91NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN:1})
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exten => _91NXXNXXXXXX,n,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
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[trunklocal]
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;
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; Local seven-digit dialing accessed through trunk interface
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;
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exten => _9NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
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[trunktollfree]
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;
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; Long distance context accessed through trunk interface
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;
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exten => _91800NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
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exten => _91888NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
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exten => _91877NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
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exten => _91866NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
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[international]
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;
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; Master context for international long distance
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;
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ignorepat => 9
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include => longdistance
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include => trunkint
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[longdistance]
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;
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; Master context for long distance
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;
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ignorepat => 9
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include => local
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include => trunkld
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[local]
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;
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; Master context for local, toll-free, and iaxtel calls only
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;
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ignorepat => 9
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include => default
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include => trunklocal
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include => iaxtel700
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include => trunktollfree
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include => iaxprovider
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;Include parkedcalls (or the context you define in features conf)
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;to enable call parking.
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include => parkedcalls
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;
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; You can use an alternative switch type as well, to resolve
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; extensions that are not known here, for example with remote
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; IAX switching you transparently get access to the remote
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; Asterisk PBX
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;
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; switch => IAX2/user:password@bigserver/local
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;
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; An "lswitch" is like a switch but is literal, in that
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; variable substitution is not performed at load time
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; but is passed to the switch directly (presumably to
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; be substituted in the switch routine itself)
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;
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; lswitch => Loopback/12${EXTEN}@othercontext
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;
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; An "eswitch" is like a switch but the evaluation of
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; variable substitution is performed at runtime before
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; being passed to the switch routine.
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;
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; eswitch => IAX2/context@${CURSERVER}
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; The following two contexts are a template to enable the ability to dial
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; ISN numbers. For more information about what an ISN number is, please see
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; http://www.freenum.org.
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;
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; This is the dialing hook. use:
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; include => outbound-freenum
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[outbound-freenum]
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; We'll add more digits as needed. The purpose is to dial things
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; like extension numbers at domains (ITAD number) so we're matching
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; on lengths of 1 through 6 prior to the separator (the asterisk [*])
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;
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exten => _X*X!,1,Goto(outbound-freenum2,${EXTEN},1)
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exten => _XX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
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exten => _XXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
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exten => _XXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
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exten => _XXXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
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exten => _XXXXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
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[outbound-freenum2]
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; This is the handler which performs the dialing logic. It is called
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; from the [outbound-freenum] context
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;
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exten => _X!,1,Verbose(2,Performing ISN lookup for ${EXTEN})
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same => n,Set(SUFFIX=${CUT(EXTEN,*,2-)}) ; make sure the suffix is all digits as well
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same => n,GotoIf($["${FILTER(0-9,${SUFFIX})}" != "${SUFFIX}"]?fn-CONGESTION,1)
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; filter out bad characters per the README-SERIOUSLY.best-practices.txt document
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same => n,Set(TIMEOUT(absolute)=10800)
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same => n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)}) ; perform our lookup with freenum.org
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same => n,GotoIf($["${isnresult}" != ""]?from)
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same => n,Set(DIALSTATUS=CONGESTION)
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same => n,Goto(fn-CONGESTION,1)
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same => n(from),Set(__SIPFROMUSER=${CALLERID(num)})
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same => n,GotoIf($["${GLOBAL(FREENUMDOMAIN)}" = ""]?dial) ; check if we set the FREENUMDOMAIN global variable in [global]
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same => n,Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)}) ; if we did set it, then we'll use it for our outbound dialing domain
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same => n(dial),Dial(SIP/${isnresult},40)
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same => n,Goto(fn-${DIALSTATUS},1)
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exten => fn-BUSY,1,Busy()
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exten => _f[n]-.,1,NoOp(ISN: ${DIALSTATUS})
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same => n,Congestion()
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[macro-trunkdial]
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;
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; Standard trunk dial macro (hangs up on a dialstatus that should
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; terminate call)
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|
; ${ARG1} - What to dial
|
|
;
|
|
exten => s,1,Dial(${ARG1})
|
|
exten => s,n,Goto(s-${DIALSTATUS},1)
|
|
exten => s-NOANSWER,1,Hangup
|
|
exten => s-BUSY,1,Hangup
|
|
exten => _s-.,1,NoOp
|
|
|
|
[stdexten]
|
|
;
|
|
; Standard extension subroutine:
|
|
; ${EXTEN} - Extension
|
|
; ${ARG1} - Device(s) to ring
|
|
; ${ARG2} - Optional context in Voicemail
|
|
;
|
|
; Note that the current version will drop through to the next priority in the
|
|
; case of their pressing '#'. This gives more flexibility in what do to next:
|
|
; you can prompt for a new extension, or drop the call, or send them to a
|
|
; general delivery mailbox, or...
|
|
;
|
|
; The use of the LOCAL() function is purely for convenience. Any variable
|
|
; initially declared as LOCAL() will disappear when the innermost Gosub context
|
|
; in which it was declared returns. Note also that you can declare a LOCAL()
|
|
; variable on top of an existing variable, and its value will revert to its
|
|
; previous value (before being declared as LOCAL()) upon Return.
|
|
;
|
|
exten => _X.,50000(stdexten),NoOp(Start stdexten)
|
|
exten => _X.,n,Set(LOCAL(ext)=${EXTEN})
|
|
exten => _X.,n,Set(LOCAL(dev)=${ARG1})
|
|
exten => _X.,n,Set(LOCAL(cntx)=${ARG2})
|
|
exten => _X.,n,Set(LOCAL(mbx)=${ext}${IF($[!${ISNULL(${cntx})}]?@${cntx})})
|
|
exten => _X.,n,Dial(${dev},20) ; Ring the interface, 20 seconds maximum
|
|
exten => _X.,n,Goto(stdexten-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
|
|
|
|
exten => stdexten-NOANSWER,1,Voicemail(${mbx},u) ; If unavailable, send to voicemail w/ unavail announce
|
|
exten => stdexten-NOANSWER,n,Return() ; If they press #, return to start
|
|
|
|
exten => stdexten-BUSY,1,Voicemail(${mbx},b) ; If busy, send to voicemail w/ busy announce
|
|
exten => stdexten-BUSY,n,Return() ; If they press #, return to start
|
|
|
|
exten => _stde[x]te[n]-.,1,Goto(stdexten-NOANSWER,1) ; Treat anything else as no answer
|
|
|
|
exten => a,1,VoicemailMain(${mbx}) ; If they press *, send the user into VoicemailMain
|
|
exten => a,n,Return()
|
|
|
|
[stdPrivacyexten]
|
|
;
|
|
; Standard extension subroutine:
|
|
; ${ARG1} - Extension
|
|
; ${ARG2} - Device(s) to ring
|
|
; ${ARG3} - Optional DONTCALL context name to jump to (assumes the s,1 extension-priority)
|
|
; ${ARG4} - Optional TORTURE context name to jump to (assumes the s,1 extension-priority)`
|
|
; ${ARG5} - Context in voicemail (if empty, then "default")
|
|
;
|
|
; See above note in stdexten about priority handling on exit.
|
|
;
|
|
exten => _X.,60000(stdPrivacyexten),NoOp(Start stdPrivacyexten)
|
|
exten => _X.,n,Set(LOCAL(ext)=${ARG1})
|
|
exten => _X.,n,Set(LOCAL(dev)=${ARG2})
|
|
exten => _X.,n,Set(LOCAL(dontcntx)=${ARG3})
|
|
exten => _X.,n,Set(LOCAL(tortcntx)=${ARG4})
|
|
exten => _X.,n,Set(LOCAL(cntx)=${ARG5})
|
|
|
|
exten => _X.,n,Set(LOCAL(mbx)="${ext}"$["${cntx}" ? "@${cntx}" :: ""])
|
|
exten => _X.,n,Dial(${dev},20,p) ; Ring the interface, 20 seconds maximum, call screening
|
|
; option (or use P for databased call _X.creening)
|
|
exten => _X.,n,Goto(stdexten-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
|
|
|
|
exten => stdexten-NOANSWER,1,Voicemail(${mbx},u) ; If unavailable, send to voicemail w/ unavail announce
|
|
exten => stdexten-NOANSWER,n,NoOp(Finish stdPrivacyexten NOANSWER)
|
|
exten => stdexten-NOANSWER,n,Return() ; If they press #, return to start
|
|
|
|
exten => stdexten-BUSY,1,Voicemail(${mbx},b) ; If busy, send to voicemail w/ busy announce
|
|
exten => stdexten-BUSY,n,NoOp(Finish stdPrivacyexten BUSY)
|
|
exten => stdexten-BUSY,n,Return() ; If they press #, return to start
|
|
|
|
exten => stdexten-DONTCALL,1,Goto(${dontcntx},s,1) ; Callee chose to send this call to a polite "Don't call again" script.
|
|
|
|
exten => stdexten-TORTURE,1,Goto(${tortcntx},s,1) ; Callee chose to send this call to a telemarketer torture script.
|
|
|
|
exten => _stde[x]te[n]-.,1,Goto(stdexten-NOANSWER,1) ; Treat anything else as no answer
|
|
|
|
exten => a,1,VoicemailMain(${mbx}) ; If they press *, send the user into VoicemailMain
|
|
exten => a,n,Return
|
|
|
|
[macro-page];
|
|
;
|
|
; Paging macro:
|
|
;
|
|
; Check to see if SIP device is in use and DO NOT PAGE if they are
|
|
;
|
|
; ${ARG1} - Device to page
|
|
|
|
exten => s,1,ChanIsAvail(${ARG1},s) ; s is for ANY call
|
|
exten => s,n,GoToIf($[${AVAILORIGCHAN} = ""]?fail:autoanswer)
|
|
exten => s,n(autoanswer),Set(_ALERT_INFO="RA") ; This is for the PolyComs
|
|
exten => s,n,SIPAddHeader(Call-Info: Answer-After=0) ; This is for the Grandstream, Snoms, and Others
|
|
exten => s,n,NoOp() ; Add others here and Post on the Wiki!!!!
|
|
exten => s,n,Dial(${ARG1})
|
|
exten => s,n(fail),Hangup
|
|
|
|
|
|
[demo]
|
|
include => stdexten
|
|
;
|
|
; We start with what to do when a call first comes in.
|
|
;
|
|
exten => s,1,Wait(1) ; Wait a second, just for fun
|
|
exten => s,n,Answer ; Answer the line
|
|
exten => s,n,Set(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds
|
|
exten => s,n,Set(TIMEOUT(response)=10) ; Set Response Timeout to 10 seconds
|
|
exten => s,n(restart),BackGround(demo-congrats) ; Play a congratulatory message
|
|
exten => s,n(instruct),BackGround(demo-instruct) ; Play some instructions
|
|
exten => s,n,WaitExten ; Wait for an extension to be dialed.
|
|
|
|
exten => 2,1,BackGround(demo-moreinfo) ; Give some more information.
|
|
exten => 2,n,Goto(s,instruct)
|
|
|
|
exten => 3,1,Set(LANGUAGE()=fr) ; Set language to french
|
|
exten => 3,n,Goto(s,restart) ; Start with the congratulations
|
|
|
|
exten => 1000,1,Goto(default,s,1)
|
|
;
|
|
; We also create an example user, 1234, who is on the console and has
|
|
; voicemail, etc.
|
|
;
|
|
exten => 1234,1,Playback(transfer,skip) ; "Please hold while..."
|
|
; (but skip if channel is not up)
|
|
exten => 1234,n,Gosub(${EXTEN},stdexten(${GLOBAL(CONSOLE)}))
|
|
exten => 1234,n,Goto(default,s,1) ; exited Voicemail
|
|
|
|
exten => 1235,1,Voicemail(1234,u) ; Right to voicemail
|
|
|
|
exten => 1236,1,Dial(Console/dsp) ; Ring forever
|
|
exten => 1236,n,Voicemail(1234,b) ; Unless busy
|
|
|
|
;
|
|
; # for when they're done with the demo
|
|
;
|
|
exten => #,1,Playback(demo-thanks) ; "Thanks for trying the demo"
|
|
exten => #,n,Hangup ; Hang them up.
|
|
|
|
;
|
|
; A timeout and "invalid extension rule"
|
|
;
|
|
exten => t,1,Goto(#,1) ; If they take too long, give up
|
|
exten => i,1,Playback(invalid) ; "That's not valid, try again"
|
|
|
|
;
|
|
; Create an extension, 500, for dialing the
|
|
; Asterisk demo.
|
|
;
|
|
exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
|
|
exten => 500,n,Dial(IAX2/guest@pbx.digium.com/s@default) ; Call the Asterisk demo
|
|
exten => 500,n,Playback(demo-nogo) ; Couldn't connect to the demo site
|
|
exten => 500,n,Goto(s,6) ; Return to the start over message.
|
|
|
|
;
|
|
; Create an extension, 600, for evaluating echo latency.
|
|
;
|
|
exten => 600,1,Playback(demo-echotest) ; Let them know what's going on
|
|
exten => 600,n,Echo ; Do the echo test
|
|
exten => 600,n,Playback(demo-echodone) ; Let them know it's over
|
|
exten => 600,n,Goto(s,6) ; Start over
|
|
|
|
;
|
|
; You can use the Macro Page to intercom a individual user
|
|
exten => 76245,1,Macro(page,SIP/Grandstream1)
|
|
; or if your peernames are the same as extensions
|
|
exten => _7XXX,1,Macro(page,SIP/${EXTEN})
|
|
;
|
|
;
|
|
; System Wide Page at extension 7999
|
|
;
|
|
exten => 7999,1,Set(TIMEOUT(absolute)=60)
|
|
exten => 7999,2,Page(Local/Grandstream1@page&Local/Xlite1@page&Local/1234@page/n,d)
|
|
|
|
; Give voicemail at extension 8500
|
|
;
|
|
exten => 8500,1,VoicemailMain
|
|
exten => 8500,n,Goto(s,6)
|
|
;
|
|
; Here's what a phone entry would look like (IXJ for example)
|
|
;
|
|
;exten => 1265,1,Dial(Phone/phone0,15)
|
|
;exten => 1265,n,Goto(s,5)
|
|
|
|
;
|
|
; The page context calls up the page macro that sets variables needed for auto-answer
|
|
; It is in is own context to make calling it from the Page() application as simple as
|
|
; Local/{peername}@page
|
|
;
|
|
[page]
|
|
exten => _X.,1,Macro(page,SIP/${EXTEN})
|
|
|
|
;[mainmenu]
|
|
;
|
|
; Example "main menu" context with submenu
|
|
;
|
|
;exten => s,1,Answer
|
|
;exten => s,n,Background(thanks) ; "Thanks for calling press 1 for sales, 2 for support, ..."
|
|
;exten => s,n,WaitExten
|
|
;exten => 1,1,Goto(submenu,s,1)
|
|
;exten => 2,1,Hangup
|
|
;include => default
|
|
;
|
|
;[submenu]
|
|
;exten => s,1,Ringing ; Make them comfortable with 2 seconds of ringback
|
|
;exten => s,n,Wait,2
|
|
;exten => s,n,Background(submenuopts) ; "Thanks for calling the sales department. Press 1 for steve, 2 for..."
|
|
;exten => s,n,WaitExten
|
|
;exten => 1,1,Goto(default,steve,1)
|
|
;exten => 2,1,Goto(default,mark,2)
|
|
|
|
[default]
|
|
;
|
|
; By default we include the demo. In a production system, you
|
|
; probably don't want to have the demo there.
|
|
;
|
|
include => demo
|
|
|
|
;
|
|
; An extension like the one below can be used for FWD, Nikotel, sipgate etc.
|
|
; Note that you must have a [sipprovider] section in sip.conf
|
|
;
|
|
;exten => _41X.,1,Dial(SIP/${FILTER(0-9,${EXTEN:2})}@sipprovider,,r)
|
|
|
|
; Real extensions would go here. Generally you want real extensions to be
|
|
; 4 or 5 digits long (although there is no such requirement) and start with a
|
|
; single digit that is fairly large (like 6 or 7) so that you have plenty of
|
|
; room to overlap extensions and menu options without conflict. You can alias
|
|
; them with names, too, and use global variables
|
|
|
|
;exten => 6245,hint,SIP/Grandstream1&SIP/Xlite1(Joe Schmoe) ; Channel hints for presence
|
|
;exten => 6245,1,Dial(SIP/Grandstream1,20,rt) ; permit transfer
|
|
;exten => 6245,n(dial),Dial(${HINT},20,rtT) ; Use hint as listed
|
|
;exten => 6245,n,Voicemail(6245,u) ; Voicemail (unavailable)
|
|
;exten => 6245,s+1,Hangup ; s+1, same as n
|
|
;exten => 6245,dial+101,Voicemail(6245,b) ; Voicemail (busy)
|
|
;exten => 6361,1,Dial(IAX2/JaneDoe,,rm) ; ring without time limit
|
|
;exten => 6389,1,Dial(MGCP/aaln/1@192.168.0.14)
|
|
;exten => 6390,1,Dial(JINGLE/caller/callee) ; Dial via jingle using labels
|
|
;exten => 6391,1,Dial(JINGLE/asterisk@digium.com/mogorman@astjab.org) ;Dial via jingle using asterisk as the transport and calling mogorman.
|
|
;exten => 6394,1,Dial(Local/6275/n) ; this will dial ${MARK}
|
|
|
|
;exten => 6275,1,Gosub(${EXTEN},stdexten(${MARK}))
|
|
; assuming ${MARK} is something like DAHDI/2
|
|
;exten => 6275,n,Goto(default,s,1) ; exited Voicemail
|
|
;exten => mark,1,Goto(6275,1) ; alias mark to 6275
|
|
;exten => 6536,1,Gosub(${EXTEN},stdexten(${WIL}))
|
|
; Ditto for wil
|
|
;exten => 6536,n,Goto(default,s,1) ; exited Voicemail
|
|
;exten => wil,1,Goto(6236,1)
|
|
|
|
;If you want to subscribe to the status of a parking space, this is
|
|
;how you do it. Subscribe to extension 6600 in sip, and you will see
|
|
;the status of the first parking lot with this extensions' help
|
|
;exten => 6600,hint,park:701@parkedcalls
|
|
;exten => 6600,1,noop
|
|
;
|
|
; Some other handy things are an extension for checking voicemail via
|
|
; voicemailmain
|
|
;
|
|
;exten => 8500,1,VoicemailMain
|
|
;exten => 8500,n,Hangup
|
|
;
|
|
; Or a conference room (you'll need to edit meetme.conf to enable this room)
|
|
;
|
|
;exten => 8600,1,Meetme(1234)
|
|
;
|
|
; Or playing an announcement to the called party, as soon it answers
|
|
;
|
|
;exten = 8700,1,Dial(${MARK},30,A(/path/to/my/announcemsg))
|
|
;
|
|
|
|
; example of a compartmentalized company called "acme"
|
|
;
|
|
; this is the context that your incoming IAX/SIP trunk dumps you in...
|
|
;[acme-incoming]
|
|
;exten => s,1,Wait(1)
|
|
;exten => s,n,Answer()
|
|
;exten => s,n(menu),Playback(acme/vm-brief-menu)
|
|
;exten => s,n(exten),Background(vm-enter-num-to-call)
|
|
;exten => s,n,WaitExten(5)
|
|
;exten => s,n(goodbye),Playback(vm-goodbye)
|
|
;exten => s,n(end),Hangup()
|
|
;
|
|
;include => acme-extens
|
|
;
|
|
;exten => i,1,Playback(vm-invalid)
|
|
;exten => i,n,Goto(s,exten) ; optionally, transfer to operator
|
|
;
|
|
;exten => t,1,Goto(s,goodbye)
|
|
;
|
|
; this is the context our internal SIP hardphones use (see sip.conf)
|
|
;
|
|
;[acme-internal]
|
|
;exten => s,1,Answer()
|
|
;exten => s,n(exten),Background(vm-enter-num-to-call)
|
|
;exten => s,n,WaitExten(5)
|
|
;exten => s,n(goodbye),Playback(vm-goodbye)
|
|
;exten => s,n(end),Hangup()
|
|
;
|
|
;include => trunkint
|
|
;include => trunkld
|
|
;include => trunklocal
|
|
;
|
|
;include => acme-extens
|
|
;
|
|
; you can test what your system sounds like to outside callers by dialing this
|
|
;exten => 777,1,DISA(no-password,acme-incoming)
|
|
;
|
|
; grouping of acme's extensions... never used directly, always included.
|
|
;
|
|
;[acme-extens]
|
|
;include => stdexten
|
|
;exten => 111,1,Gosub(111,stdexten(SIP/pete_1,acme))
|
|
;exten => 111,n,Goto(s,exten)
|
|
;
|
|
;exten => 112,1,Gosub(112,stdexten(SIP/nancy_1,acme))
|
|
;exten => 112,n,Goto(s,end)
|
|
;
|
|
; end of acme example
|
|
|
|
;
|
|
; Time context: you can patch this in via the following.
|
|
;
|
|
; [acme-internal]
|
|
; ...
|
|
; exten => 777,1,Gosub(time)
|
|
; exten => 777,n,Hangup()
|
|
;
|
|
; ...
|
|
; include => time
|
|
;
|
|
; Note: if you're geographically spread out, you can have SIP extensions
|
|
; specify their own local timezone in sip.conf as:
|
|
;
|
|
; [boi]
|
|
; type=friend
|
|
; context=acme-internal
|
|
; callerid="Boise Ofc. <2083451111>"
|
|
; ...
|
|
; ; use system-wide default timezone of MST7MDT
|
|
;
|
|
; [lws]
|
|
; type=friend
|
|
; context=acme-internal
|
|
; callerid="Lewiston Ofc. <2087431111>"
|
|
; ...
|
|
; setvar=timezone=PST8PDT
|
|
;
|
|
; "timezone" isn't a 'reserved' name in any way, and other places where
|
|
; the timezone is significant (e.g. calls to "SayUnixTime()", etc) will
|
|
; require modification as well. Note that voicemail.conf already has
|
|
; a mechanism for timezones.
|
|
;
|
|
|
|
[time]
|
|
exten => _X.,30000(time),NoOp(Time: ${EXTEN} ${timezone})
|
|
exten => _X.,n,Wait(0.25)
|
|
exten => _X.,n,Answer()
|
|
; the amount of delay is set for English; you may need to adjust this time
|
|
; for other languages if there's no pause before the synchronizing beep.
|
|
exten => _X.,n,Set(FUTURETIME=$[${EPOCH} + 12])
|
|
exten => _X.,n,SayUnixTime(${FUTURETIME},Zulu,HNS)
|
|
exten => _X.,n,SayPhonetic(z)
|
|
; use the timezone associated with the extension (sip only), or system-wide
|
|
; default if one hasn't been set.
|
|
exten => _X.,n,SayUnixTime(${FUTURETIME},${timezone},HNS)
|
|
exten => _X.,n,Playback(spy-local)
|
|
exten => _X.,n,WaitUntil(${FUTURETIME})
|
|
exten => _X.,n,Playback(beep)
|
|
exten => _X.,n,Return()
|
|
|
|
;
|
|
; ANI context: use in the same way as "time" above
|
|
;
|
|
|
|
[ani]
|
|
exten => _X.,40000(ani),NoOp(ANI: ${EXTEN})
|
|
exten => _X.,n,Wait(0.25)
|
|
exten => _X.,n,Answer()
|
|
exten => _X.,n,Playback(vm-from)
|
|
exten => _X.,n,SayDigits(${CALLERID(ani)})
|
|
exten => _X.,n,Wait(1.25)
|
|
exten => _X.,n,SayDigits(${CALLERID(ani)}) ; playback again in case of missed digit
|
|
exten => _X.,n,Return()
|
|
|
|
; For more information on applications, just type "core show applications" at your
|
|
; friendly Asterisk CLI prompt.
|
|
;
|
|
; "core show application <command>" will show details of how you
|
|
; use that particular application in this file, the dial plan.
|
|
; "core show functions" will list all dialplan functions
|
|
; "core show function <COMMAND>" will show you more information about
|
|
; one function. Remember that function names are UPPER CASE.
|