From f38742190538c10fbcf866b7169826165488b0c6 Mon Sep 17 00:00:00 2001
From: "Kevin P. Fleming" <kpfleming@digium.com>
Date: Tue, 4 Oct 2005 22:51:59 +0000
Subject: [PATCH] make sample config files easier to ready (issue #5371)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6720 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
configs/alarmreceiver.conf.sample | 36 +++--
configs/codecs.conf.sample | 3 +-
configs/extensions.conf.sample | 32 ++--
configs/iax.conf.sample | 93 ++++++------
configs/iaxprov.conf.sample | 39 +++--
configs/indications.conf.sample | 18 ++-
configs/logger.conf.sample | 27 ++--
configs/manager.conf.sample | 20 +--
configs/meetme.conf.sample | 4 +-
configs/mgcp.conf.sample | 3 +-
configs/modules.conf.sample | 11 +-
configs/musiconhold.conf.sample | 3 +-
configs/queues.conf.sample | 42 ++---
configs/sip.conf.sample | 51 ++++---
configs/voicemail.conf.sample | 47 +++---
configs/vpb.conf.sample | 31 ++--
configs/zapata.conf.sample | 244 +++++++++++++++---------------
17 files changed, 373 insertions(+), 331 deletions(-)
diff --git a/configs/alarmreceiver.conf.sample b/configs/alarmreceiver.conf.sample
index 0c97a86ef8..bf767dea3e 100755
--- a/configs/alarmreceiver.conf.sample
+++ b/configs/alarmreceiver.conf.sample
@@ -22,8 +22,9 @@ timestampformat = %a %b %d, %Y @ %H:%M:%S %Z
;eventcmd = yourprogram -yourargs ...
;
-; Specify a spool directory for the event files. This setting is required if you want the app to be useful.
-; Event files written to the spool directory will be of the template event-XXXXXX, where XXXXXX is a random
+; Specify a spool directory for the event files. This setting is required
+; if you want the app to be useful. Event files written to the spool
+; directory will be of the template event-XXXXXX, where XXXXXX is a random
; and unique alphanumeric string.
;
; Default is none, and the events will be dropped on the floor.
@@ -32,8 +33,9 @@ timestampformat = %a %b %d, %Y @ %H:%M:%S %Z
eventspooldir = /tmp
;
-; The alarmreceiver app can either log the events one-at-a-time to individual files in the spool
-; directory, or it can store them until the caller disconnects and write them all to one file.
+; The alarmreceiver app can either log the events one-at-a-time to individual
+; files in the spool directory, or it can store them until the caller
+; disconnects and write them all to one file.
;
; The default setting for logindividualevents is no.
;
@@ -41,32 +43,34 @@ eventspooldir = /tmp
logindividualevents = no
;
-; The timeout for receiving the first DTMF digit is adjustable from 1000 msec. to 10000 msec. The
-; default is 2000 msec. Note: if you wish to test the receiver by entering digits manually, set this
-; to a reasonable time out like 10000 milliseconds.
+; The timeout for receiving the first DTMF digit is adjustable from 1000 msec.
+; to 10000 msec. The default is 2000 msec. Note: if you wish to test the
+; receiver by entering digits manually, set this to a reasonable time out
+; like 10000 milliseconds.
fdtimeout = 2000
;
-; The timeout for receiving subsequent DTMF digits is adjustable from 110 msec. to 4000 msec. The
-; default is 200 msec. Note: if you wish to test the receiver by entering digits manually, set this
-; to a reasonable time out like 4000 milliseconds.
+; The timeout for receiving subsequent DTMF digits is adjustable from
+; 110 msec. to 4000 msec. The default is 200 msec. Note: if you wish to test
+; the receiver by entering digits manually, set this to a reasonable time out
+; like 4000 milliseconds.
;
sdtimeout = 200
;
-; The loudness of the ACK and Kissoff tones is adjustable from 100 to 8192. The default is 8192
-; This shouldn't need to be messed with, but is included just in case there are problems with
-; signal levels.
+; The loudness of the ACK and Kissoff tones is adjustable from 100 to 8192.
+; The default is 8192. This shouldn't need to be messed with, but is included
+; just in case there are problems with signal levels.
;
loudness = 8192
;
-; The db-family setting allows the user to capture statistics on the number of calls, and the errors
-; the alarm receiver sees. The default is for no db-family name to be defined and the database logging
-; to be turned off.
+; The db-family setting allows the user to capture statistics on the number of
+; calls, and the errors the alarm receiver sees. The default is for no
+; db-family name to be defined and the database logging to be turned off.
;
;db-family = yourfamily:
diff --git a/configs/codecs.conf.sample b/configs/codecs.conf.sample
index 6918b29075..c8caeab603 100755
--- a/configs/codecs.conf.sample
+++ b/configs/codecs.conf.sample
@@ -12,7 +12,8 @@ complexity => 2
enhancement => true
; voice activity detection [true / false]
-; reduces bitrate when no voice detected, used only for CBR (implicit in VBR/ABR)
+; reduces bitrate when no voice detected, used only for CBR
+; (implicit in VBR/ABR)
vad => true
; variable bit rate [true / false]
diff --git a/configs/extensions.conf.sample b/configs/extensions.conf.sample
index e845974813..d773cbbc3e 100755
--- a/configs/extensions.conf.sample
+++ b/configs/extensions.conf.sample
@@ -52,13 +52,15 @@ clearglobalvars=no
;
priorityjumping=no
;
-; You can include other config files, use the #include command (without the ';')
-; Note that this is different from the "include" command that includes contexts within
-; other contexts. The #include command works in all asterisk configuration files.
+; You can include other config files, use the #include command
+; (without the ';'). Note that this is different from the "include" command
+; that includes contexts within other contexts. The #include command works
+; in all asterisk configuration files.
;#include "filename.conf"
; The "Globals" category contains global variables that can be referenced
-; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental variable
+; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental
+; variables,
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
@@ -73,10 +75,14 @@ TRUNK=Zap/g2 ; Trunk interface
; in zapata.conf) to dial, i.e. group 2, and how to choose a channel to use in
; the specified group. The four possible options are:
;
-; g: select the lowest-numbered non-busy Zap channel (aka. ascending sequential hunt group).
-; G: select the highest-numbered non-busy Zap channel (aka. descending sequential hunt group).
-; r: use a round-robin search, starting at the next highest channel than last time (aka. ascending rotary hunt group).
-; R: use a round-robin search, starting at the next lowest channel than last time (aka. descending rotary hunt group).
+; g: select the lowest-numbered non-busy Zap channel
+; (aka. ascending sequential hunt group).
+; G: select the highest-numbered non-busy Zap channel
+; (aka. descending sequential hunt group).
+; r: use a round-robin search, starting at the next highest channel than last
+; time (aka. ascending rotary hunt group).
+; R: use a round-robin search, starting at the next lowest channel than last
+; time (aka. descending rotary hunt group).
;
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
;TRUNK=IAX2/user:pass@provider
@@ -443,11 +449,11 @@ include => demo
;exten => _41X.,1,Dial(SIP/${EXTEN:2}@sipprovider,,r)
;exten => _42X.,1,Dial(SIP/user:passwd@${EXTEN:2}@otherprovider.net,30,rT)
-; 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
+; 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
diff --git a/configs/iax.conf.sample b/configs/iax.conf.sample
index 0ebcdc83a1..3fb5d338a7 100755
--- a/configs/iax.conf.sample
+++ b/configs/iax.conf.sample
@@ -74,11 +74,11 @@ disallow=lpc10 ; Icky sound quality... Mr. Roboto.
; The jitter buffer's function is to compensate for varying
; network delay.
;
-; There are presently two jitterbuffer implementations available for * and chan_iax2;
-; the classic and the new, channel/application independent implementation. These
-; are controlled at compile-time. The new jitterbuffer additionally has support for PLC
-; which greatly improves quality as the jitterbuffer adapts size, and in compensating for lost
-; packets.
+; There are presently two jitterbuffer implementations available for Asterisk
+; and chan_iax2; the classic and the new, channel/application independent
+; implementation. These are controlled at compile-time. The new jitterbuffer
+; additionally has support for PLC which greatly improves quality as the
+; jitterbuffer adapts size, and in compensating for lost packets.
;
; All the jitter buffer settings except dropcount are in milliseconds.
; The jitter buffer works for INCOMING audio - the outbound audio
@@ -90,7 +90,8 @@ disallow=lpc10 ; Icky sound quality... Mr. Roboto.
; forcejitterbuffer=yes|no: in the ideal world, when we bridge VoIP channels
; we don't want to do jitterbuffering on the switch, since the endpoints
; can each handle this. However, some endpoints may have poor jitterbuffers
-; themselves, so this option will force * to always jitterbuffer, even in this case.
+; themselves, so this option will force * to always jitterbuffer, even in this
+; case.
; [This option presently applies only to the new jitterbuffer implementation]
;
; dropcount: the jitter buffer is sized such that no more than "dropcount"
@@ -105,15 +106,17 @@ disallow=lpc10 ; Icky sound quality... Mr. Roboto.
;
; resyncthreshold: when the jitterbuffer notices a significant change in delay
; that continues over a few frames, it will resync, assuming that the change in
-; delay was caused by a timestamping mix-up. The threshold for noticing a change
-; in delay is measured as twice the measured jitter plus this resync threshold.
-; Resycning can be disabled by setting this parameter to -1.
+; delay was caused by a timestamping mix-up. The threshold for noticing a
+; change in delay is measured as twice the measured jitter plus this resync
+; threshold.
+; Resyncing can be disabled by setting this parameter to -1.
; [This option presently applies only to the new jitterbuffer implementation]
;
-; maxjitterinterps: the maximum number of interpolation frames the jitterbuffer should
-; return in a row. Since some clients do not send CNG/DTX frames to indicate
-; silence, the jitterbuffer will assume silence has begun after returning this
-; many interpolations. This prevents interpolating throughout a long silence.
+; maxjitterinterps: the maximum number of interpolation frames the jitterbuffer
+; should return in a row. Since some clients do not send CNG/DTX frames to
+; indicate silence, the jitterbuffer will assume silence has begun after
+; returning this many interpolations. This prevents interpolating throughout
+; a long silence.
; [This option presently applies only to the new jitterbuffer implementation]
;
; maxexcessbuffer: If conditions improve after a period of high jitter,
@@ -147,11 +150,11 @@ forcejitterbuffer=no
;trunkfreq=20 ; How frequently to send trunk msgs (in ms)
; Should we send timestamps for the individual sub-frames within trunk frames?
-; There is a small bandwidth use for these (less than 1kbps/call), but they ensure
-; that frame timestamps get sent end-to-end properly. If both ends of all your trunks
-; go directly to TDM, _and_ your trunkfreq equals the frame length for your codecs, you
-; can probably suppress these. The receiver must also support this feature, although
-; they do not also need to have it enabled.
+; There is a small bandwidth use for these (less than 1kbps/call), but they
+; ensure that frame timestamps get sent end-to-end properly. If both ends of
+; all your trunks go directly to TDM, _and_ your trunkfreq equals the frame
+; length for your codecs, you can probably suppress these. The receiver must
+; also support this feature, although they do not also need to have it enabled.
;
; trunktimestamps=yes
;
@@ -217,22 +220,21 @@ tos=lowdelay
;
;mailboxdetail=yes
;
-; If regcontext is specified, Asterisk will dynamically
-; create and destroy a NoOp priority 1 extension for a given
-; peer who registers or unregisters with us. The actual extension
-; is the 'regexten' parameter of the registering peer or its
-; name if 'regexten' is not provided. More than one regexten may be supplied
-; if they are separated by '&'. Patterns may be used in regexten.
+; If regcontext is specified, Asterisk will dynamically create and destroy
+; a NoOp priority 1 extension for a given peer who registers or unregisters
+; with us. The actual extension is the 'regexten' parameter of the registering
+; peer or its name if 'regexten' is not provided. More than one regexten
+; may be supplied if they are separated by '&'. Patterns may be used in
+; regexten.
;
;regcontext=iaxregistrations
;
-; If we don't get ACK to our NEW within 2000ms, and autokill is set
-; to yes, then we cancel the whole thing (that's enough time for one
-; retransmission only). This is used to keep things from stalling for a long
-; time for a host that is not available, but would be ill advised for bad
-; connections. In addition to 'yes' or 'no' you can also specify a number
-; of milliseconds. See 'qualify' for individual peers to turn on for just
-; a specific peer.
+; If we don't get ACK to our NEW within 2000ms, and autokill is set to yes,
+; then we cancel the whole thing (that's enough time for one retransmission
+; only). This is used to keep things from stalling for a long time for a host
+; that is not available, but would be ill advised for bad connections. In
+; addition to 'yes' or 'no' you can also specify a number of milliseconds.
+; See 'qualify' for individual peers to turn on for just a specific peer.
;
autokill=yes
;
@@ -274,8 +276,8 @@ autokill=yes
; has expired based on its registration interval, used the stored
; address information regardless. (yes|no)
-; Guest sections for unauthenticated connection attempts. Just
-; specify an empty secret, or provide no secret section.
+; Guest sections for unauthenticated connection attempts. Just specify an
+; empty secret, or provide no secret section.
;
[guest]
type=user
@@ -310,14 +312,13 @@ inkeys=freeworlddialup
;context=dundi-e164-local
;
-; Further user sections may be added, specifying a context and a
-; secret used for connections with that given authentication name.
-; Limited IP based access control is allowed by use of "allow" and
-; "deny" keywords. Multiple rules are permitted. Multiple permitted
-; contexts may be specified, in which case the first will be the default.
-; You can also override caller*ID so that when you receive a call you
-; set the Caller*ID to be what you want instead of trusting what
-; the remote user provides
+; Further user sections may be added, specifying a context and a secret used
+; for connections with that given authentication name. Limited IP based
+; access control is allowed by use of "allow" and "deny" keywords. Multiple
+; rules are permitted. Multiple permitted contexts may be specified, in
+; which case the first will be the default. You can also override caller*ID
+; so that when you receive a call you set the Caller*ID to be what you want
+; instead of trusting what the remote user provides
;
; There are three authentication methods that are supported: md5, plaintext,
; and rsa. The least secure is "plaintext", which sends passwords cleartext
@@ -372,11 +373,10 @@ host=216.207.245.47
;jitterbuffer=no ; Turn off jitter buffer for this peer
;
-; Peers can remotely register as well, so that they can be
-; mobile. Default IP's can also optionally be given but
-; are not required. Caller*ID can be suggested to the other
-; side as well if it is for example a phone instead of another
-; PBX.
+; Peers can remotely register as well, so that they can be mobile. Default
+; IP's can also optionally be given but are not required. Caller*ID can be
+; suggested to the other side as well if it is for example a phone instead of
+; another PBX.
;
;[dynamichost]
@@ -410,3 +410,4 @@ host=216.207.245.47
;secret=moofoo
;context=default
;permit=0.0.0.0/0.0.0.0
+
diff --git a/configs/iaxprov.conf.sample b/configs/iaxprov.conf.sample
index f39db1834a..ad13166ed0 100755
--- a/configs/iaxprov.conf.sample
+++ b/configs/iaxprov.conf.sample
@@ -1,25 +1,22 @@
;
; IAX2 Provisioning Information
;
-; Contains provisioning information for templates
-; and for specific service entries.
+; Contains provisioning information for templates and for specific service
+; entries.
;
-; Templates provide a group of settings from which provisioning takes
-; place. A template may be based upon any template that has been
-; specified before it. If the template that an entry is based on is not
-; specified then it is presumed to be 'default' (unless it is the first
-; of course).
+; Templates provide a group of settings from which provisioning takes place.
+; A template may be based upon any template that has been specified before
+; it. If the template that an entry is based on is not specified then it is
+; presumed to be 'default' (unless it is the first of course).
;
-; Templates which begin with 'si-' are used for provisioning
-; units with specific service identifiers. For example the
-; entry "si-000364000126" would be used when the device with the
-; corresponding service identifier of "000364000126" attempts
-; to register or make a call.
+; Templates which begin with 'si-' are used for provisioning units with
+; specific service identifiers. For example the entry "si-000364000126"
+; would be used when the device with the corresponding service identifier of
+; "000364000126" attempts to register or make a call.
;
[default]
;
-; The port number the device should use to bind to. The default
-; is 4569
+; The port number the device should use to bind to. The default is 4569.
;
;port=4569
;
@@ -27,14 +24,13 @@
;
;server=192.168.69.3
;
-; altserver is the BACKUP server for registration and placing calls
-; in the event the primary server is unavailable.
+; altserver is the BACKUP server for registration and placing calls in the
+; event the primary server is unavailable.
;
;altserver=192.168.69.4
;
-; port is the port number to use for IAX2 outbound. The
-; connections to the server and altserver -- default is of course
-; 4569.
+; port is the port number to use for IAX2 outbound. The connections to the
+; server and altserver -- default is of course 4569.
;serverport=4569
;
; language is the preferred language for the device
@@ -78,9 +74,10 @@ tos=lowdelay
;
;[*]
;
-; If specified, the '*' provisioning is used for all devices which do
-; not have another provisioning entry within the file. If unspecified, no
+; If specified, the '*' provisioning is used for all devices which do not
+; have another provisioning entry within the file. If unspecified, no
; provisioning will take place for devices which have no entry. DO NOT
; USE A '*' PROVISIONING ENTRY UNLESS YOU KNOW WHAT YOU'RE DOING.
;
;template=default
+
diff --git a/configs/indications.conf.sample b/configs/indications.conf.sample
index e1e1e88fff..4ea4d0bdd0 100755
--- a/configs/indications.conf.sample
+++ b/configs/indications.conf.sample
@@ -16,7 +16,7 @@ country=us ; default location
; [example]
; description = string
-; The full name of your country, in English
+; The full name of your country, in English.
; alias = iso[,iso]*
; List of other countries 2-letter iso codes, which have the same
; tone indications.
@@ -31,14 +31,16 @@ country=us ; default location
; callwaiting = tonelist
; Set of tones played when there is a call waiting in the background.
; dialrecall = tonelist
-; Not well defined, many phone systems play a recall dial tone after hook flash
+; Not well defined; many phone systems play a recall dial tone after hook
+; flash.
; record = tonelist
-; Set of tones played when call recording is in progress
+; Set of tones played when call recording is in progress.
; info = tonelist
-; Set of tones played with special information messages (e.g., "number is out of service")
+; Set of tones played with special information messages (e.g., "number is
+; out of service")
; 'name' = tonelist
-; Every other variable will be available as a shortcut for the "PlayList" command
-; but will not automaticly be used by Asterisk.
+; Every other variable will be available as a shortcut for the "PlayList" command
+; but will not be used automatically by Asterisk.
;
;
; The tonelist itself is defined by a comma-separated sequence of elements.
@@ -587,8 +589,8 @@ stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/1
description = South Africa
; http://www.cisco.com/univercd/cc/td/doc/product/tel_pswt/vco_prod/safr_sup/saf02.htm
; (definitions for other countries can also be found there)
-; Note, though, that South Africa uses two switch types in their network - Alcatel
-; switches - mainly in the Western Cape, and Siemens elsewhere.
+; Note, though, that South Africa uses two switch types in their network --
+; Alcatel switches -- mainly in the Western Cape, and Siemens elsewhere.
; The former use 383+417 in dial, ringback etc. The latter use 400*33
; I've provided both, uncomment the ones you prefer
ringcadance = 400,200,400,2000
diff --git a/configs/logger.conf.sample b/configs/logger.conf.sample
index 008554271a..f2ff0ea7eb 100755
--- a/configs/logger.conf.sample
+++ b/configs/logger.conf.sample
@@ -16,10 +16,12 @@
; This appends the hostname to the name of the log files.
;appendhostname = yes
;
-; This determines whether or not we log queue events to a file (defaults to yes).
+; This determines whether or not we log queue events to a file
+; (defaults to yes).
;queue_log = no
;
-; This determines whether or not we log generic events to a file (defaults to yes).
+; This determines whether or not we log generic events to a file
+; (defaults to yes).
;event_log = no
;
;
@@ -44,17 +46,16 @@
;
; Special filename "console" represents the system console
;
-; We highly recommend that you DO NOT turn on debug mode if you
-; are simply running a production system. Debug mode turns on a
-; LOT of extra messages, most of which you are unlikely to understand
-; without an understanding of the underlying code. Do NOT report
-; debug messages as code issues, unless you have a specific issue that
-; you are attempting to debug. They are messages for just that --
-; debugging -- and do not rise to the level of something that merit
-; your attention as an Asterisk administrator. Debug messages are also
-; very verbose and can and do fill up logfiles quickly; this is another
-; reason not to have debug mode on a production system unless you are
-; in the process of debugging a specific issue.
+; We highly recommend that you DO NOT turn on debug mode if you are simply
+; running a production system. Debug mode turns on a LOT of extra messages,
+; most of which you are unlikely to understand without an understanding of
+; the underlying code. Do NOT report debug messages as code issues, unless
+; you have a specific issue that you are attempting to debug. They are
+; messages for just that -- debugging -- and do not rise to the level of
+; something that merit your attention as an Asterisk administrator. Debug
+; messages are also very verbose and can and do fill up logfiles quickly;
+; this is another reason not to have debug mode on a production system unless
+; you are in the process of debugging a specific issue.
;
;debug => debug
console => notice,warning,error
diff --git a/configs/manager.conf.sample b/configs/manager.conf.sample
index 4141aa416b..ff37f8a1bc 100755
--- a/configs/manager.conf.sample
+++ b/configs/manager.conf.sample
@@ -1,23 +1,19 @@
;
; AMI - The Asterisk Manager Interface
;
-; Third party application call management support
-; and PBX event supervision
+; Third party application call management support and PBX event supervision
;
-; This configuration file is read every time someone
-; logs in
+; This configuration file is read every time someone logs in
;
-; Use the "show manager commands" at the CLI to list
-; availabale manager commands and their authorization
-; levels.
+; Use the "show manager commands" at the CLI to list available manager commands
+; and their authorization levels.
;
; "show manager command <command>" will show a help text.
;
-; ------------------- SECURITY NOTE -----------------
-; Note that you should not enable the AMI on a public
-; IP address. If needed, block this TCP port with
-; iptables (or another FW software) and reach it
-; with IPsec, SSH or SSL vpn tunnel
+; ---------------------------- SECURITY NOTE -------------------------------
+; Note that you should not enable the AMI on a public IP address. If needed,
+; block this TCP port with iptables (or another FW software) and reach it
+; with IPsec, SSH, or SSL vpn tunnel
;
[general]
enabled = no
diff --git a/configs/meetme.conf.sample b/configs/meetme.conf.sample
index 8a26c5464a..b47ed0f2cd 100755
--- a/configs/meetme.conf.sample
+++ b/configs/meetme.conf.sample
@@ -1,6 +1,5 @@
;
-; Configuration file for MeetMe simple conference rooms
-; for Asterisk of course.
+; Configuration file for MeetMe simple conference rooms for Asterisk of course.
;
; This configuration file is read every time you call app meetme()
;
@@ -10,3 +9,4 @@
;
;conf => 1234
;conf => 2345,9938
+
diff --git a/configs/mgcp.conf.sample b/configs/mgcp.conf.sample
index f9ffc01d86..cf7b2c9160 100755
--- a/configs/mgcp.conf.sample
+++ b/configs/mgcp.conf.sample
@@ -45,7 +45,8 @@
;
;context=local
;host=dynamic
-;dtmfmode=none ; DTMF Mode can be 'none', 'rfc2833', or 'inband' or 'hybrid' which starts in none and moves to inband. Default is none.
+;dtmfmode=none ; DTMF Mode can be 'none', 'rfc2833', or 'inband' or
+ ; 'hybrid' which starts in none and moves to inband. Default is none.
;slowsequence=yes ; The DPH100M does not follow MGCP standards for sequencing
;line => aaln/1
diff --git a/configs/modules.conf.sample b/configs/modules.conf.sample
index 7162b72da3..f4e08dc1e2 100755
--- a/configs/modules.conf.sample
+++ b/configs/modules.conf.sample
@@ -7,11 +7,12 @@
[modules]
autoload=yes
;
-; Any modules that need to be loaded before the Asterisk core has been initialized
-; (just after the logger has been initialized) can be loaded using 'preload'. This
-; will frequently be needed if you wish to map all module configuration files into
-; Realtime storage, since the Realtime driver will need to be loaded before the
-; modules using those configuration files are initialized.
+; Any modules that need to be loaded before the Asterisk core has been
+; initialized (just after the logger has been initialized) can be loaded
+; using 'preload'. This will frequently be needed if you wish to map all
+; module configuration files into Realtime storage, since the Realtime
+; driver will need to be loaded before the modules using those configuration
+; files are initialized.
;
; An example of loading ODBC support would be:
;preload => res_odbc.so
diff --git a/configs/musiconhold.conf.sample b/configs/musiconhold.conf.sample
index f17501ea2d..6b3e7b6944 100755
--- a/configs/musiconhold.conf.sample
+++ b/configs/musiconhold.conf.sample
@@ -26,7 +26,8 @@ directory=/var/lib/asterisk/mohmp3
;application=/usr/bin/streamplayer 192.168.100.52 888
;format=ulaw
-; mpg123 on Solaris does not always exit properly; madplay may be a better choice
+; mpg123 on Solaris does not always exit properly; madplay may be a better
+; choice
;[solaris]
;mode=custom
;directory=/var/lib/asterisk/mohmp3
diff --git a/configs/queues.conf.sample b/configs/queues.conf.sample
index 2acb6c5bc6..ba7a082b53 100755
--- a/configs/queues.conf.sample
+++ b/configs/queues.conf.sample
@@ -9,8 +9,8 @@
;
persistentmembers = yes
;
-; Note that a timeout to fail out of a queue may be passed as part of application call
-; from extensions.conf:
+; Note that a timeout to fail out of a queue may be passed as part of
+; an application call from extensions.conf:
; Queue(queuename|[options]|[optionalurl]|[announceoverride]|[timeout])
; example: Queue(dave|t|||45)
@@ -43,7 +43,8 @@ persistentmembers = yes
;strategy = ringall
;
; Second settings for service level (default 0)
-; Used for service level statistics (calls answered within service level time frame)
+; Used for service level statistics (calls answered within service level time
+; frame)
;servicelevel = 60
;
; A context may be specified, in which if the user types a SINGLE
@@ -94,7 +95,8 @@ persistentmembers = yes
;
; What's the rounding time for the seconds?
-; If this is non zero then we announce the seconds as well as the minutes rounded to this value
+; If this is non-zero, then we announce the seconds as well as the minutes
+; rounded to this value.
;
; announce-round-seconds = 10
;
@@ -119,26 +121,29 @@ persistentmembers = yes
; To enable monitoring, simply specify "monitor-format"; it will be disabled
; otherwise.
;
-; You can specify the monitor filename with by calling Set(MONITOR_FILENAME=foo)
-; Otherwise it will use ${UNIQUEID}
+; You can specify the monitor filename with by calling
+; Set(MONITOR_FILENAME=foo)
+; Otherwise it will use MONITOR_FILENAME=${UNIQUEID}
;
; monitor-format = gsm|wav|wav49
;
-; If you wish to have the two files joined together when the call ends set this to yes
+; If you wish to have the two files joined together when the call ends, set this
+; to yes.
;
; monitor-join = yes
;
-; This setting controls whether callers can join a queue with no members. There are three
-; choices:
+; This setting controls whether callers can join a queue with no members. There
+; are three choices:
;
-; yes - callers can join a queue with no members or only unavailable members
-; no - callers cannot join a queue with no members
-; strict - callers cannot join a queue with no members or only unavailable members
+; yes - callers can join a queue with no members or only unavailable members
+; no - callers cannot join a queue with no members
+; strict - callers cannot join a queue with no members or only unavailable
+; members
;
; joinempty = yes
;
-; If you wish to remove callers from the queue when new callers cannot join, set this setting
-; to one of the same choices for 'joinempty'
+; If you wish to remove callers from the queue when new callers cannot join,
+; set this setting to one of the same choices for 'joinempty'
;
; leavewhenempty = yes
;
@@ -155,14 +160,15 @@ persistentmembers = yes
;
; eventmemberstatusoff = no
;
-; If you wish to report the caller's hold time to the member before they are connected
-; to the caller, set this to yes.
+; If you wish to report the caller's hold time to the member before they are
+; connected to the caller, set this to yes.
;
; reportholdtime = no
;
;
-; If you wish to have a delay before the member is connected to the caller (or before the member
-; hears any announcement messages), set this to the number of seconds to delay.
+; If you wish to have a delay before the member is connected to the caller (or
+; before the member hears any announcement messages), set this to the number of
+; seconds to delay.
;
; memberdelay = 0
;
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 74b6161ee3..c1ad195f95 100755
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -108,12 +108,11 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;notifyringing = yes ; Notify subscriptions on RINGING state
;
-; If regcontext is specified, Asterisk will dynamically
-; create and destroy a NoOp priority 1 extension for a given
-; peer who registers or unregisters with us. The actual extension
-; is the 'regexten' parameter of the registering peer or its
-; name if 'regexten' is not provided. More than one regexten may be supplied
-; if they are separated by '&'. Patterns may be used in regexten.
+; If regcontext is specified, Asterisk will dynamically create and destroy a
+; NoOp priority 1 extension for a given peer who registers or unregisters with
+; us. The actual extension is the 'regexten' parameter of the registering
+; peer or its name if 'regexten' is not provided. More than one regexten may
+; be supplied if they are separated by '&'. Patterns may be used in regexten.
;
;regcontext=sipregistrations
;
@@ -121,12 +120,12 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Format for the register statement is:
; register => user[:secret[:authuser]]@host[:port][/extension]
;
-; If no extension is given, the 's' extension is used. The extension
-; needs to be defined in extensions.conf to be able to accept calls
-; from this SIP proxy (provider)
+; If no extension is given, the 's' extension is used. The extension needs to
+; be defined in extensions.conf to be able to accept calls from this SIP proxy
+; (provider).
;
-; host is either a host name defined in DNS or the name of a
-; section defined below.
+; host is either a host name defined in DNS or the name of a section defined
+; below.
;
; Examples:
;
@@ -137,12 +136,13 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;
;register => 2345:password@sip_proxy/1234
;
-; Register 2345 at sip provider 'sip_proxy'. Calls from this provider connect to local
-; extension 1234 in extensions.conf default context, unless you define
-; unless you configure a [sip_proxy] section below, and configure a context.
-; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
-; Tip 2: Use separate type=peer and type=user sections for SIP providers
-; (instead of type=friend) if you have calls in both directions
+; Register 2345 at sip provider 'sip_proxy'. Calls from this provider
+; connect to local extension 1234 in extensions.conf, default context,
+; unless you configure a [sip_proxy] section below, and configure a
+; context.
+; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
+; Tip 2: Use separate type=peer and type=user sections for SIP providers
+; (instead of type=friend) if you have calls in both directions
;registertimeout=20 ; retry registration calls every 20 seconds (default)
;registerattempts=10 ; Number of registration attempts before we give up
@@ -151,9 +151,9 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Default is 10 tries
;callevents=no ; generate manager events when sip ua performs events (e.g. hold)
-;---------------------------------------------- NAT SUPPORT ------------------------
-; The externip, externhost and localnet settings are used if you use Asterisk behind
-; a NAT device to communicate with services on the outside.
+;----------------------------------------- NAT SUPPORT ------------------------
+; The externip, externhost and localnet settings are used if you use Asterisk
+; behind a NAT device to communicate with services on the outside.
;externip = 200.201.202.203 ; Address that we're going to put in outbound SIP messages
; if we're behind a NAT
@@ -176,10 +176,10 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
; The nat= setting is used when Asterisk is on a public IP, communicating with
-; devices hidden behind a NAT device (broadband router).
-; If you have one-way audio problems, you usually have problems with your NAT
-; configuration or your firewalls support of SIP+RTP ports.
-; You configure Asterisk choice of RTP ports for incoming audio in rtp.conf
+; devices hidden behind a NAT device (broadband router). If you have one-way
+; audio problems, you usually have problems with your NAT configuration or your
+; firewall's support of SIP+RTP ports. You configure Asterisk choice of RTP
+; ports for incoming audio in rtp.conf
;
;nat=no ; Global NAT settings (Affects all peers and users)
; yes = Always ignore info and assume NAT
@@ -242,7 +242,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; You may also add auth= statements to [peer] definitions
; Peer auth= override all other authentication settings if we match on realm
-;-----------------------------------------------------------------------------------
+;------------------------------------------------------------------------------
; Users and peers have different settings available. Friends have all settings,
; since a friend is both a peer and a user
;
@@ -341,6 +341,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;allow=alaw
;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
;allow=g729 ; Pass-thru only unless g729 license obtained
+;astdb=chan2ext/SIP/grandstream1=1234 ; ensures an astDB entry exists
;[xlite1]
diff --git a/configs/voicemail.conf.sample b/configs/voicemail.conf.sample
index 8680190c00..696ce48ebc 100755
--- a/configs/voicemail.conf.sample
+++ b/configs/voicemail.conf.sample
@@ -10,8 +10,8 @@ serveremail=asterisk
;serveremail=asterisk@linux-support.net
; Should the email contain the voicemail as an attachment
attach=yes
-; Maximum number of messages per folder. If not specified a default value (100) is used.
-; Maximum value for this option is 9999.
+; Maximum number of messages per folder. If not specified, a default value
+; (100) is used. Maximum value for this option is 9999.
;maxmsg=100
; Maximum length of a voicemail message in seconds
;maxmessage=180
@@ -28,13 +28,12 @@ maxsilence=10
silencethreshold=128
; Max number of failed login attempts
maxlogins=3
-; If you need to have an external program, i.e. /usr/bin/myapp
-; called when a voicemail is left, delivered, or your voicemailbox
-; is checked, uncomment this:
+; If you need to have an external program, i.e. /usr/bin/myapp called when a
+; voicemail is left, delivered, or your voicemailbox is checked, uncomment
+; this:
;externnotify=/usr/bin/myapp
-; If you need to have an external program, i.e. /usr/bin/myapp
-; called when a voicemail password is changed,
-; uncomment this:
+; If you need to have an external program, i.e. /usr/bin/myapp called when a
+; voicemail password is changed, uncomment this:
;externpass=/usr/bin/myapp
; For the directory, you can override the intro file if you want
;directoryintro=dir-intro
@@ -54,13 +53,15 @@ maxlogins=3
;usedirectory=yes
;
; Change the from, body and/or subject, variables:
-; VM_NAME, VM_DUR, VM_MSGNUM, VM_MAILBOX, VM_CALLERID, VM_CIDNUM, VM_CIDNAME, VM_DATE
+; VM_NAME, VM_DUR, VM_MSGNUM, VM_MAILBOX, VM_CALLERID, VM_CIDNUM,
+; VM_CIDNAME, VM_DATE
;
-; Note: The emailbody config row can be up to 512 characters due to a limitation in
-; asterisk config files.
+; Note: The emailbody config row can only be up to 512 characters due to a
+; limitation in the Asterisk configuration subsystem.
;emailsubject=[PBX]: New message ${VM_MSGNUM} in mailbox ${VM_MAILBOX}
-; The following definition is very close to the default, but the default shows just
-; the CIDNAME, if it is not null, else just the CIDNUM, or "an unknown caller" if they are both null.
+; The following definition is very close to the default, but the default shows
+; just the CIDNAME, if it is not null, otherise just the CIDNUM, or "an unknown
+; caller", if they are both null.
;emailbody=Dear ${VM_NAME}:\n\n\tjust wanted to let you know you were just left a ${VM_DUR} long message (number ${VM_MSGNUM})\nin mailbox ${VM_MAILBOX} from ${VM_CALLERID}, on ${VM_DATE}, so you might\nwant to check it when you get a chance. Thanks!\n\n\t\t\t\t--Asterisk\n
;
; You can also change the Pager From: string, the pager body and/or subject.
@@ -69,7 +70,8 @@ maxlogins=3
;pagersubject=New VM
;pagerbody=New ${VM_DUR} long msg in box ${VM_MAILBOX}\nfrom ${VM_CALLERID}, on ${VM_DATE}
;
-; Set the date format on outgoing mails. Valid arguments can be found on the strftime(3) man page
+; Set the date format on outgoing mails. Valid arguments can be found on the
+; strftime(3) man page
;
; Default
emaildateformat=%A, %B %d, %Y at %r
@@ -93,7 +95,8 @@ emaildateformat=%A, %B %d, %Y at %r
; variable substitution is done on the values below.
;
; Supported values:
-; 'filename' filename of a soundfile (single ticks around the filename required)
+; 'filename' filename of a soundfile (single ticks around the filename
+; required)
; ${VAR} variable substitution
; A or a Day of week (Saturday, Sunday, ...)
; B or b or h Month name (January, February, ...)
@@ -105,8 +108,10 @@ emaildateformat=%A, %B %d, %Y at %r
; M Minute, with 00 pronounced as "o'clock"
; N Minute, with 00 pronounced as "hundred" (US military time)
; P or p AM or PM
-; Q "today", "yesterday" or ABdY (*note: not standard strftime value)
-; q "" (for today), "yesterday", weekday, or ABdY (*note: not standard strftime value)
+; Q "today", "yesterday" or ABdY
+; (*note: not standard strftime value)
+; q "" (for today), "yesterday", weekday, or ABdY
+; (*note: not standard strftime value)
; R 24 hour time, including minute
;
;
@@ -114,11 +119,13 @@ emaildateformat=%A, %B %d, %Y at %r
;
; Each mailbox is listed in the form <mailbox>=<password>,<name>,<email>,<pager_email>,<options>
; if the e-mail is specified, a message will be sent when a message is
-; received, to the given mailbox. If pager is specified, a message will be sent there as well. If the password is prefixed by '-' then it is considered to be unchangable
+; received, to the given mailbox. If pager is specified, a message will be
+; sent there as well. If the password is prefixed by '-', then it is
+; considered to be unchangable.
;
; Advanced options example is extension 4069
-; NOTE: All options can be expressed globally in the general section, and overriden in the per-mailbox
-; settings, unless listed otherwise.
+; NOTE: All options can be expressed globally in the general section, and
+; overriden in the per-mailbox settings, unless listed otherwise.
;
; tz=central ; Timezone from zonemessages above. Irrelevant if envelope=no.
; attach=yes ; Attach the voicemail to the notification email *NOT* the pager email
diff --git a/configs/vpb.conf.sample b/configs/vpb.conf.sample
index ebdbfbcdb3..d16283802c 100755
--- a/configs/vpb.conf.sample
+++ b/configs/vpb.conf.sample
@@ -1,17 +1,28 @@
+;
; V6PCI/V12PCI config file for VoiceTronix Hardware
-; Options
-; For [general] section
+;
+; Options for [general] section
+;
; type = v12pci|v6pci|v4pci
; cards = number of cards
-; indication = 1 ( To use Asterisk indication tones)
-; ecsuppthres = 0|2048|4096 (none,-24db,-18db only for use with OpenLine4)
-; dtmfidd = 3000 (Inter Digit Delay timeout for when collecting DTMF tones for dialling from a Station port, in ms)
-; ast-dtmf-det=1 ( To use Asterisk DTMF detection )
-; relaxdtmf=1 ( Used with ast-dtmf-det )
-; break-for-dtmf=no (When a native bridge occurs between 2 vpb channels, it will only break the connection for '#' and '*')
-; timer_period_ring=4000 (Set the maximum period between received rings, default 4000ms)
+; To use Asterisk indication tones
+; indication = 1
+; none,-24db,-18db only for use with OpenLine4
+; ecsuppthres = 0|2048|4096
+; Inter Digit Delay timeout for when collecting DTMF tones for dialling
+; from a Station port, in ms
+; dtmfidd = 3000
+; To use Asterisk DTMF detection
+; ast-dtmf-det=1
+; Used with ast-dtmf-det
+; relaxdtmf=1
+; When a native bridge occurs between 2 vpb channels, it will only break
+; the connection for '#' and '*'
+; break-for-dtmf=no
+; Set the maximum period between received rings, default 4000ms
+; timer_period_ring=4000
;
-; For [interface] section
+; Options for [interface] section
; board = board_number (1, 2, 3, ...)
; channel = channel_number (1,2,3...)
; mode = fxo|immediate|dialtone -- for type of line and line handling
diff --git a/configs/zapata.conf.sample b/configs/zapata.conf.sample
index ebeb82d614..06aa482837 100755
--- a/configs/zapata.conf.sample
+++ b/configs/zapata.conf.sample
@@ -103,9 +103,9 @@ switchtype=national
;privateprefix = +497115678
;unknownprefix =
;
-; PRI resetinterval: sets the time in seconds between restart of unused channels, defaults to 3600
-; minimum 60 seconds
-; some PBXs don't like channel restarts. so set the interval to a very long interval e.g. 100000000
+; PRI resetinterval: sets the time in seconds between restart of unused
+; channels, defaults to 3600; minimum 60 seconds. Some PBXs don't like
+; channel restarts. so set the interval to a very long interval e.g. 100000000
; or 'never' to disable *entirely*.
;
;resetinterval = 3600
@@ -129,58 +129,66 @@ switchtype=national
; priexclusive = yes
;
; ISDN Timers
-; All of the ISDN timers and counters that are used are configurable. Specify
-; the timer name, and its value (in ms for timers)
+; All of the ISDN timers and counters that are used are configurable. Specify
+; the timer name, and its value (in ms for timers).
;
; pritimer => t200,1000
; pritimer => t313,4000
;
; To enable transmission of facility-based ISDN supplementary services (such
-; as caller name from CPE over facility) enable this option.
+; as caller name from CPE over facility), enable this option.
; facilityenable = yes
;
;
; Signalling method (default is fxs). Valid values:
-; em: E & M
-; 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))
-; 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
+; em: E & M
+; 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))
+; 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
; gr303fxoks_net: GR-303 Signalling, FXO Loopstart, Network side
; gr303fxsks_cpe: GR-303 Signalling, FXS Loopstart, CPE 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
+; 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
+;
; 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)
+; 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)
;
signalling=fxo_ls
;
-; For Feature Group D Tandem access, to set the default CIC and OZZ use
-; these parameters:
+; For Feature Group D Tandem access, to set the default CIC and OZZ use these
+; parameters:
;defaultozz=0000
;defaultcic=303
;
@@ -197,7 +205,8 @@ signalling=fxo_ls
;
rxwink=300 ; Atlas seems to use long (250ms) winks
;
-; How long generated tones (DTMF and MF) will be played on the channel (in miliseconds)
+; How long generated tones (DTMF and MF) will be played on the channel
+; (in miliseconds)
;toneduration=100
;
; Whether or not to do distinctive ring detection on FXO lines
@@ -210,12 +219,15 @@ rxwink=300 ; Atlas seems to use long (250ms) winks
usecallerid=yes
;
; Type of caller ID signalling in use
-; bell = bell202 as used in US, v23 = v23 as used in the UK, dtmf = DTMF as used in Denmark, Sweden and Netherlands
+; bell = bell202 as used in US
+; v23 = v23 as used in the UK
+; dtmf = DTMF as used in Denmark, Sweden and Netherlands
;
;cidsignalling=bell
;
; What signals the start of caller ID
-; ring = a ring signals the start, polarity = polarity reversal signals the start
+; ring = a ring signals the start
+; polarity = polarity reversal signals the start
;
;cidstart=ring
;
@@ -227,12 +239,14 @@ hidecallerid=no
;
callwaiting=yes
;
-; Whether or not restrict outgoing caller ID (will be sent as ANI only, not available for the user)
+; Whether or not restrict outgoing caller ID (will be sent as ANI only, not
+; available for the user)
; Mostly use with FXS ports
;
;restrictcid=no
;
-; Whether or not use the caller ID presentation for the outgoing call that the calling switch is sending
+; Whether or not use the caller ID presentation for the outgoing call that the
+; calling switch is sending.
;
usecallingpres=yes
;
@@ -271,31 +285,29 @@ callreturn=yes
;
; Stutter dialtone support: If a mailbox is specified without a voicemail
; context, then when voicemail is received in a mailbox in the default
-; voicemail context in voicemail.conf, taking the phone off hook will
-; cause a stutter dialtone instead of a normal one.
+; voicemail context in voicemail.conf, taking the phone off hook will cause a
+; stutter dialtone instead of a normal one.
;
-; If a mailbox is specified *with* a voicemail context, the same will
-; result if voicemail recieved in mailbox in the specified voicemail
-; context
+; If a mailbox is specified *with* a voicemail context, the same will result
+; if voicemail recieved in mailbox in the specified voicemail context.
;
; for default voicemail context, the example below is fine:
;
;mailbox=1234
;
-; for any other voicemail context, the following will produce the
-; stutter tone:
+; for any other voicemail context, the following will produce the stutter tone:
;
;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.
+; 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.
;
echocancel=yes
;
-; Generally, it is not necessary (and in fact undesirable) to echo cancel
-; when the circuit path is entirely TDM. You may, however, reverse this
-; behavior by enabling the echo cancel during pure TDM bridging below.
+; Generally, it is not necessary (and in fact undesirable) to echo cancel when
+; the circuit path is entirely TDM. You may, however, reverse this behavior
+; by enabling the echo cancel during pure TDM bridging below.
;
echocancelwhenbridged=yes
;
@@ -309,10 +321,9 @@ echocancelwhenbridged=yes
;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.
+; 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
;
@@ -321,8 +332,8 @@ echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
;
-; Logical groups can be assigned to allow outgoing rollover. Groups
-; range from 0 to 63, and multiple groups can be specified.
+; Logical groups can be assigned to allow outgoing rollover. Groups range
+; from 0 to 63, and multiple groups can be specified.
;
group=1
;
@@ -335,19 +346,18 @@ callgroup=1
pickupgroup=1
;
-; Specify whether the channel should be answered immediately or
-; if the simple switch should provide dialtone, read digits, etc.
+; Specify whether the channel should be answered immediately or if the simple
+; switch should provide dialtone, read digits, etc.
;
immediate=no
;
-; Specify whether flash-hook transfers to 'busy' channels should complete
-; or return to the caller performing the transfer (default is yes).
+; Specify whether flash-hook transfers to 'busy' channels should complete or
+; return to the caller performing the transfer (default is yes).
;
;transfertobusy=no
;
-; CallerID can be set to "asreceived" or a specific number
-; if you want to override it. Note that "asreceived" only
-; applies to trunk interfaces.
+; CallerID can be set to "asreceived" or a specific number if you want to
+; override it. Note that "asreceived" only applies to trunk interfaces.
;
;callerid=2564286000
;
@@ -373,39 +383,36 @@ immediate=no
;
;busydetect=yes
;
-; If busydetect is enabled, is also possible to specify how many
-; busy tones to wait for before hanging up. The default is 4, but
-; better results can be achieved if set to 6 or even 8. Mind that
-; higher the number, more time is needed to hangup a channel, but
-; lower is probability to get random hangups
+; If busydetect is enabled, it is also possible to specify how many busy tones
+; to wait for before hanging up. The default is 4, but better results can be
+; achieved if 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=4
;
-; If busydetect is enabled, is also possible to specify the
-; cadence of your busy signal. In many countries it is 500mec
-; on, 500msec off.
-; Without busypattern specified, we'll accept any regular
-; sound-silence pattern than 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.
+; If busydetect is enabled, it is also possible to specify the cadence of your
+; busy signal. In many countries, it is 500msec on, 500msec off. Without
+; busypattern specified, we'll accept any regular sound-silence pattern that
+; repeats <busycount> times as a busy signal. If you specify busypattern,
+; then we'll further check the length of the sound (tone) and silence, which
+; will further reduce the chance of a false positive.
;
;busypattern=500,500
;
-; NOTE: In the Asterisk Makefile you'll find further options to tweak
-; the busy detector. If your country has a busy tone with the same
-; lengh tone and silence (as many countries do), consider defining
-; the -DBUSYDETECT_COMPARE_TONE_AND_SILENCE option.
+; NOTE: In the Asterisk Makefile you'll find further options to tweak the busy
+; detector. If your country has a busy tone with the same length tone and
+; silence (as many countries do), consider defining the
+; -DBUSYDETECT_COMPARE_TONE_AND_SILENCE option.
;
; Use a polarity reversal to mark when a outgoing call is answered by the
; remote party.
;
;answeronpolarityswitch=yes
;
-; In some countries, a polarity reversal is used to signal the disconnect
-; of a phone line. If the hanguponpolarityswitch option is selected, the
-; call will be considered "hung up" on a polarity reversal
+; 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
;
@@ -413,13 +420,13 @@ immediate=no
; 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.
+; 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"
+; Few zones are supported at the time of this writing, but may be selected
+; with "progzone"
;
-; This feature can also easily detect false hangups. The symptoms of this
-; is being disconnected in the middle of a call for no reason.
+; 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=us
@@ -446,15 +453,15 @@ immediate=no
;
;musiconhold=default
;
-; PRI channels can have an idle extension and a minunused number. So long
-; as at least "minunused" channels are idle, chan_zap will try to call
-; "idledial" on them, and then dump them into the PBX in the "idleext"
-; extension (which is of the form exten@context). When channels are needed
-; the "idle" calls are disconnected (so long as there are at least "minidle"
-; calls still running, of course) to make more channels available. The
-; primary use of this is to create a dynamic service, where idle channels
-; are bundled through multilink PPP, thus more efficiently utilizing
-; combined voice/data services than conventional fixed mappings/muxings.
+; PRI channels can have an idle extension and a minunused number. So long as
+; at least "minunused" channels are idle, chan_zap will try to call "idledial"
+; on them, and then dump them into the PBX in the "idleext" extension (which
+; is of the form exten@context). When channels are needed the "idle" calls
+; are disconnected (so long as there are at least "minidle" calls still
+; running, of course) to make more channels available. The primary use of
+; this is to create a dynamic service, where idle channels are bundled through
+; multilink PPP, thus more efficiently utilizing combined voice/data services
+; than conventional fixed mappings/muxings.
;
;idledial=6999
;idleext=6999@dialout
@@ -465,10 +472,10 @@ immediate=no
;
;jitterbuffers=4
;
-; You can define your own custom ring cadences here. You can define up to
-; 8 pairs. If the silence is negative, it indicates where the callerid
-; spill is to be placed. Also, if you define any custom cadences, the
-; default cadences will be turned off.
+; You can define your own custom ring cadences here. You can define up to 8
+; pairs. If the silence is negative, it indicates where the callerid spill is
+; to be placed. Also, if you define any custom cadences, the default cadences
+; will be turned off.
;
; Syntax is: cadence=ring,silence[,ring,silence[...]]
;
@@ -479,11 +486,11 @@ immediate=no
;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
+; Each channel consists of the channel number or range. It inherits the
+; parameters that were specified above its declaration.
;
-; For GR-303, CRV's are created like channels except they must start
-; with the trunk group followed by a colon, e.g.:
+; For GR-303, CRV's are created like channels except they must start with the
+; trunk group followed by a colon, e.g.:
;
; crv => 1:1
; crv => 2:1-2,5-8
@@ -506,9 +513,8 @@ immediate=no
;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.
+; For example, maybe we have some other channels which start out in a
+; different context and use E & M signalling instead.
;
;context=remote
;sigalling=em
@@ -538,9 +544,9 @@ immediate=no
;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.
+; Sample PRI (CPE) config: Specify the switchtype, the signalling as either
+; pri_cpe or pri_net for CPE or Network termination, and generally you will
+; want to create a single "group" for all channels of the PRI.
;
; switchtype = national
; signalling = pri_cpe
--
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