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

@@ -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:

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]

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

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
+

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
+

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

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

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

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
+

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
 

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

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

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
 ;

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]

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

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

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