[Asterisk-cvs] asterisk/configs alarmreceiver.conf.sample, 1.1,
1.2 codecs.conf.sample, 1.4, 1.5 extensions.conf.sample, 1.46,
1.47 iax.conf.sample, 1.55, 1.56 iaxprov.conf.sample, 1.1,
1.2 indications.conf.sample, 1.29, 1.30 logger.conf.sample,
1.14, 1.15 manager.conf.sample, 1.6, 1.7 meetme.conf.sample,
1.4, 1.5 mgcp.conf.sample, 1.11, 1.12 modules.conf.sample, 1.6,
1.7 musiconhold.conf.sample, 1.9, 1.10 queues.conf.sample,
1.31, 1.32 sip.conf.sample, 1.72, 1.73 voicemail.conf.sample,
1.53, 1.54 vpb.conf.sample, 1.11, 1.12 zapata.conf.sample,
1.53, 1.54
kpfleming
kpfleming
Tue Oct 4 18:56:12 CDT 2005
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Update of /usr/cvsroot/asterisk/configs
In directory mongoose.digium.com:/tmp/cvs-serv1572/configs
Modified Files:
alarmreceiver.conf.sample codecs.conf.sample
extensions.conf.sample iax.conf.sample iaxprov.conf.sample
indications.conf.sample logger.conf.sample manager.conf.sample
meetme.conf.sample mgcp.conf.sample modules.conf.sample
musiconhold.conf.sample queues.conf.sample sip.conf.sample
voicemail.conf.sample vpb.conf.sample zapata.conf.sample
Log Message:
make sample config files easier to ready (issue #5371)
Index: alarmreceiver.conf.sample
===================================================================
RCS file: /usr/cvsroot/asterisk/configs/alarmreceiver.conf.sample,v
retrieving revision 1.1
retrieving revision 1.2
diff -u -d -r1.1 -r1.2
--- alarmreceiver.conf.sample 21 Jun 2004 19:28:34 -0000 1.1
+++ alarmreceiver.conf.sample 4 Oct 2005 22:51:59 -0000 1.2
@@ -22,8 +22,9 @@
;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 @@
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 @@
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:
Index: codecs.conf.sample
===================================================================
RCS file: /usr/cvsroot/asterisk/configs/codecs.conf.sample,v
retrieving revision 1.4
retrieving revision 1.5
diff -u -d -r1.4 -r1.5
--- codecs.conf.sample 26 Aug 2005 20:14:06 -0000 1.4
+++ codecs.conf.sample 4 Oct 2005 22:51:59 -0000 1.5
@@ -12,7 +12,8 @@
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]
Index: extensions.conf.sample
===================================================================
RCS file: /usr/cvsroot/asterisk/configs/extensions.conf.sample,v
retrieving revision 1.46
retrieving revision 1.47
diff -u -d -r1.46 -r1.47
--- extensions.conf.sample 27 Jul 2005 05:45:52 -0000 1.46
+++ extensions.conf.sample 4 Oct 2005 22:51:59 -0000 1.47
@@ -52,13 +52,15 @@
;
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 @@
; 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 at provider
@@ -443,11 +449,11 @@
;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
Index: iax.conf.sample
===================================================================
RCS file: /usr/cvsroot/asterisk/configs/iax.conf.sample,v
retrieving revision 1.55
retrieving revision 1.56
diff -u -d -r1.55 -r1.56
--- iax.conf.sample 15 Sep 2005 02:25:06 -0000 1.55
+++ iax.conf.sample 4 Oct 2005 22:51:59 -0000 1.56
@@ -74,11 +74,11 @@
; 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 @@
; 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 @@
;
; 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 @@
;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 @@
;
;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 @@
; 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 @@
;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 @@
;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 @@
;secret=moofoo
;context=default
;permit=0.0.0.0/0.0.0.0
+
Index: iaxprov.conf.sample
===================================================================
RCS file: /usr/cvsroot/asterisk/configs/iaxprov.conf.sample,v
retrieving revision 1.1
retrieving revision 1.2
diff -u -d -r1.1 -r1.2
--- iaxprov.conf.sample 7 Jul 2004 09:34:01 -0000 1.1
+++ iaxprov.conf.sample 4 Oct 2005 22:51:59 -0000 1.2
@@ -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 @@
;
;[*]
;
-; 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
+
Index: indications.conf.sample
===================================================================
RCS file: /usr/cvsroot/asterisk/configs/indications.conf.sample,v
retrieving revision 1.29
retrieving revision 1.30
diff -u -d -r1.29 -r1.30
--- indications.conf.sample 23 Aug 2005 12:38:48 -0000 1.29
+++ indications.conf.sample 4 Oct 2005 22:51:59 -0000 1.30
@@ -16,7 +16,7 @@
; [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 @@
; 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 @@
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
Index: logger.conf.sample
===================================================================
RCS file: /usr/cvsroot/asterisk/configs/logger.conf.sample,v
retrieving revision 1.14
retrieving revision 1.15
diff -u -d -r1.14 -r1.15
--- logger.conf.sample 22 Aug 2005 21:19:59 -0000 1.14
+++ logger.conf.sample 4 Oct 2005 22:51:59 -0000 1.15
@@ -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
Index: manager.conf.sample
===================================================================
RCS file: /usr/cvsroot/asterisk/configs/manager.conf.sample,v
retrieving revision 1.6
retrieving revision 1.7
diff -u -d -r1.6 -r1.7
--- manager.conf.sample 4 Oct 2005 22:25:15 -0000 1.6
+++ manager.conf.sample 4 Oct 2005 22:51:59 -0000 1.7
@@ -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
Index: meetme.conf.sample
===================================================================
RCS file: /usr/cvsroot/asterisk/configs/meetme.conf.sample,v
retrieving revision 1.4
retrieving revision 1.5
diff -u -d -r1.4 -r1.5
--- meetme.conf.sample 17 Mar 2005 15:56:55 -0000 1.4
+++ meetme.conf.sample 4 Oct 2005 22:51:59 -0000 1.5
@@ -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
+
Index: mgcp.conf.sample
===================================================================
RCS file: /usr/cvsroot/asterisk/configs/mgcp.conf.sample,v
retrieving revision 1.11
retrieving revision 1.12
diff -u -d -r1.11 -r1.12
--- mgcp.conf.sample 5 Apr 2005 21:40:37 -0000 1.11
+++ mgcp.conf.sample 4 Oct 2005 22:51:59 -0000 1.12
@@ -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
Index: modules.conf.sample
===================================================================
RCS file: /usr/cvsroot/asterisk/configs/modules.conf.sample,v
retrieving revision 1.6
retrieving revision 1.7
diff -u -d -r1.6 -r1.7
--- modules.conf.sample 5 Jul 2005 22:11:42 -0000 1.6
+++ modules.conf.sample 4 Oct 2005 22:51:59 -0000 1.7
@@ -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
Index: musiconhold.conf.sample
===================================================================
RCS file: /usr/cvsroot/asterisk/configs/musiconhold.conf.sample,v
retrieving revision 1.9
retrieving revision 1.10
diff -u -d -r1.9 -r1.10
--- musiconhold.conf.sample 25 Aug 2005 16:11:46 -0000 1.9
+++ musiconhold.conf.sample 4 Oct 2005 22:51:59 -0000 1.10
@@ -26,7 +26,8 @@
;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
Index: queues.conf.sample
===================================================================
RCS file: /usr/cvsroot/asterisk/configs/queues.conf.sample,v
retrieving revision 1.31
retrieving revision 1.32
diff -u -d -r1.31 -r1.32
--- queues.conf.sample 2 Sep 2005 19:27:01 -0000 1.31
+++ queues.conf.sample 4 Oct 2005 22:51:59 -0000 1.32
@@ -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 @@
;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 @@
;
; 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 @@
; 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 @@
;
; 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
;
Index: sip.conf.sample
===================================================================
RCS file: /usr/cvsroot/asterisk/configs/sip.conf.sample,v
retrieving revision 1.72
retrieving revision 1.73
diff -u -d -r1.72 -r1.73
--- sip.conf.sample 4 Oct 2005 19:05:40 -0000 1.72
+++ sip.conf.sample 4 Oct 2005 22:51:59 -0000 1.73
@@ -108,12 +108,11 @@
;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 @@
; 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 @@
;
;register => 2345:password at 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 @@
; 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 @@
;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 @@
; 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 @@
;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]
Index: voicemail.conf.sample
===================================================================
RCS file: /usr/cvsroot/asterisk/configs/voicemail.conf.sample,v
retrieving revision 1.53
retrieving revision 1.54
diff -u -d -r1.53 -r1.54
--- voicemail.conf.sample 1 Oct 2005 01:24:15 -0000 1.53
+++ voicemail.conf.sample 4 Oct 2005 22:51:59 -0000 1.54
@@ -10,8 +10,8 @@
;serveremail=asterisk at 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 @@
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 @@
;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 @@
;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 @@
; 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 @@
; 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 @@
;
; 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
Index: vpb.conf.sample
===================================================================
RCS file: /usr/cvsroot/asterisk/configs/vpb.conf.sample,v
retrieving revision 1.11
retrieving revision 1.12
diff -u -d -r1.11 -r1.12
--- vpb.conf.sample 22 Jun 2005 23:54:47 -0000 1.11
+++ vpb.conf.sample 4 Oct 2005 22:51:59 -0000 1.12
@@ -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
Index: zapata.conf.sample
===================================================================
RCS file: /usr/cvsroot/asterisk/configs/zapata.conf.sample,v
retrieving revision 1.53
retrieving revision 1.54
diff -u -d -r1.53 -r1.54
--- zapata.conf.sample 1 Sep 2005 19:02:37 -0000 1.53
+++ zapata.conf.sample 4 Oct 2005 22:51:59 -0000 1.54
@@ -103,9 +103,9 @@
;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 @@
; 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 @@
;
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 @@
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 @@
;
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 @@
;
; 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 at 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 @@
;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 @@
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 @@
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 @@
;
;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 @@
; 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 @@
;
;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 at 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 at 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 at dialout
@@ -465,10 +472,10 @@
;
;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 @@
;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 @@
;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 @@
;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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