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
1.2-netsec
Kevin P. Fleming 20 years ago
parent e338682949
commit f387421905

@ -22,8 +22,9 @@ timestampformat = %a %b %d, %Y @ %H:%M:%S %Z
;eventcmd = yourprogram -yourargs ... ;eventcmd = yourprogram -yourargs ...
; ;
; Specify a spool directory for the event files. This setting is required if you want the app to be useful. ; Specify a spool directory for the event files. This setting is required
; Event files written to the spool directory will be of the template event-XXXXXX, where XXXXXX is a random ; 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. ; and unique alphanumeric string.
; ;
; Default is none, and the events will be dropped on the floor. ; 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 eventspooldir = /tmp
; ;
; The alarmreceiver app can either log the events one-at-a-time to individual files in the spool ; The alarmreceiver app can either log the events one-at-a-time to individual
; directory, or it can store them until the caller disconnects and write them all to one file. ; 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. ; The default setting for logindividualevents is no.
; ;
@ -41,32 +43,34 @@ eventspooldir = /tmp
logindividualevents = no logindividualevents = no
; ;
; The timeout for receiving the first DTMF digit is adjustable from 1000 msec. to 10000 msec. The ; The timeout for receiving the first DTMF digit is adjustable from 1000 msec.
; default is 2000 msec. Note: if you wish to test the receiver by entering digits manually, set this ; to 10000 msec. The default is 2000 msec. Note: if you wish to test the
; to a reasonable time out like 10000 milliseconds. ; receiver by entering digits manually, set this to a reasonable time out
; like 10000 milliseconds.
fdtimeout = 2000 fdtimeout = 2000
; ;
; The timeout for receiving subsequent DTMF digits is adjustable from 110 msec. to 4000 msec. The ; The timeout for receiving subsequent DTMF digits is adjustable from
; default is 200 msec. Note: if you wish to test the receiver by entering digits manually, set this ; 110 msec. to 4000 msec. The default is 200 msec. Note: if you wish to test
; to a reasonable time out like 4000 milliseconds. ; the receiver by entering digits manually, set this to a reasonable time out
; like 4000 milliseconds.
; ;
sdtimeout = 200 sdtimeout = 200
; ;
; The loudness of the ACK and Kissoff tones is adjustable from 100 to 8192. The default is 8192 ; The loudness of the ACK and Kissoff tones is adjustable from 100 to 8192.
; This shouldn't need to be messed with, but is included just in case there are problems with ; The default is 8192. This shouldn't need to be messed with, but is included
; signal levels. ; just in case there are problems with signal levels.
; ;
loudness = 8192 loudness = 8192
; ;
; The db-family setting allows the user to capture statistics on the number of calls, and the errors ; The db-family setting allows the user to capture statistics on the number of
; the alarm receiver sees. The default is for no db-family name to be defined and the database logging ; calls, and the errors the alarm receiver sees. The default is for no
; to be turned off. ; db-family name to be defined and the database logging to be turned off.
; ;
;db-family = yourfamily: ;db-family = yourfamily:

@ -12,7 +12,8 @@ complexity => 2
enhancement => true enhancement => true
; voice activity detection [true / false] ; 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 vad => true
; variable bit rate [true / false] ; variable bit rate [true / false]

@ -52,13 +52,15 @@ clearglobalvars=no
; ;
priorityjumping=no priorityjumping=no
; ;
; You can include other config files, use the #include command (without the ';') ; You can include other config files, use the #include command
; Note that this is different from the "include" command that includes contexts within ; (without the ';'). Note that this is different from the "include" command
; other contexts. The #include command works in all asterisk configuration files. ; that includes contexts within other contexts. The #include command works
; in all asterisk configuration files.
;#include "filename.conf" ;#include "filename.conf"
; The "Globals" category contains global variables that can be referenced ; 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 ; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
; ;
[globals] [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 ; 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: ; 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 lowest-numbered non-busy Zap channel
; G: select the highest-numbered non-busy Zap channel (aka. descending sequential hunt group). ; (aka. ascending sequential hunt group).
; r: use a round-robin search, starting at the next highest channel than last time (aka. ascending rotary hunt group). ; G: select the highest-numbered non-busy Zap channel
; R: use a round-robin search, starting at the next lowest channel than last time (aka. descending rotary hunt group). ; (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) TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
;TRUNK=IAX2/user:pass@provider ;TRUNK=IAX2/user:pass@provider
@ -443,11 +449,11 @@ include => demo
;exten => _41X.,1,Dial(SIP/${EXTEN:2}@sipprovider,,r) ;exten => _41X.,1,Dial(SIP/${EXTEN:2}@sipprovider,,r)
;exten => _42X.,1,Dial(SIP/user:passwd@${EXTEN:2}@otherprovider.net,30,rT) ;exten => _42X.,1,Dial(SIP/user:passwd@${EXTEN:2}@otherprovider.net,30,rT)
; Real extensions would go here. Generally you want real extensions to be 4 or 5 ; Real extensions would go here. Generally you want real extensions to be
; digits long (although there is no such requirement) and start with a single ; 4 or 5 digits long (although there is no such requirement) and start with a
; digit that is fairly large (like 6 or 7) so that you have plenty of room to ; single digit that is fairly large (like 6 or 7) so that you have plenty of
; overlap extensions and menu options without conflict. You can alias them with ; room to overlap extensions and menu options without conflict. You can alias
; names, too and use global variables ; them with names, too, and use global variables
;exten => 6245,hint,SIP/Grandstream1&SIP/Xlite1,Joe Schmoe ; Channel hints for presence ;exten => 6245,hint,SIP/Grandstream1&SIP/Xlite1,Joe Schmoe ; Channel hints for presence
;exten => 6245,1,Dial(SIP/Grandstream1,20,rt) ; permit transfer ;exten => 6245,1,Dial(SIP/Grandstream1,20,rt) ; permit transfer

@ -74,11 +74,11 @@ disallow=lpc10 ; Icky sound quality... Mr. Roboto.
; The jitter buffer's function is to compensate for varying ; The jitter buffer's function is to compensate for varying
; network delay. ; network delay.
; ;
; There are presently two jitterbuffer implementations available for * and chan_iax2; ; There are presently two jitterbuffer implementations available for Asterisk
; the classic and the new, channel/application independent implementation. These ; and chan_iax2; the classic and the new, channel/application independent
; are controlled at compile-time. The new jitterbuffer additionally has support for PLC ; implementation. These are controlled at compile-time. The new jitterbuffer
; which greatly improves quality as the jitterbuffer adapts size, and in compensating for lost ; additionally has support for PLC which greatly improves quality as the
; packets. ; jitterbuffer adapts size, and in compensating for lost packets.
; ;
; All the jitter buffer settings except dropcount are in milliseconds. ; All the jitter buffer settings except dropcount are in milliseconds.
; The jitter buffer works for INCOMING audio - the outbound audio ; 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 ; 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 ; we don't want to do jitterbuffering on the switch, since the endpoints
; can each handle this. However, some endpoints may have poor jitterbuffers ; 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] ; [This option presently applies only to the new jitterbuffer implementation]
; ;
; dropcount: the jitter buffer is sized such that no more than "dropcount" ; 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 ; resyncthreshold: when the jitterbuffer notices a significant change in delay
; that continues over a few frames, it will resync, assuming that the change in ; 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 ; delay was caused by a timestamping mix-up. The threshold for noticing a
; in delay is measured as twice the measured jitter plus this resync threshold. ; change in delay is measured as twice the measured jitter plus this resync
; Resycning can be disabled by setting this parameter to -1. ; threshold.
; Resyncing can be disabled by setting this parameter to -1.
; [This option presently applies only to the new jitterbuffer implementation] ; [This option presently applies only to the new jitterbuffer implementation]
; ;
; maxjitterinterps: the maximum number of interpolation frames the jitterbuffer should ; maxjitterinterps: the maximum number of interpolation frames the jitterbuffer
; return in a row. Since some clients do not send CNG/DTX frames to indicate ; should return in a row. Since some clients do not send CNG/DTX frames to
; silence, the jitterbuffer will assume silence has begun after returning this ; indicate silence, the jitterbuffer will assume silence has begun after
; many interpolations. This prevents interpolating throughout a long silence. ; returning this many interpolations. This prevents interpolating throughout
; a long silence.
; [This option presently applies only to the new jitterbuffer implementation] ; [This option presently applies only to the new jitterbuffer implementation]
; ;
; maxexcessbuffer: If conditions improve after a period of high jitter, ; 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) ;trunkfreq=20 ; How frequently to send trunk msgs (in ms)
; Should we send timestamps for the individual sub-frames within trunk frames? ; 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 ; There is a small bandwidth use for these (less than 1kbps/call), but they
; that frame timestamps get sent end-to-end properly. If both ends of all your trunks ; ensure that frame timestamps get sent end-to-end properly. If both ends of
; go directly to TDM, _and_ your trunkfreq equals the frame length for your codecs, you ; all your trunks go directly to TDM, _and_ your trunkfreq equals the frame
; can probably suppress these. The receiver must also support this feature, although ; length for your codecs, you can probably suppress these. The receiver must
; they do not also need to have it enabled. ; also support this feature, although they do not also need to have it enabled.
; ;
; trunktimestamps=yes ; trunktimestamps=yes
; ;
@ -217,22 +220,21 @@ tos=lowdelay
; ;
;mailboxdetail=yes ;mailboxdetail=yes
; ;
; If regcontext is specified, Asterisk will dynamically ; If regcontext is specified, Asterisk will dynamically create and destroy
; create and destroy a NoOp priority 1 extension for a given ; a NoOp priority 1 extension for a given peer who registers or unregisters
; peer who registers or unregisters with us. The actual extension ; with us. The actual extension is the 'regexten' parameter of the registering
; is the 'regexten' parameter of the registering peer or its ; peer or its name if 'regexten' is not provided. More than one regexten
; name if 'regexten' is not provided. More than one regexten may be supplied ; may be supplied if they are separated by '&'. Patterns may be used in
; if they are separated by '&'. Patterns may be used in regexten. ; regexten.
; ;
;regcontext=iaxregistrations ;regcontext=iaxregistrations
; ;
; If we don't get ACK to our NEW within 2000ms, and autokill is set ; If we don't get ACK to our NEW within 2000ms, and autokill is set to yes,
; to yes, then we cancel the whole thing (that's enough time for one ; then we cancel the whole thing (that's enough time for one retransmission
; retransmission only). This is used to keep things from stalling for a long ; only). This is used to keep things from stalling for a long time for a host
; time for a host that is not available, but would be ill advised for bad ; that is not available, but would be ill advised for bad connections. In
; connections. In addition to 'yes' or 'no' you can also specify a number ; addition to 'yes' or 'no' you can also specify a number of milliseconds.
; of milliseconds. See 'qualify' for individual peers to turn on for just ; See 'qualify' for individual peers to turn on for just a specific peer.
; a specific peer.
; ;
autokill=yes autokill=yes
; ;
@ -274,8 +276,8 @@ autokill=yes
; has expired based on its registration interval, used the stored ; has expired based on its registration interval, used the stored
; address information regardless. (yes|no) ; address information regardless. (yes|no)
; Guest sections for unauthenticated connection attempts. Just ; Guest sections for unauthenticated connection attempts. Just specify an
; specify an empty secret, or provide no secret section. ; empty secret, or provide no secret section.
; ;
[guest] [guest]
type=user type=user
@ -310,14 +312,13 @@ inkeys=freeworlddialup
;context=dundi-e164-local ;context=dundi-e164-local
; ;
; Further user sections may be added, specifying a context and a ; Further user sections may be added, specifying a context and a secret used
; secret used for connections with that given authentication name. ; for connections with that given authentication name. Limited IP based
; Limited IP based access control is allowed by use of "allow" and ; access control is allowed by use of "allow" and "deny" keywords. Multiple
; "deny" keywords. Multiple rules are permitted. Multiple permitted ; rules are permitted. Multiple permitted contexts may be specified, in
; contexts may be specified, in which case the first will be the default. ; which case the first will be the default. You can also override caller*ID
; You can also override caller*ID so that when you receive a call you ; so that when you receive a call you set the Caller*ID to be what you want
; set the Caller*ID to be what you want instead of trusting what ; instead of trusting what the remote user provides
; the remote user provides
; ;
; There are three authentication methods that are supported: md5, plaintext, ; There are three authentication methods that are supported: md5, plaintext,
; and rsa. The least secure is "plaintext", which sends passwords cleartext ; 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 ;jitterbuffer=no ; Turn off jitter buffer for this peer
; ;
; Peers can remotely register as well, so that they can be ; Peers can remotely register as well, so that they can be mobile. Default
; mobile. Default IP's can also optionally be given but ; IP's can also optionally be given but are not required. Caller*ID can be
; are not required. Caller*ID can be suggested to the other ; suggested to the other side as well if it is for example a phone instead of
; side as well if it is for example a phone instead of another ; another PBX.
; PBX.
; ;
;[dynamichost] ;[dynamichost]
@ -410,3 +410,4 @@ host=216.207.245.47
;secret=moofoo ;secret=moofoo
;context=default ;context=default
;permit=0.0.0.0/0.0.0.0 ;permit=0.0.0.0/0.0.0.0

@ -1,25 +1,22 @@
; ;
; IAX2 Provisioning Information ; IAX2 Provisioning Information
; ;
; Contains provisioning information for templates ; Contains provisioning information for templates and for specific service
; and for specific service entries. ; entries.
; ;
; Templates provide a group of settings from which provisioning takes ; Templates provide a group of settings from which provisioning takes place.
; place. A template may be based upon any template that has been ; A template may be based upon any template that has been specified before
; specified before it. If the template that an entry is based on is not ; it. If the template that an entry is based on is not specified then it is
; specified then it is presumed to be 'default' (unless it is the first ; presumed to be 'default' (unless it is the first of course).
; of course).
; ;
; Templates which begin with 'si-' are used for provisioning ; Templates which begin with 'si-' are used for provisioning units with
; units with specific service identifiers. For example the ; specific service identifiers. For example the entry "si-000364000126"
; entry "si-000364000126" would be used when the device with the ; would be used when the device with the corresponding service identifier of
; corresponding service identifier of "000364000126" attempts ; "000364000126" attempts to register or make a call.
; to register or make a call.
; ;
[default] [default]
; ;
; The port number the device should use to bind to. The default ; The port number the device should use to bind to. The default is 4569.
; is 4569
; ;
;port=4569 ;port=4569
; ;
@ -27,14 +24,13 @@
; ;
;server=192.168.69.3 ;server=192.168.69.3
; ;
; altserver is the BACKUP server for registration and placing calls ; altserver is the BACKUP server for registration and placing calls in the
; in the event the primary server is unavailable. ; event the primary server is unavailable.
; ;
;altserver=192.168.69.4 ;altserver=192.168.69.4
; ;
; port is the port number to use for IAX2 outbound. The ; port is the port number to use for IAX2 outbound. The connections to the
; connections to the server and altserver -- default is of course ; server and altserver -- default is of course 4569.
; 4569.
;serverport=4569 ;serverport=4569
; ;
; language is the preferred language for the device ; 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 ; If specified, the '*' provisioning is used for all devices which do not
; not have another provisioning entry within the file. If unspecified, no ; have another provisioning entry within the file. If unspecified, no
; provisioning will take place for devices which have no entry. DO NOT ; provisioning will take place for devices which have no entry. DO NOT
; USE A '*' PROVISIONING ENTRY UNLESS YOU KNOW WHAT YOU'RE DOING. ; USE A '*' PROVISIONING ENTRY UNLESS YOU KNOW WHAT YOU'RE DOING.
; ;
;template=default ;template=default

@ -16,7 +16,7 @@ country=us ; default location
; [example] ; [example]
; description = string ; description = string
; The full name of your country, in English ; The full name of your country, in English.
; alias = iso[,iso]* ; alias = iso[,iso]*
; List of other countries 2-letter iso codes, which have the same ; List of other countries 2-letter iso codes, which have the same
; tone indications. ; tone indications.
@ -31,14 +31,16 @@ country=us ; default location
; callwaiting = tonelist ; callwaiting = tonelist
; Set of tones played when there is a call waiting in the background. ; Set of tones played when there is a call waiting in the background.
; dialrecall = tonelist ; 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 ; record = tonelist
; Set of tones played when call recording is in progress ; Set of tones played when call recording is in progress.
; info = tonelist ; 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 ; 'name' = tonelist
; Every other variable will be available as a shortcut for the "PlayList" command ; Every other variable will be available as a shortcut for the "PlayList" command
; but will not automaticly be used by Asterisk. ; but will not be used automatically by Asterisk.
; ;
; ;
; The tonelist itself is defined by a comma-separated sequence of elements. ; 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 description = South Africa
; http://www.cisco.com/univercd/cc/td/doc/product/tel_pswt/vco_prod/safr_sup/saf02.htm ; 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) ; (definitions for other countries can also be found there)
; Note, though, that South Africa uses two switch types in their network - Alcatel ; Note, though, that South Africa uses two switch types in their network --
; switches - mainly in the Western Cape, and Siemens elsewhere. ; Alcatel switches -- mainly in the Western Cape, and Siemens elsewhere.
; The former use 383+417 in dial, ringback etc. The latter use 400*33 ; The former use 383+417 in dial, ringback etc. The latter use 400*33
; I've provided both, uncomment the ones you prefer ; I've provided both, uncomment the ones you prefer
ringcadance = 400,200,400,2000 ringcadance = 400,200,400,2000

@ -16,10 +16,12 @@
; This appends the hostname to the name of the log files. ; This appends the hostname to the name of the log files.
;appendhostname = yes ;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 ;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 ;event_log = no
; ;
; ;
@ -44,17 +46,16 @@
; ;
; Special filename "console" represents the system console ; Special filename "console" represents the system console
; ;
; We highly recommend that you DO NOT turn on debug mode if you ; We highly recommend that you DO NOT turn on debug mode if you are simply
; are simply running a production system. Debug mode turns on a ; running a production system. Debug mode turns on a LOT of extra messages,
; LOT of extra messages, most of which you are unlikely to understand ; most of which you are unlikely to understand without an understanding of
; without an understanding of the underlying code. Do NOT report ; the underlying code. Do NOT report debug messages as code issues, unless
; debug messages as code issues, unless you have a specific issue that ; you have a specific issue that you are attempting to debug. They are
; you are attempting to debug. They are messages for just that -- ; messages for just that -- debugging -- and do not rise to the level of
; debugging -- and do not rise to the level of something that merit ; something that merit your attention as an Asterisk administrator. Debug
; your attention as an Asterisk administrator. Debug messages are also ; messages are also very verbose and can and do fill up logfiles quickly;
; very verbose and can and do fill up logfiles quickly; this is another ; this is another reason not to have debug mode on a production system unless
; reason not to have debug mode on a production system unless you are ; you are in the process of debugging a specific issue.
; in the process of debugging a specific issue.
; ;
;debug => debug ;debug => debug
console => notice,warning,error console => notice,warning,error

@ -1,23 +1,19 @@
; ;
; AMI - The Asterisk Manager Interface ; AMI - The Asterisk Manager Interface
; ;
; Third party application call management support ; Third party application call management support and PBX event supervision
; and PBX event supervision
; ;
; This configuration file is read every time someone ; This configuration file is read every time someone logs in
; logs in
; ;
; Use the "show manager commands" at the CLI to list ; Use the "show manager commands" at the CLI to list available manager commands
; availabale manager commands and their authorization ; and their authorization levels.
; levels.
; ;
; "show manager command <command>" will show a help text. ; "show manager command <command>" will show a help text.
; ;
; ------------------- SECURITY NOTE ----------------- ; ---------------------------- SECURITY NOTE -------------------------------
; Note that you should not enable the AMI on a public ; Note that you should not enable the AMI on a public IP address. If needed,
; IP address. If needed, block this TCP port with ; block this TCP port with iptables (or another FW software) and reach it
; iptables (or another FW software) and reach it ; with IPsec, SSH, or SSL vpn tunnel
; with IPsec, SSH or SSL vpn tunnel
; ;
[general] [general]
enabled = no enabled = no

@ -1,6 +1,5 @@
; ;
; Configuration file for MeetMe simple conference rooms ; Configuration file for MeetMe simple conference rooms for Asterisk of course.
; for Asterisk of course.
; ;
; This configuration file is read every time you call app meetme() ; This configuration file is read every time you call app meetme()
; ;
@ -10,3 +9,4 @@
; ;
;conf => 1234 ;conf => 1234
;conf => 2345,9938 ;conf => 2345,9938

@ -45,7 +45,8 @@
; ;
;context=local ;context=local
;host=dynamic ;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 ;slowsequence=yes ; The DPH100M does not follow MGCP standards for sequencing
;line => aaln/1 ;line => aaln/1

@ -7,11 +7,12 @@
[modules] [modules]
autoload=yes autoload=yes
; ;
; Any modules that need to be loaded before the Asterisk core has been initialized ; Any modules that need to be loaded before the Asterisk core has been
; (just after the logger has been initialized) can be loaded using 'preload'. This ; initialized (just after the logger has been initialized) can be loaded
; will frequently be needed if you wish to map all module configuration files into ; using 'preload'. This will frequently be needed if you wish to map all
; Realtime storage, since the Realtime driver will need to be loaded before the ; module configuration files into Realtime storage, since the Realtime
; modules using those configuration files are initialized. ; driver will need to be loaded before the modules using those configuration
; files are initialized.
; ;
; An example of loading ODBC support would be: ; An example of loading ODBC support would be:
;preload => res_odbc.so ;preload => res_odbc.so

@ -26,7 +26,8 @@ directory=/var/lib/asterisk/mohmp3
;application=/usr/bin/streamplayer 192.168.100.52 888 ;application=/usr/bin/streamplayer 192.168.100.52 888
;format=ulaw ;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] ;[solaris]
;mode=custom ;mode=custom
;directory=/var/lib/asterisk/mohmp3 ;directory=/var/lib/asterisk/mohmp3

@ -9,8 +9,8 @@
; ;
persistentmembers = yes persistentmembers = yes
; ;
; Note that a timeout to fail out of a queue may be passed as part of application call ; Note that a timeout to fail out of a queue may be passed as part of
; from extensions.conf: ; an application call from extensions.conf:
; Queue(queuename|[options]|[optionalurl]|[announceoverride]|[timeout]) ; Queue(queuename|[options]|[optionalurl]|[announceoverride]|[timeout])
; example: Queue(dave|t|||45) ; example: Queue(dave|t|||45)
@ -43,7 +43,8 @@ persistentmembers = yes
;strategy = ringall ;strategy = ringall
; ;
; Second settings for service level (default 0) ; 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 ;servicelevel = 60
; ;
; A context may be specified, in which if the user types a SINGLE ; 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? ; 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 ; announce-round-seconds = 10
; ;
@ -119,26 +121,29 @@ persistentmembers = yes
; To enable monitoring, simply specify "monitor-format"; it will be disabled ; To enable monitoring, simply specify "monitor-format"; it will be disabled
; otherwise. ; otherwise.
; ;
; You can specify the monitor filename with by calling Set(MONITOR_FILENAME=foo) ; You can specify the monitor filename with by calling
; Otherwise it will use ${UNIQUEID} ; Set(MONITOR_FILENAME=foo)
; Otherwise it will use MONITOR_FILENAME=${UNIQUEID}
; ;
; monitor-format = gsm|wav|wav49 ; 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 ; monitor-join = yes
; ;
; This setting controls whether callers can join a queue with no members. There are three ; This setting controls whether callers can join a queue with no members. There
; choices: ; are three choices:
; ;
; yes - callers can 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 ; no - callers cannot join a queue with no members
; strict - callers cannot join a queue with no members or only unavailable members ; strict - callers cannot join a queue with no members or only unavailable
; members
; ;
; joinempty = yes ; joinempty = yes
; ;
; If you wish to remove callers from the queue when new callers cannot join, set this setting ; If you wish to remove callers from the queue when new callers cannot join,
; to one of the same choices for 'joinempty' ; set this setting to one of the same choices for 'joinempty'
; ;
; leavewhenempty = yes ; leavewhenempty = yes
; ;
@ -155,14 +160,15 @@ persistentmembers = yes
; ;
; eventmemberstatusoff = no ; eventmemberstatusoff = no
; ;
; If you wish to report the caller's hold time to the member before they are connected ; If you wish to report the caller's hold time to the member before they are
; to the caller, set this to yes. ; connected to the caller, set this to yes.
; ;
; reportholdtime = no ; reportholdtime = no
; ;
; ;
; If you wish to have a delay before the member is connected to the caller (or before the member ; If you wish to have a delay before the member is connected to the caller (or
; hears any announcement messages), set this to the number of seconds to delay. ; before the member hears any announcement messages), set this to the number of
; seconds to delay.
; ;
; memberdelay = 0 ; memberdelay = 0
; ;

@ -108,12 +108,11 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;notifyringing = yes ; Notify subscriptions on RINGING state ;notifyringing = yes ; Notify subscriptions on RINGING state
; ;
; If regcontext is specified, Asterisk will dynamically ; If regcontext is specified, Asterisk will dynamically create and destroy a
; create and destroy a NoOp priority 1 extension for a given ; NoOp priority 1 extension for a given peer who registers or unregisters with
; peer who registers or unregisters with us. The actual extension ; us. The actual extension is the 'regexten' parameter of the registering
; is the 'regexten' parameter of the registering peer or its ; peer or its name if 'regexten' is not provided. More than one regexten may
; name if 'regexten' is not provided. More than one regexten may be supplied ; be supplied if they are separated by '&'. Patterns may be used in regexten.
; if they are separated by '&'. Patterns may be used in regexten.
; ;
;regcontext=sipregistrations ;regcontext=sipregistrations
; ;
@ -121,12 +120,12 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Format for the register statement is: ; Format for the register statement is:
; register => user[:secret[:authuser]]@host[:port][/extension] ; register => user[:secret[:authuser]]@host[:port][/extension]
; ;
; If no extension is given, the 's' extension is used. The extension ; If no extension is given, the 's' extension is used. The extension needs to
; needs to be defined in extensions.conf to be able to accept calls ; be defined in extensions.conf to be able to accept calls from this SIP proxy
; from this SIP proxy (provider) ; (provider).
; ;
; host is either a host name defined in DNS or the name of a ; host is either a host name defined in DNS or the name of a section defined
; section defined below. ; below.
; ;
; Examples: ; Examples:
; ;
@ -137,9 +136,10 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; ;
;register => 2345:password@sip_proxy/1234 ;register => 2345:password@sip_proxy/1234
; ;
; Register 2345 at sip provider 'sip_proxy'. Calls from this provider connect to local ; Register 2345 at sip provider 'sip_proxy'. Calls from this provider
; extension 1234 in extensions.conf default context, unless you define ; connect to local extension 1234 in extensions.conf, default context,
; unless you configure a [sip_proxy] section below, and configure a 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 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 ; Tip 2: Use separate type=peer and type=user sections for SIP providers
; (instead of type=friend) if you have calls in both directions ; (instead of type=friend) if you have calls in both directions
@ -151,9 +151,9 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Default is 10 tries ; Default is 10 tries
;callevents=no ; generate manager events when sip ua performs events (e.g. hold) ;callevents=no ; generate manager events when sip ua performs events (e.g. hold)
;---------------------------------------------- NAT SUPPORT ------------------------ ;----------------------------------------- NAT SUPPORT ------------------------
; The externip, externhost and localnet settings are used if you use Asterisk behind ; The externip, externhost and localnet settings are used if you use Asterisk
; a NAT device to communicate with services on the outside. ; 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 ;externip = 200.201.202.203 ; Address that we're going to put in outbound SIP messages
; if we're behind a NAT ; 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 ;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 ; The nat= setting is used when Asterisk is on a public IP, communicating with
; devices hidden behind a NAT device (broadband router). ; devices hidden behind a NAT device (broadband router). If you have one-way
; If you have one-way audio problems, you usually have problems with your NAT ; audio problems, you usually have problems with your NAT configuration or your
; configuration or your firewalls support of SIP+RTP ports. ; firewall's support of SIP+RTP ports. You configure Asterisk choice of RTP
; You configure Asterisk choice of RTP ports for incoming audio in rtp.conf ; ports for incoming audio in rtp.conf
; ;
;nat=no ; Global NAT settings (Affects all peers and users) ;nat=no ; Global NAT settings (Affects all peers and users)
; yes = Always ignore info and assume NAT ; 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 ; You may also add auth= statements to [peer] definitions
; Peer auth= override all other authentication settings if we match on realm ; Peer auth= override all other authentication settings if we match on realm
;----------------------------------------------------------------------------------- ;------------------------------------------------------------------------------
; Users and peers have different settings available. Friends have all settings, ; Users and peers have different settings available. Friends have all settings,
; since a friend is both a peer and a user ; 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=alaw
;allow=g723.1 ; Asterisk only supports g723.1 pass-thru! ;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
;allow=g729 ; Pass-thru only unless g729 license obtained ;allow=g729 ; Pass-thru only unless g729 license obtained
;astdb=chan2ext/SIP/grandstream1=1234 ; ensures an astDB entry exists
;[xlite1] ;[xlite1]

@ -10,8 +10,8 @@ serveremail=asterisk
;serveremail=asterisk@linux-support.net ;serveremail=asterisk@linux-support.net
; Should the email contain the voicemail as an attachment ; Should the email contain the voicemail as an attachment
attach=yes attach=yes
; Maximum number of messages per folder. If not specified a default value (100) is used. ; Maximum number of messages per folder. If not specified, a default value
; Maximum value for this option is 9999. ; (100) is used. Maximum value for this option is 9999.
;maxmsg=100 ;maxmsg=100
; Maximum length of a voicemail message in seconds ; Maximum length of a voicemail message in seconds
;maxmessage=180 ;maxmessage=180
@ -28,13 +28,12 @@ maxsilence=10
silencethreshold=128 silencethreshold=128
; Max number of failed login attempts ; Max number of failed login attempts
maxlogins=3 maxlogins=3
; If you need to have an external program, i.e. /usr/bin/myapp ; If you need to have an external program, i.e. /usr/bin/myapp called when a
; called when a voicemail is left, delivered, or your voicemailbox ; voicemail is left, delivered, or your voicemailbox is checked, uncomment
; is checked, uncomment this: ; this:
;externnotify=/usr/bin/myapp ;externnotify=/usr/bin/myapp
; If you need to have an external program, i.e. /usr/bin/myapp ; If you need to have an external program, i.e. /usr/bin/myapp called when a
; called when a voicemail password is changed, ; voicemail password is changed, uncomment this:
; uncomment this:
;externpass=/usr/bin/myapp ;externpass=/usr/bin/myapp
; For the directory, you can override the intro file if you want ; For the directory, you can override the intro file if you want
;directoryintro=dir-intro ;directoryintro=dir-intro
@ -54,13 +53,15 @@ maxlogins=3
;usedirectory=yes ;usedirectory=yes
; ;
; Change the from, body and/or subject, variables: ; 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 ; Note: The emailbody config row can only be up to 512 characters due to a
; asterisk config files. ; limitation in the Asterisk configuration subsystem.
;emailsubject=[PBX]: New message ${VM_MSGNUM} in mailbox ${VM_MAILBOX} ;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 following definition is very close to the default, but the default shows
; the CIDNAME, if it is not null, else just the CIDNUM, or "an unknown caller" if they are both null. ; 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 ;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. ; You can also change the Pager From: string, the pager body and/or subject.
@ -69,7 +70,8 @@ maxlogins=3
;pagersubject=New VM ;pagersubject=New VM
;pagerbody=New ${VM_DUR} long msg in box ${VM_MAILBOX}\nfrom ${VM_CALLERID}, on ${VM_DATE} ;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 ; Default
emaildateformat=%A, %B %d, %Y at %r 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. ; variable substitution is done on the values below.
; ;
; Supported values: ; 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 ; ${VAR} variable substitution
; A or a Day of week (Saturday, Sunday, ...) ; A or a Day of week (Saturday, Sunday, ...)
; B or b or h Month name (January, February, ...) ; 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" ; M Minute, with 00 pronounced as "o'clock"
; N Minute, with 00 pronounced as "hundred" (US military time) ; N Minute, with 00 pronounced as "hundred" (US military time)
; P or p AM or PM ; P or p AM or PM
; Q "today", "yesterday" or ABdY (*note: not standard strftime value) ; Q "today", "yesterday" or ABdY
; q "" (for today), "yesterday", weekday, or ABdY (*note: not standard strftime value) ; (*note: not standard strftime value)
; q "" (for today), "yesterday", weekday, or ABdY
; (*note: not standard strftime value)
; R 24 hour time, including minute ; 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> ; 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 ; 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 ; Advanced options example is extension 4069
; NOTE: All options can be expressed globally in the general section, and overriden in the per-mailbox ; NOTE: All options can be expressed globally in the general section, and
; settings, unless listed otherwise. ; overriden in the per-mailbox settings, unless listed otherwise.
; ;
; tz=central ; Timezone from zonemessages above. Irrelevant if envelope=no. ; tz=central ; Timezone from zonemessages above. Irrelevant if envelope=no.
; attach=yes ; Attach the voicemail to the notification email *NOT* the pager email ; attach=yes ; Attach the voicemail to the notification email *NOT* the pager email

@ -1,17 +1,28 @@
;
; V6PCI/V12PCI config file for VoiceTronix Hardware ; V6PCI/V12PCI config file for VoiceTronix Hardware
; Options ;
; For [general] section ; Options for [general] section
;
; type = v12pci|v6pci|v4pci ; type = v12pci|v6pci|v4pci
; cards = number of cards ; cards = number of cards
; indication = 1 ( To use Asterisk indication tones) ; To use Asterisk indication tones
; ecsuppthres = 0|2048|4096 (none,-24db,-18db only for use with OpenLine4) ; indication = 1
; dtmfidd = 3000 (Inter Digit Delay timeout for when collecting DTMF tones for dialling from a Station port, in ms) ; none,-24db,-18db only for use with OpenLine4
; ast-dtmf-det=1 ( To use Asterisk DTMF detection ) ; ecsuppthres = 0|2048|4096
; relaxdtmf=1 ( Used with ast-dtmf-det ) ; Inter Digit Delay timeout for when collecting DTMF tones for dialling
; break-for-dtmf=no (When a native bridge occurs between 2 vpb channels, it will only break the connection for '#' and '*') ; from a Station port, in ms
; timer_period_ring=4000 (Set the maximum period between received rings, default 4000ms) ; 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, ...) ; board = board_number (1, 2, 3, ...)
; channel = channel_number (1,2,3...) ; channel = channel_number (1,2,3...)
; mode = fxo|immediate|dialtone -- for type of line and line handling ; mode = fxo|immediate|dialtone -- for type of line and line handling

@ -103,9 +103,9 @@ switchtype=national
;privateprefix = +497115678 ;privateprefix = +497115678
;unknownprefix = ;unknownprefix =
; ;
; PRI resetinterval: sets the time in seconds between restart of unused channels, defaults to 3600 ; PRI resetinterval: sets the time in seconds between restart of unused
; minimum 60 seconds ; channels, defaults to 3600; minimum 60 seconds. Some PBXs don't like
; some PBXs don't like channel restarts. so set the interval to a very long interval e.g. 100000000 ; channel restarts. so set the interval to a very long interval e.g. 100000000
; or 'never' to disable *entirely*. ; or 'never' to disable *entirely*.
; ;
;resetinterval = 3600 ;resetinterval = 3600
@ -130,13 +130,13 @@ switchtype=national
; ;
; ISDN Timers ; ISDN Timers
; All of the ISDN timers and counters that are used are configurable. Specify ; All of the ISDN timers and counters that are used are configurable. Specify
; the timer name, and its value (in ms for timers) ; the timer name, and its value (in ms for timers).
; ;
; pritimer => t200,1000 ; pritimer => t200,1000
; pritimer => t313,4000 ; pritimer => t313,4000
; ;
; To enable transmission of facility-based ISDN supplementary services (such ; 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 ; facilityenable = yes
; ;
; ;
@ -145,7 +145,8 @@ switchtype=national
; em_w: E & M Wink ; em_w: E & M Wink
; featd: Feature Group D (The fake, Adtran style, DTMF) ; featd: Feature Group D (The fake, Adtran style, DTMF)
; featdmf: Feature Group D (The real thing, MF (domestic, US)) ; 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 ; featdmf_ta: Feature Group D (The real thing, MF (domestic, US)) through
; a Tandem Access point
; featb: Feature Group B (MF (domestic, US)) ; featb: Feature Group B (MF (domestic, US))
; fxs_ls: FXS (Loop Start) ; fxs_ls: FXS (Loop Start)
; fxs_gs: FXS (Ground Start) ; fxs_gs: FXS (Ground Start)
@ -163,24 +164,31 @@ switchtype=national
; sf_featdmf: SF Feature Group D (The real thing, MF (domestic, US)) ; sf_featdmf: SF Feature Group D (The real thing, MF (domestic, US))
; sf_featb: SF Feature Group B (MF (domestic, US)) ; sf_featb: SF Feature Group B (MF (domestic, US))
; e911: E911 (MF) style signalling ; e911: E911 (MF) style signalling
;
; The following are used for Radio interfaces: ; The following are used for Radio interfaces:
; fxs_rx: Receive audio/COR on an FXS kewlstart interface (FXO at the channel bank) ; fxs_rx: Receive audio/COR on an FXS kewlstart interface (FXO at the
; fxs_tx: Transmit audio/PTT on an FXS loopstart interface (FXO at the channel bank) ; channel bank)
; fxo_rx: Receive audio/COR on an FXO loopstart interface (FXS at the channel bank) ; fxs_tx: Transmit audio/PTT on an FXS loopstart interface (FXO at the
; fxo_tx: Transmit audio/PTT on an FXO groundstart interface (FXS at the channel bank) ; 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_rx: Receive audio/COR on an E&M interface (1-way)
; em_tx: Transmit audio/PTT 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_txrx: Receive audio/COR AND Transmit audio/PTT on an E&M interface
; em_rxtx: same as em_txrx (for our dyslexic friends) ; (2-way)
; em_rxtx: Same as em_txrx (for our dyslexic friends)
; sf_rx: Receive audio/COR on an SF interface (1-way) ; sf_rx: Receive audio/COR on an SF interface (1-way)
; sf_tx: Transmit audio/PTT 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_txrx: Receive audio/COR AND Transmit audio/PTT on an SF interface
; sf_rxtx: same as sf_txrx (for our dyslexic friends) ; (2-way)
; sf_rxtx: Same as sf_txrx (for our dyslexic friends)
; ;
signalling=fxo_ls signalling=fxo_ls
; ;
; For Feature Group D Tandem access, to set the default CIC and OZZ use ; For Feature Group D Tandem access, to set the default CIC and OZZ use these
; these parameters: ; parameters:
;defaultozz=0000 ;defaultozz=0000
;defaultcic=303 ;defaultcic=303
; ;
@ -197,7 +205,8 @@ signalling=fxo_ls
; ;
rxwink=300 ; Atlas seems to use long (250ms) winks 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 ;toneduration=100
; ;
; Whether or not to do distinctive ring detection on FXO lines ; 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 usecallerid=yes
; ;
; Type of caller ID signalling in use ; 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 ;cidsignalling=bell
; ;
; What signals the start of caller ID ; 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 ;cidstart=ring
; ;
@ -227,12 +239,14 @@ hidecallerid=no
; ;
callwaiting=yes 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 ; Mostly use with FXS ports
; ;
;restrictcid=no ;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 usecallingpres=yes
; ;
@ -271,31 +285,29 @@ callreturn=yes
; ;
; Stutter dialtone support: If a mailbox is specified without a voicemail ; Stutter dialtone support: If a mailbox is specified without a voicemail
; context, then when voicemail is received in a mailbox in the default ; context, then when voicemail is received in a mailbox in the default
; voicemail context in voicemail.conf, taking the phone off hook will ; voicemail context in voicemail.conf, taking the phone off hook will cause a
; cause a stutter dialtone instead of a normal one. ; stutter dialtone instead of a normal one.
; ;
; If a mailbox is specified *with* a voicemail context, the same will ; If a mailbox is specified *with* a voicemail context, the same will result
; result if voicemail recieved in mailbox in the specified voicemail ; if voicemail recieved in mailbox in the specified voicemail context.
; context
; ;
; for default voicemail context, the example below is fine: ; for default voicemail context, the example below is fine:
; ;
;mailbox=1234 ;mailbox=1234
; ;
; for any other voicemail context, the following will produce the ; for any other voicemail context, the following will produce the stutter tone:
; stutter tone:
; ;
;mailbox=1234@context ;mailbox=1234@context
; ;
; Enable echo cancellation ; Enable echo cancellation
; Use either "yes", "no", or a power of two from 32 to 256 if you wish ; Use either "yes", "no", or a power of two from 32 to 256 if you wish to
; to actually set the number of taps of cancellation. ; actually set the number of taps of cancellation.
; ;
echocancel=yes echocancel=yes
; ;
; Generally, it is not necessary (and in fact undesirable) to echo cancel ; Generally, it is not necessary (and in fact undesirable) to echo cancel when
; when the circuit path is entirely TDM. You may, however, reverse this ; the circuit path is entirely TDM. You may, however, reverse this behavior
; behavior by enabling the echo cancel during pure TDM bridging below. ; by enabling the echo cancel during pure TDM bridging below.
; ;
echocancelwhenbridged=yes echocancelwhenbridged=yes
; ;
@ -309,10 +321,9 @@ echocancelwhenbridged=yes
;echotraining=yes ;echotraining=yes
;echotraining=800 ;echotraining=800
; ;
; If you are having trouble with DTMF detection, you can relax the ; If you are having trouble with DTMF detection, you can relax the DTMF
; DTMF detection parameters. Relaxing them may make the DTMF detector ; detection parameters. Relaxing them may make the DTMF detector more likely
; more likely to have "talkoff" where DTMF is detected when it ; to have "talkoff" where DTMF is detected when it shouldn't be.
; shouldn't be.
; ;
;relaxdtmf=yes ;relaxdtmf=yes
; ;
@ -321,8 +332,8 @@ echocancelwhenbridged=yes
rxgain=0.0 rxgain=0.0
txgain=0.0 txgain=0.0
; ;
; Logical groups can be assigned to allow outgoing rollover. Groups ; Logical groups can be assigned to allow outgoing rollover. Groups range
; range from 0 to 63, and multiple groups can be specified. ; from 0 to 63, and multiple groups can be specified.
; ;
group=1 group=1
; ;
@ -335,19 +346,18 @@ callgroup=1
pickupgroup=1 pickupgroup=1
; ;
; Specify whether the channel should be answered immediately or ; Specify whether the channel should be answered immediately or if the simple
; if the simple switch should provide dialtone, read digits, etc. ; switch should provide dialtone, read digits, etc.
; ;
immediate=no immediate=no
; ;
; Specify whether flash-hook transfers to 'busy' channels should complete ; Specify whether flash-hook transfers to 'busy' channels should complete or
; or return to the caller performing the transfer (default is yes). ; return to the caller performing the transfer (default is yes).
; ;
;transfertobusy=no ;transfertobusy=no
; ;
; CallerID can be set to "asreceived" or a specific number ; CallerID can be set to "asreceived" or a specific number if you want to
; if you want to override it. Note that "asreceived" only ; override it. Note that "asreceived" only applies to trunk interfaces.
; applies to trunk interfaces.
; ;
;callerid=2564286000 ;callerid=2564286000
; ;
@ -373,39 +383,36 @@ immediate=no
; ;
;busydetect=yes ;busydetect=yes
; ;
; If busydetect is enabled, is also possible to specify how many ; If busydetect is enabled, it is also possible to specify how many busy tones
; busy tones to wait for before hanging up. The default is 4, but ; to wait for before hanging up. The default is 4, but better results can be
; better results can be achieved if set to 6 or even 8. Mind that ; achieved if set to 6 or even 8. Mind that the higher the number, the more
; higher the number, more time is needed to hangup a channel, but ; time that will be needed to hangup a channel, but lowers the probability
; lower is probability to get random hangups ; that you will get random hangups.
; ;
;busycount=4 ;busycount=4
; ;
; If busydetect is enabled, is also possible to specify the ; If busydetect is enabled, it is also possible to specify the cadence of your
; cadence of your busy signal. In many countries it is 500mec ; busy signal. In many countries, it is 500msec on, 500msec off. Without
; on, 500msec off. ; busypattern specified, we'll accept any regular sound-silence pattern that
; Without busypattern specified, we'll accept any regular ; repeats <busycount> times as a busy signal. If you specify busypattern,
; sound-silence pattern than repeats busycount times as a busy ; then we'll further check the length of the sound (tone) and silence, which
; signal. ; will further reduce the chance of a false positive.
; 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 ;busypattern=500,500
; ;
; NOTE: In the Asterisk Makefile you'll find further options to tweak ; NOTE: In the Asterisk Makefile you'll find further options to tweak the busy
; the busy detector. If your country has a busy tone with the same ; detector. If your country has a busy tone with the same length tone and
; lengh tone and silence (as many countries do), consider defining ; silence (as many countries do), consider defining the
; the -DBUSYDETECT_COMPARE_TONE_AND_SILENCE option. ; -DBUSYDETECT_COMPARE_TONE_AND_SILENCE option.
; ;
; Use a polarity reversal to mark when a outgoing call is answered by the ; Use a polarity reversal to mark when a outgoing call is answered by the
; remote party. ; remote party.
; ;
;answeronpolarityswitch=yes ;answeronpolarityswitch=yes
; ;
; In some countries, a polarity reversal is used to signal the disconnect ; In some countries, a polarity reversal is used to signal the disconnect of a
; of a phone line. If the hanguponpolarityswitch option is selected, the ; phone line. If the hanguponpolarityswitch option is selected, the call will
; call will be considered "hung up" on a polarity reversal ; be considered "hung up" on a polarity reversal.
; ;
;hanguponpolarityswitch=yes ;hanguponpolarityswitch=yes
; ;
@ -415,11 +422,11 @@ immediate=no
; This feature is HIGHLY EXPERIMENTAL and can easily detect false answers, ; 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 ; Few zones are supported at the time of this writing, but may be selected
; be selected with "progzone" ; with "progzone"
; ;
; This feature can also easily detect false hangups. The symptoms of this ; This feature can also easily detect false hangups. The symptoms of this is
; is being disconnected in the middle of a call for no reason. ; being disconnected in the middle of a call for no reason.
; ;
;callprogress=yes ;callprogress=yes
;progzone=us ;progzone=us
@ -446,15 +453,15 @@ immediate=no
; ;
;musiconhold=default ;musiconhold=default
; ;
; PRI channels can have an idle extension and a minunused number. So long ; PRI channels can have an idle extension and a minunused number. So long as
; as at least "minunused" channels are idle, chan_zap will try to call ; at least "minunused" channels are idle, chan_zap will try to call "idledial"
; "idledial" on them, and then dump them into the PBX in the "idleext" ; on them, and then dump them into the PBX in the "idleext" extension (which
; extension (which is of the form exten@context). When channels are needed ; is of the form exten@context). When channels are needed the "idle" calls
; the "idle" calls are disconnected (so long as there are at least "minidle" ; are disconnected (so long as there are at least "minidle" calls still
; calls still running, of course) to make more channels available. The ; running, of course) to make more channels available. The primary use of
; primary use of this is to create a dynamic service, where idle channels ; this is to create a dynamic service, where idle channels are bundled through
; are bundled through multilink PPP, thus more efficiently utilizing ; multilink PPP, thus more efficiently utilizing combined voice/data services
; combined voice/data services than conventional fixed mappings/muxings. ; than conventional fixed mappings/muxings.
; ;
;idledial=6999 ;idledial=6999
;idleext=6999@dialout ;idleext=6999@dialout
@ -465,10 +472,10 @@ immediate=no
; ;
;jitterbuffers=4 ;jitterbuffers=4
; ;
; You can define your own custom ring cadences here. You can define up to ; You can define your own custom ring cadences here. You can define up to 8
; 8 pairs. If the silence is negative, it indicates where the callerid ; pairs. If the silence is negative, it indicates where the callerid spill is
; spill is to be placed. Also, if you define any custom cadences, the ; to be placed. Also, if you define any custom cadences, the default cadences
; default cadences will be turned off. ; will be turned off.
; ;
; Syntax is: cadence=ring,silence[,ring,silence[...]] ; Syntax is: cadence=ring,silence[,ring,silence[...]]
; ;
@ -479,11 +486,11 @@ immediate=no
;cadence=125,125,125,125,125,-4000 ;cadence=125,125,125,125,125,-4000
;cadence=1000,500,2500,-5000 ;cadence=1000,500,2500,-5000
; ;
; Each channel consists of the channel number or range. It ; Each channel consists of the channel number or range. It inherits the
; inherits the parameters that were specified above its declaration ; parameters that were specified above its declaration.
; ;
; For GR-303, CRV's are created like channels except they must start ; For GR-303, CRV's are created like channels except they must start with the
; with the trunk group followed by a colon, e.g.: ; trunk group followed by a colon, e.g.:
; ;
; crv => 1:1 ; crv => 1:1
; crv => 2:1-2,5-8 ; crv => 2:1-2,5-8
@ -506,9 +513,8 @@ immediate=no
;callerid="Main TA 750" <(256) 428-6127> ;callerid="Main TA 750" <(256) 428-6127>
;channel => 44 ;channel => 44
; ;
; For example, maybe we have some other channels ; For example, maybe we have some other channels which start out in a
; which start out in a different context and use ; different context and use E & M signalling instead.
; E & M signalling instead.
; ;
;context=remote ;context=remote
;sigalling=em ;sigalling=em
@ -538,9 +544,9 @@ immediate=no
;callerid="Larry Moe" <(256) 428-6234> ;callerid="Larry Moe" <(256) 428-6234>
;channel => 28 ;channel => 28
; ;
; Sample PRI (CPE) config: Specify the switchtype, the signalling as ; Sample PRI (CPE) config: Specify the switchtype, the signalling as either
; either pri_cpe or pri_net for CPE or Network termination, and generally ; pri_cpe or pri_net for CPE or Network termination, and generally you will
; you will want to create a single "group" for all channels of the PRI. ; want to create a single "group" for all channels of the PRI.
; ;
; switchtype = national ; switchtype = national
; signalling = pri_cpe ; signalling = pri_cpe

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