openpkg-packages/asterisk/asterisk.txt

<file name="asterisk.conf">
;;
;;  asterisk.conf -- Asterisk master configuration
;;

[directories]
astetcdir          = @l_prefix@/etc/asterisk
astmoddir          = @l_prefix@/lib/asterisk/modules
astagidir          = @l_prefix@/share/asterisk/agi-bin
astvarlibdir       = @l_prefix@/share/asterisk
astspooldir        = @l_prefix@/var/asterisk/spool
astrundir          = @l_prefix@/var/asterisk/run
astlogdir          = @l_prefix@/var/asterisk/log
astdbdir           = @l_prefix@/var/asterisk/db

[files]
astctlowner        = @l_rusr@
astctlgroup        = @l_rgrp@
astctlpermissions  = 700
astctl             = asterisk.ctl

[options]
systemname         = openpkg-pbx
runuser            = @l_rusr@
rungroup           = @l_rgrp@
verbose            = 0
alwaysfork         = yes
dumpcore           = no
quiet              = yes
highpriority       = no
initcrypto         = no
nocolor            = yes
execincludes       = no
;timestamp         = yes
;optiondebug       = no
;nofork            = no
;console           = no
;dontwarn          = no

</file>
<file name="modules.conf">
;;
;;  modules.conf -- Asterisk functionality module configuration
;;

[modules]
autoload = yes
noload   = chan_iax2.so
noload   = chan_mgcp.so
noload   = chan_skinny.so
noload   = chan_gtalk.so
noload   = chan_ooh323.so

[global]

</file>
<file name="logger.conf">
;;
;;  logger.conf -- Asterisk logging configuration
;;

[general]
dateformat   = %F %T
queue_log    = no
event_log    = no

[logfiles]
console      = error,warning,notice,verbose
asterisk.log = error,warning,notice ; verbose,debug

</file>
<file name="manager.conf">
;;
;;  manager.conf -- Asterisk internal manager API configuration
;;

[general]
enabled            = yes
webenabled         = no
bindaddr           = 127.0.0.1
port               = 5038
sslenable          = no
sslbindaddr        = 127.0.0.1
sslbindport        = 5039
sslcert            = @l_prefix@/etc/asterisk/asterisk.pem
displayconnects    = yes
allowmultiplelogin = yes
timestampevents    = yes

[asterisk]
secret             = asterisk
deny               = 0.0.0.0/0.0.0.0
permit             = 127.0.0.1/255.0.0.0
read               = system,call,agent,user,config,log,verbose,dtmf,reporting,cdr,dialplan
write              = system,call,agent,user,config,command,reporting

</file>
<file name="http.conf">
;;
;;  http.conf -- Asterisk HTTP Server Interface
;;

[general]
enabled            = no
bindaddr           = 127.0.0.1
bindport           = 8088
sslenable          = no
slbindport         = 8089
sslbindaddr        = 127.0.0.1
sslcert            = @l_prefix@/etc/asterisk/asterisk.pem
enablestatic       = yes
prefix             = asterisk
redirect           = / /asterisk/static/docs/index.html

[post_mappings]
uploads            = @l_prefix@/var/asterisk/spool/uploads/

</file>
<file name="sip.conf">
;;
;;  sip.conf -- Asterisk SIP configuration
;;

[general]
useragent     = OpenPKG Asterisk PBX
realm         = example
bindaddr      = 127.0.0.1
bindport      = 5060
tcpenable     = yes
tcpbindaddr   = 127.0.0.1:5060
tlsenable     = no
tlsbindaddr   = 127.0.0.1:5061
tlscipher     = ALL:!ADH:!EXPORT56:RC4+RSA:+HIGH:+MEDIUM:+LOW:+SSLv2:+EXP:+eNULL
tlscertfile   = asterisk.pem
tlscafile     = asterisk.pem
srvlookup     = yes
useclientcode = yes
allowguest    = yes
canreinvite   = no
nat           = no
disallow      = all
allow         = speex
allow         = g726
allow         = ulaw
allow         = alaw
allow         = gsm
videosupport  = no
;allow        = h263
;allow        = h263p
context       = external
;register     = NNNNNNN:XXXXXX:NNNNNNN@sipgate.de/s
;tos          = 0x18

;[sipgate]
;type          = peer
;defaultuser   = NNNNNNN
;host          = sipgate.de
;fromuser      = NNNNNNN
;fromdomain    = sipgate.de
;canreinvite   = no
;disallow      = all
;allow         = speex
;allow         = g726
;allow         = ulaw
;allow         = alaw
;allow         = gsm
;context       = external

;[gw]
;type          = friend
;defaultuser   = gw
;callerid      = "ISDN-to-SIP" <gw>
;fromdomain    = example.com
;secret        = asterisk
;host          = dynamic
;canreinvite   = no
;disallow      = all
;allow         = g726
;allow         = ulaw
;allow         = alaw
;allow         = gsm
;dtmfmode      = rfc2833
;qualify       = yes
;insecure      = yes
;context       = external
;nat           = no

[foo]
type          = friend
defaultuser   = foo
callerid      = "Mr. Foo" <foo>
fromdomain    = example.com
secret        = asterisk
host          = dynamic
disallow      = all
allow         = speex
allow         = g726
allow         = ulaw
allow         = alaw
dtmfmode      = rfc2833
qualify       = yes
context       = internal
nat           = no

[bar]
type          = friend
defaultuser   = bar
callerid      = "Mr. Bar" <bar>
fromdomain    = example.com
secret        = asterisk
host          = dynamic
disallow      = all
allow         = speex
allow         = g726
allow         = ulaw
allow         = alaw
dtmfmode      = rfc2833
qualify       = yes
context       = internal
nat           = no

</file>
<file name="rtp.conf">
;;
;;  rtp.conf -- Asterisk RTP configuration
;;

[general]
rtpstart      = 7070
rtpend        = 7089

</file>
<file name="extensions.conf">
;;
;;  extensions.conf -- Asterisk inbound & outbound call configuration
;;

[general]
static          = yes
writeprotect    = yes
autofallthrough = yes

[globals]
MEETME_SPOOLDIR = @l_prefix@/var/asterisk/spool/meetme
STAFF           = SIP/foo&SIP/bar
CONSOLE         = Console/dsp
DOLLAR          = $

;;
;;  SPECIAL CONTEXTS
;;

[macro-dial]
exten           = s,1,Dial(${ARG1},${ARG2},j${ARG3})
exten           = s,n,Goto(s-${DIALSTATUS},1)
exten           = s-BUSY,1,Voicemail(u${ARG1})
exten           = s-BUSY,2,Busy
exten           = s-CONGESTION,1,Busy
exten           = s-CANCEL,1,Busy
exten           = s-ANSWER,1,Hangup
exten           = s-NOANSWER,1,Hangup
exten           = s-CHANUNAVAIL,1,Hangup
exten           = _s-.,1,Goto(s-NOANSWER,1)

[default]
;   currently empty

;;
;;  EXTERNAL DIAL CONTEXT
;;

[external]
include         = default

;   external incoming SIP connection
exten           = example,hint,${STAFF}
exten           = example,1,Goto(s,1)
exten           = s,n,Ringing
exten           = s,n,Wait(1)
exten           = s,n,Answer
exten           = s,n,Macro(dial,${STAFF},30,gTtr)

;   external to internal mapping
exten           = foo,hint,SIP/foo
exten           = foo,1,Goto(internal,foo,1)
exten           = bar,hint,SIP/bar
exten           = bar,1,Goto(internal,bar,1)

;;
;;  INTERNAL DIAL CONTEXT
;;

[internal]
include         = default
;include        = parkedcalls

;   internal to external mapping
exten           = example,1,Goto(external,example,1)

;   internal user <foo> #11
exten           = foo,hint,SIP/foo
exten           = foo,1,Goto(11,1)
exten           = 11,hint,SIP/foo
exten           = 11,1,Macro(dial,SIP/foo,30,gTtr)

;   internal user <bar> #12
exten           = bar,hint,SIP/bar
exten           = bar,1,Goto(12,1)
exten           = 12,hint,SIP/bar
exten           = 12,1,Macro(dial,SIP/bar,30,gTtr)

;   internal group <all> #20
exten           = all,1,Goto(20,1)
exten           = 20/foo,1,Macro(dial,SIP/bar,60)
exten           = 20/bar,1,Macro(dial,SIP/foo,60)

;   internal service <conference> #7<n>
exten           = conference,1,Goto(70,1)
exten           = _7[0-9],1,Set(confno=${EXTEN:1})
exten           = _7[0-9],n,Goto(7,enter)
exten           = 7,1,Set(TIMEOUT(digit)=3)
exten           = 7,n,Set(TIMEOUT(response)=6)
exten           = 7,n(repeat),Read(confno,conf-getconfno,3)
exten           = 7,n,GotoIf($[${confno} >= 0 & ${confno} <= 9]?enter)
exten           = 7,n,Playback(conf-invalid)
exten           = 7,n,Goto(repeat)
exten           = 7,n(enter),Playback(conf-placeintoconf)
exten           = 7,n,SayNumber(${confno})
exten           = 7,n,Set(SPYGROUP=conference-${confno})
exten           = 7,n,Set(confopt=cCpsMvio)
exten           = 7,n,GotoIf($[${confno} >= 4 & ${confno} <= 9]?l1:l2)
exten           = 7,n(l1),Set(confopt=${confopt}i)
exten           = 7,n(l2),GotoIf($[${confno} >= 7 & ${confno} <= 9]?l3:l4)
exten           = 7,n(l3),Set(confopt=${confopt}r)
exten           = 7,n,Set(MEETME_RECORDINGFILE=${MEETME_SPOOLDIR}/meetme-conference-${confno}-${STRFTIME(${EPOCH},UTC,%Y%m%d%H%M)})
exten           = 7,n,Set(MEETME_RECORDINGFORMAT=wav49)
exten           = 7,n,Playback(this-call-may-be-monitored-or-recorded)
exten           = 7,n(l4),MeetMe(${confno},${confopt})
exten           = 7,n,Playback(beep)
exten           = 7,n,Wait(1)
exten           = 7,n,Playback(vm-goodbye)
exten           = 7,n,Hangup

;   internal service <voicemail> #80/#*<n>
exten           = voicemail,1,Goto(80,1)
exten           = 80,1,VoicemailMain(s${CALLERIDNUM})
exten           = 80,n,Hangup
exten           = _*XX,1,Voicemail(u${EXTEN:1})
exten           = _*XX,n,Hangup

;   internal service <echo> #81
exten           = echo,1,Goto(81,1)
exten           = 81,1,Answer
exten           = 81,n,Playback(demo-echotest)
exten           = 81,n,Wait(1)
exten           = 81,n,Playback(beep)
exten           = 81,n,Echo
exten           = 81,n,Wait(1)
exten           = 81,n,Playback(demo-echodone)
exten           = 81,n,Wait(1)
exten           = 81,n,Playback(vm-goodbye)
exten           = 81,n,Hangup

;   internal service <reload> #82
exten           = reload,1,Goto(82,1)
exten           = 82,1,Answer
exten           = 82,n,Read(pin,conf-getpin,4)
exten           = 82,n,GotoIf($[${pin} = 1234]?ok)
exten           = 82,n,Playback(conf-invalidpin)
exten           = 82,n,Hangup
exten           = 82,n(ok),Playback(beep)
exten           = 82,n,Wait(1)
exten           = 82,n,Playback(beep)
exten           = 82,n,Wait(1)
exten           = 82,n,Playback(beep)
exten           = 82,n,Wait(1)
exten           = 82,n,System(@l_prefix@/sbin/asterisk -rx reload)
exten           = 82,n,Hangup

;   external outgoing ISDN (via SIP-to-ISDN gateway call-through)
;exten          = _0.,1,Set(number=${EXTEN:1})
;exten          = _0.,n,Set(enum=${ENUMLOOKUP(+${number},ALL)})
;exten          = _0.,n,Set(enum_is_sip_url=${REGEX("^SIP/.+" ${enum})})
;exten          = _0.,n,GotoIf($["${enum_is_sip_url}" = "1"]?sip:isdn)
;exten          = _0.,n(sip),Dial(${enum},60,o)
;exten          = _0.,n,Goto(_0.,7)
;exten          = _0.,n(isdn),Dial(SIP/gw,60,D(w1234w0#31#${number}#))
;exten          = _0.,n,Hangup

;   internal outgoing SIP call (part 1/2)
;   (notice sort-order trickery!)
include         = internal-siponly

[internal-siponly]
;   internal outgoing SIP call (part 2/2)
;   (notice sort-order trickery!)
exten           = _.[@].,1,Dial(SIP/${EXTEN}@${SIPDOMAIN},60,o)
exten           = _.[@].,n,Hangup
exten           = _.[@].,102,Busy

</file>
<file name="enum.conf">
;;
;;  enum.conf -- Asterisk ENUM configuration
;;

[general]
search   = e164.arpa
search   = e164.org

</file>
<file name="musiconhold.conf">
;;
;;  musiconhold.conf -- Asterisk music-on-hold configuration
;;

[default]
mode             = files
directory        = @l_prefix@/share/asterisk/moh

</file>
<file name="voicemail.conf">
;;
;;  voicemail.conf -- Asterisk voice mail configuration
;;

[general]
format           = wav49
serveremail      = example@example.com
attach           = yes
maxmsg           = 20
maxsecs          = 180
minsecs          = 3
maxgreet         = 60
skipms           = 3000
maxsilence       = 10
silencethreshold = 128
maxlogins        = 3
charset          = ISO-8859-1
pbxskip          = yes
fromstring       = Asterisk PBX
usedirectory     = yes
emailsubject     = [PBX]: New voice message ${VM_MSGNUM} in mailbox ${VM_MAILBOX}
emailbody        = Dear ${VM_NAME},\n\njust wanted to let you know you were left a ${VM_DUR} long\nvoice message (number ${VM_MSGNUM}) in voice mailbox ${VM_MAILBOX}\nfrom caller ${VM_CALLERID},\non ${VM_DATE}.\nYou might want to check it when you get a chance. Thanks!\n\n\t\t\t\t-- OpenPKG Asterisk PBX\n
pagerfromstring  = Asterisk PBX
pagersubject     = New VM
pagerbody        = New ${VM_DUR} long msg in box ${VM_MAILBOX}\nfrom ${VM_CALLERID}, on ${VM_DATE}
emaildateformat  = %A, %d %B %Y %H:%M:%S %r
mailcmd          = @l_prefix@/sbin/sendmail -t

[default]
1                = 1,Example,example@example.com,,|delete=yes

</file>
<file name="cdr.conf">
;;
;;  cdr.conf -- Asterisk Call Detail Record (CDR) configuration
;;

[general]
enable             = yes
unanswered         = no
batch              = no
size               = 100
time               = 300
scheduleronly      = no
safeshutdown       = yes
endbeforehexten    = yes

</file>
<file name="cdr_custom.conf">
;;
;;  cdr_custom.conf -- Asterisk Call Detail Record (CDR) via Comma Separated Value (CSV) format configuration
;;

[mappings]
master.csv         = "${CDR(start)}", "${CDR(answer)}", "${CDR(end)}", "${CDR(duration)}", "${CDR(billsec)}", "${CDR(clid)}", "${CDR(src)}", "${CDR(dst)}", "${CDR(dcontext)}", "${CDR(channel)}", "${CDR(dstchannel)}", "${CDR(lastapp)}", "${CDR(lastdata)}", "${CDR(disposition)}", "${CDR(amaflags)}", "${CDR(accountcode)}", "${CDR(uniqueid)}", "${CDR(userfield)}"

</file>
<file name="cdr_sqlite3_custom.conf">
;;
;;  cdr_sqlite3_custom.conf -- Asterisk Call Detail Record (CDR) via SQLite RDBMS format configuration
;;

[master]
table              = cdr
columns            = start, answer, end, duration, billsec, clid, src, dst, dcontext, channel, dstchannel, lastapp, lastdata, disposition, amaflags, accountcode, uniqueid, userfield
values             = "${CDR(start)}", "${CDR(answer)}", "${CDR(end)}", "${CDR(duration)}", "${CDR(billsec)}", "${CDR(clid)}", "${CDR(src)}", "${CDR(dst)}", "${CDR(dcontext)}", "${CDR(channel)}", "${CDR(dstchannel)}", "${CDR(lastapp)}", "${CDR(lastdata)}", "${CDR(disposition)}", "${CDR(amaflags)}", "${CDR(accountcode)}", "${CDR(uniqueid)}", "${CDR(userfield)}"

</file>
<file name="cdr_manager.conf">
;;
;;  cdr_manager.conf -- Asterisk Call Detail Record (CDR) via Asterisk Manager Interface (AMI) configuration
;;

[general]
enabled            = yes

</file>
<file name="meetme.conf">
;;
;;  meetme.conf -- Asterisk conference configuration
;;

[general]
audiobuffers     = 32
;schedule        = yes
;logmembercount  = yes
;fuzzystart      = 300
;earlyalert      = 3600
;endalert        = 120

[rooms]
conf             = 0
conf             = 1
conf             = 2
conf             = 3
conf             = 4
conf             = 5
conf             = 6
conf             = 7
conf             = 8
conf             = 9,1234,1234

</file>
<file name="codecs.conf">
;;
;;  codecs.conf -- Asterisk codec configuration
;;

[speex]
quality            = 6
complexity         = 4
enhancement        = true
vad                = true
vbr                = true
abr                = 8000
vbr_quality        = 5
dtx                = false
preprocess         = false
pp_vad             = false
pp_agc             = false
pp_agc_level       = 8000
pp_denoise         = false
pp_dereverb        = false
pp_dereverb_decay  = 0.4
pp_dereverb_level  = 0.3

[plc]
genericplc         = true

</file>
<file name="zapata.conf">
;;
;;  zapata.conf -- Asterisk Zaptel channel configuration
;;

;   (an empty configuration is ok, but required even for Zaptel "ztdummy" only)
[trunkgroups]
[channels]

</file>
<file name="capi.conf">
;;
;;  capi.conf -- Asterisk ISDN/CAPI channel configuration
;;

[general]
nationalprefix        = 0
internationalprefix   = 00
rxgain                = 1.0
txgain                = 1.0
ulaw                  = no
debug                 = yes

[ISDN1]
isdnmode              = msn
incomingmsn           = *
controller            = 0
group                 = 1
;prefix               = 0
softdtmf              = off
relaxdtmf             = off
accountcode           =
context               = external
holdtype              = local
;immediate            = yes
echocancel            = no
echosquelch           = no
;echotail             = 64
;bridge               = yes
;callgroup            = 1
;deflect              = 1234567
devices               = 2
;wait_silence_samples = 1000
;dtmf_generate        = yes

</file>
<file name="features.conf">
;;
;;  features.conf -- Asterisk Call Features configuration
;;

[general]
;parkext              = 700
;parkpos              = 701-720
;context              = parkedcalls

</file>
<file name="jabber.conf">
;;
;;  jabber.conf -- Asterisk Jabber configuration
;;

[general]
;debug                = yes
;autoprune            = yes
;autoregister         = yes

;[asterisk]
;type                 = client
;serverhost           = jabber.example.com
;username             = asterisk@example.com/asterisk
;secret               = asterisk
;priority             = 1
;port                 = 5222
;usetls               = no
;usesasl              = no
;buddy                = buddy@example.com
;status               = available
;timeout              = 100

</file>
<file name="indications.conf">
;;
;;  indications.conf -- Asterisk Tone Indications
;;

[general]
country      = us

;   United States
;   (according to tones in North America)
[us]
description  = United States (US)
ringcadence  = 2000,4000
dial         = 350+440
busy         = 480+620/500,0/500
ring         = 440+480/2000,0/4000
congestion   = 480+620/250,0/250
callwaiting  = 440/300,0/10000
dialrecall   = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
record       = 1400/500,0/15000
info         = !950/330,!1400/330,!1800/330,0
stutter      = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440

;   United Kingdom
;   (partly according to BT SIN350)
[uk]
description  = United Kingdom (UK)
ringcadence  = 400,200,400,2000
dial         = 350+440
busy         = 400/375,0/375
ring         = 400+450/400,0/200,400+450/400,0/2000
congestion   = 400/400,0/350,400/225,0/525
callwaiting  = 400/100,0/4000
dialrecall   = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
record       = 1400/500,0/60000
info         = 950/330,0/15,1400/330,0/15,1800/330,0/1000
stutter      = 350+440/750,440/750

;   Germany
;   (according to http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf)
[de]
description  = Germany (DE)
ringcadence  = 1000,4000
dial         = 425
busy         = 425/480,0/480
ring         = 425/1000,0/4000
congestion   = 425/240,0/240
callwaiting  = !425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,0
dialrecall   = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
record       = 1400/80,0/15000
info         = 950/330,1400/330,1800/330,0/1000
stutter      = 425+400

</file>
<file name="res_odbc.conf">
;;
;;  res_odbc.conf -- Asterisk ODBC resource configuration
;;

[ENV]

[asterisk-sqlite]
enabled             = no
dsn                 = asterisk-sqlite
username            =
password            =
pre-connect         = no
sanitysql           = SELECT 1
;idlecheck          = 3600
backslash_is_escape = yes
share_connections   = yes
limit               = 10

</file>
<file name="func_odbc.conf">
;;
;;  func_odbc.conf -- Asterisk ODBC dialplan function configuration
;;

;   Set(<variable_name>=${ODBC_SQL(SELECT [...])})
;   Set(ODBC_SQL(UPDATE [...]))
[SQL]
prefix              = ODBC
dsn                 = asterisk-sqlite
readsql             = ${ARG1}
writesql            = ${ARG1}

;   Set(<variable_name>=${ODBC_MAP(<key>)})
;   Set(ODBC_MAP(<key>)=<value>)
[MAP]
prefix              = ODBC
dsn                 = asterisk-sqlite
readsql             = SELECT val FROM map WHERE key='${SQL_ESC(${ARG1})}'
writesql            = UPDATE map SET val='${SQL_ESC(${VAL1})}' WHERE key='${SQL_ESC(${ARG1})}'
escapecommas        = no

</file>
<file name="asterisk.pem">
-----BEGIN CERTIFICATE-----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-----END CERTIFICATE-----
-----BEGIN RSA PRIVATE KEY-----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-----END RSA PRIVATE KEY-----
</file>