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543 lines
14 KiB
543 lines
14 KiB
<file name="asterisk.conf"> |
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;; |
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;; asterisk.conf -- Asterisk master configuration |
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;; |
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[directories] |
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astetcdir = @l_prefix@/etc/asterisk |
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astmoddir = @l_prefix@/lib/asterisk/modules |
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astagidir = @l_prefix@/share/asterisk/agi-bin |
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astvarlibdir = @l_prefix@/share/asterisk |
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astspooldir = @l_prefix@/var/asterisk/spool |
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astrundir = @l_prefix@/var/asterisk/run |
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astlogdir = @l_prefix@/var/asterisk/log |
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astdbdir = @l_prefix@/var/asterisk/db |
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[files] |
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astctlowner = @l_rusr@ |
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astctlgroup = @l_rgrp@ |
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astctlpermissions = 700 |
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astctl = asterisk.ctl |
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[options] |
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systemname = openpkg-pbx |
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runuser = @l_rusr@ |
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rungroup = @l_rgrp@ |
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verbose = 0 |
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alwaysfork = yes |
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dumpcore = no |
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quiet = yes |
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highpriority = no |
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initcrypto = no |
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nocolor = yes |
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execincludes = no |
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;timestamp = yes |
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;optiondebug = no |
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;nofork = no |
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;console = no |
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;dontwarn = no |
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</file> |
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<file name="modules.conf"> |
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;; |
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;; modules.conf -- Asterisk functionality module configuration |
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;; |
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[modules] |
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autoload = yes |
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noload = chan_iax2.so ; not yet wished |
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noload = chan_agent.so ; not yet wished |
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noload = chan_mgcp.so ; not yet wished |
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noload = chan_skinny.so ; not yet wished |
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noload = pbx_dundi.so ; not yet wished |
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noload = app_queue.so ; not yet wished |
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noload = cdr_custom.so ; not yet wished |
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noload = pbx_ael.so ; not yet wished |
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noload = res_phoneprov.so ; not yet wished |
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noload = res_smdi.so ; not yet wished |
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noload = app_meetme.so ; not yet wished |
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noload = chan_ooh323.so ; not yet wished |
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load = app_conference.so ; wished |
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load = res_musiconhold.so ; wished |
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[global] |
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</file> |
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<file name="logger.conf"> |
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;; |
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;; logger.conf -- Asterisk logging configuration |
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;; |
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[general] |
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dateformat = %F %T |
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queue_log = no |
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event_log = no |
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[logfiles] |
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console = error,warning,notice,verbose |
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asterisk.log = error,warning,notice ; verbose,debug |
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</file> |
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<file name="manager.conf"> |
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;; |
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;; manager.conf -- Asterisk internal manager API configuration |
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;; |
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[general] |
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enabled = no |
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port = 5038 |
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bindaddr = 10.10.0.1 |
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displayconnects = yes |
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[asterisk] |
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secret = asterisk |
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deny = 0.0.0.0/0.0.0.0 |
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permit = 10.10.0.0/255.255.0.0 |
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read = system,call,log,verbose,command,agent,user |
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write = system,call,log,verbose,command,agent,user |
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</file> |
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<file name="sip.conf"> |
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;; |
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;; sip.conf -- Asterisk SIP configuration |
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;; |
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[general] |
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useragent = OpenPKG Asterisk PBX |
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realm = example |
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bindport = 5060 |
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bindaddr = 127.0.0.1 |
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srvlookup = yes |
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useclientcode = yes |
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allowguest = yes |
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canreinvite = no |
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disallow = all |
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allow = speex |
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allow = g726 |
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allow = ulaw |
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allow = alaw |
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allow = gsm |
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videosupport = no |
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;allow = h263 |
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;allow = h263p |
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context = external |
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;register = NNNNNNN:XXXXXX:NNNNNNN@sipgate.de/s |
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;tos = 0x18 |
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;[sipgate] |
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;type = peer |
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;username = NNNNNNN |
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;host = sipgate.de |
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;fromuser = NNNNNNN |
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;fromdomain = sipgate.de |
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;canreinvite = no |
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;disallow = all |
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;allow = speex |
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;allow = g726 |
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;allow = ulaw |
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;allow = alaw |
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;allow = gsm |
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;context = external |
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;[gw] |
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;type = friend |
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;username = gw |
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;callerid = "ISDN-to-SIP" <gw> |
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;fromdomain = example.com |
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;secret = asterisk |
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;host = dynamic |
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;canreinvite = no |
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;disallow = all |
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;allow = g726 |
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;allow = ulaw |
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;allow = alaw |
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;allow = gsm |
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;dtmfmode = rfc2833 |
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;qualify = yes |
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;insecure = yes |
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;context = external |
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[foo] |
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type = friend |
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username = foo |
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callerid = "Mr. Foo" <foo> |
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fromdomain = example.com |
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secret = asterisk |
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host = dynamic |
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disallow = all |
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allow = speex |
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allow = g726 |
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allow = ulaw |
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allow = alaw |
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dtmfmode = rfc2833 |
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qualify = yes |
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context = internal |
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[bar] |
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type = friend |
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username = bar |
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callerid = "Mr. Bar" <bar> |
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fromdomain = example.com |
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secret = asterisk |
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host = dynamic |
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disallow = all |
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allow = speex |
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allow = g726 |
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allow = ulaw |
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allow = alaw |
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dtmfmode = rfc2833 |
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qualify = yes |
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context = internal |
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</file> |
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<file name="rtp.conf"> |
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;; |
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;; rtp.conf -- Asterisk RTP configuration |
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;; |
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[general] |
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rtpstart = 7070 |
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rtpend = 7089 |
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</file> |
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<file name="extensions.conf"> |
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;; |
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;; extensions.conf -- Asterisk inbound & outbound call configuration |
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;; |
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[general] |
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static = yes |
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writeprotect = yes |
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autofallthrough = yes |
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[globals] |
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MEETME_SPOOLDIR = @l_prefix@/var/asterisk/spool/meetme |
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STAFF = SIP/foo&SIP/bar |
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CONSOLE = Console/dsp |
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DOLLAR = $ |
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;; |
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;; SPECIAL CONTEXTS |
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;; |
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[macro-dial] |
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exten = s,1,Dial(${ARG1},${ARG2},j${ARG3}) |
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exten = s,n,Goto(s-${DIALSTATUS},1) |
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exten = s-BUSY,1,Voicemail(u${ARG1}) |
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exten = s-BUSY,2,Busy |
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exten = s-CONGESTION,1,Busy |
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exten = s-CANCEL,1,Busy |
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exten = s-ANSWER,1,Hangup |
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exten = s-NOANSWER,1,Hangup |
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exten = s-CHANUNAVAIL,1,Hangup |
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exten = _s-.,1,Goto(s-NOANSWER,1) |
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[default] |
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; currently empty |
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;; |
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;; EXTERNAL DIAL CONTEXT |
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;; |
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[external] |
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include = default |
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; external incoming SIP connection |
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exten = example,hint,${STAFF} |
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exten = example,1,Goto(s,1) |
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exten = s,n,Ringing |
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exten = s,n,Wait(1) |
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exten = s,n,Answer |
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exten = s,n,Macro(dial,${STAFF},30,gTtr) |
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; external to internal mapping |
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exten = foo,hint,SIP/foo |
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exten = foo,1,Goto(internal,foo,1) |
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exten = bar,hint,SIP/bar |
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exten = bar,1,Goto(internal,bar,1) |
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;; |
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;; INTERNAL DIAL CONTEXT |
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;; |
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[internal] |
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include = default |
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;include = parkedcalls |
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; internal to external mapping |
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exten = example,1,Goto(external,example,1) |
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; internal user <foo> #11 |
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exten = foo,hint,SIP/foo |
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exten = foo,1,Goto(11,1) |
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exten = 11,hint,SIP/foo |
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exten = 11,1,Macro(dial,SIP/foo,30,gTtr) |
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; internal user <bar> #12 |
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exten = bar,hint,SIP/bar |
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exten = bar,1,Goto(12,1) |
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exten = 12,hint,SIP/bar |
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exten = 12,1,Macro(dial,SIP/bar,30,gTtr) |
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; internal group <all> #20 |
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exten = all,1,Goto(20,1) |
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exten = 20/foo,1,Macro(dial,SIP/bar,60) |
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exten = 20/bar,1,Macro(dial,SIP/foo,60) |
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; internal service <conference> #7<n> |
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exten = conference,1,Goto(70,1) |
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exten = _7[0-9],1,Set(confno=${EXTEN:1}) |
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exten = _7[0-9],n,Goto(7,enter) |
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exten = 7,1,Set(TIMEOUT(digit)=3) |
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exten = 7,n,Set(TIMEOUT(response)=6) |
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exten = 7,n(repeat),Read(confno,conf-getconfno,3) |
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exten = 7,n,GotoIf($[${confno} >= 0 & ${confno} <= 9]?enter) |
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exten = 7,n,Playback(conf-invalid) |
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exten = 7,n,Goto(repeat) |
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exten = 7,n(enter),Playback(conf-placeintoconf) |
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exten = 7,n,SayNumber(${confno}) |
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exten = 7,n,Set(SPYGROUP=conference-${confno}) |
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exten = 7,n,Set(confopt=cps) |
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exten = 7,n,GotoIf($[${confno} >= 4 & ${confno} <= 9]?l1:l2) |
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exten = 7,n(l1),Set(confopt=${confopt}i) |
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exten = 7,n(l2),GotoIf($[${confno} >= 7 & ${confno} <= 9]?l3:l4) |
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exten = 7,n(l3),Set(confopt=${confopt}r) |
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exten = 7,n,Set(MEETME_RECORDINGFILE=${MEETME_SPOOLDIR}/meetme-conference-${confno}-${TIMESTAMP}) |
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exten = 7,n,Set(MEETME_RECORDINGFORMAT=wav49) |
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exten = 7,n,Playback(this-call-may-be-monitored-or-recorded) |
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exten = 7,n(l4),MeetMe(${confno},${confopt}) |
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exten = 7,n,Playback(vm-goodbye) |
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exten = 7,n,Hangup |
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; internal service <voicemail> #80/#*<n> |
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exten = voicemail,1,Goto(80,1) |
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exten = 80,1,VoicemailMain(s${CALLERIDNUM}) |
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exten = 80,n,Hangup |
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exten = _*XX,1,Voicemail(u${EXTEN:1}) |
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exten = _*XX,n,Hangup |
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; internal service <echo> #81 |
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exten = echo,1,Goto(81,1) |
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exten = 81,1,Answer |
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exten = 81,n,Playback(demo-echotest) |
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exten = 81,n,Echo |
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exten = 81,n,Playback(demo-echodone) |
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exten = 81,n,Hangup |
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; internal service <reload> #82 |
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exten = reload,1,Goto(82,1) |
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exten = 82,1,Answer |
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exten = 82,n,Read(pin,conf-getpin,4) |
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exten = 82,n,GotoIf($[${pin} = 1234]?ok) |
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exten = 82,n,Playback(conf-invalidpin) |
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exten = 82,n,Hangup |
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exten = 82,n(ok),Playback(beep) |
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exten = 82,n,Wait(1) |
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exten = 82,n,Playback(beep) |
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exten = 82,n,Wait(1) |
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exten = 82,n,Playback(beep) |
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exten = 82,n,Wait(1) |
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exten = 82,n,System(@l_prefix@/sbin/asterisk -rx reload) |
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exten = 82,n,Hangup |
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; external outgoing ISDN (via SIP-to-ISDN gateway call-through) |
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;exten = _0.,1,Set(number=${EXTEN:1}) |
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;exten = _0.,n,Set(enum=${ENUMLOOKUP(+${number},ALL)}) |
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;exten = _0.,n,Set(enum_is_sip_url=${REGEX("^SIP/.+" ${enum})}) |
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;exten = _0.,n,GotoIf($["${enum_is_sip_url}" = "1"]?sip:isdn) |
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;exten = _0.,n(sip),Dial(${enum},60,o) |
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;exten = _0.,n,Goto(_0.,7) |
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;exten = _0.,n(isdn),Dial(SIP/gw,60,D(w1234w0#31#${number}#)) |
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;exten = _0.,n,Hangup |
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; internal outgoing SIP call (part 1/2) |
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; (notice sort-order trickery!) |
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include = internal-siponly |
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[internal-siponly] |
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; internal outgoing SIP call (part 2/2) |
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; (notice sort-order trickery!) |
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exten = _.[@].,1,Dial(SIP/${EXTEN}@${SIPDOMAIN},60,o) |
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exten = _.[@].,n,Hangup |
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exten = _.[@].,102,Busy |
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</file> |
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<file name="enum.conf"> |
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;; |
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;; enum.conf -- Asterisk ENUM configuration |
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;; |
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[general] |
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search = e164.arpa |
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search = e164.org |
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</file> |
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<file name="musiconhold.conf"> |
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;; |
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;; musiconhold.conf -- Asterisk music-on-hold configuration |
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;; |
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[default] |
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mode = files |
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directory = @l_prefix@/share/asterisk/moh |
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</file> |
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<file name="voicemail.conf"> |
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;; |
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;; voicemail.conf -- Asterisk voice mail configuration |
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;; |
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[general] |
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format = wav49 |
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serveremail = example@example.com |
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attach = yes |
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maxmsg = 20 |
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maxmessage = 180 |
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minmessage = 3 |
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maxgreet = 60 |
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skipms = 3000 |
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maxsilence = 10 |
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silencethreshold = 128 |
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maxlogins = 3 |
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charset = ISO-8859-1 |
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pbxskip = yes |
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fromstring = Asterisk PBX |
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usedirectory = yes |
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emailsubject = [PBX]: New voice message ${VM_MSGNUM} in mailbox ${VM_MAILBOX} |
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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 |
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pagerfromstring = Asterisk PBX |
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pagersubject = New VM |
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pagerbody = New ${VM_DUR} long msg in box ${VM_MAILBOX}\nfrom ${VM_CALLERID}, on ${VM_DATE} |
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emaildateformat = %A, %d %B %Y %H:%M:%S %r |
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mailcmd = @l_prefix@/sbin/sendmail -t |
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[default] |
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1 = 1,Example,example@example.com,,|delete=yes |
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</file> |
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<file name="meetme.conf"> |
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;; |
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;; meetme.conf -- Asterisk conference configuration |
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;; |
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[general] |
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audiobuffers = 32 |
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[rooms] |
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conf = 0 |
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conf = 1 |
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conf = 2 |
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conf = 3 |
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conf = 4 |
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conf = 5 |
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conf = 6 |
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conf = 7 |
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conf = 8 |
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conf = 9,1234,1234 |
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</file> |
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<file name="codecs.conf"> |
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;; |
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;; codecs.conf -- Asterisk codec configuration |
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;; |
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[speex] |
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quality = 6 |
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complexity = 4 |
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enhancement = true |
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vad = true |
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vbr = true |
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abr = 8000 |
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vbr_quality = 5 |
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dtx = false |
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preprocess = false |
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pp_vad = false |
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pp_agc = false |
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pp_agc_level = 8000 |
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pp_denoise = false |
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pp_dereverb = false |
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pp_dereverb_decay = 0.4 |
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pp_dereverb_level = 0.3 |
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[plc] |
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genericplc = true |
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</file> |
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<file name="zapata.conf"> |
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;; |
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;; zapata.conf -- Asterisk Zap channel configuration |
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;; |
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; (an empty configuration is ok, but required even for dummy Zaptel support) |
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</file> |
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<file name="capi.conf"> |
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;; |
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;; capi.conf -- Asterisk ISDN/CAPI channel configuration |
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;; |
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[general] |
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nationalprefix = 0 |
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internationalprefix = 00 |
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rxgain = 1.0 |
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txgain = 1.0 |
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ulaw = no |
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debug = yes |
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[ISDN1] |
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isdnmode = msn |
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incomingmsn = * |
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controller = 0 |
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group = 1 |
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;prefix = 0 |
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softdtmf = off |
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relaxdtmf = off |
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accountcode = |
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context = external |
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holdtype = local |
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;immediate = yes |
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echocancel = no |
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echosquelch = no |
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;echotail = 64 |
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;bridge = yes |
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;callgroup = 1 |
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;deflect = 1234567 |
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devices = 2 |
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;wait_silence_samples = 1000 |
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;dtmf_generate = yes |
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</file> |
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<file name="features.conf"> |
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;; |
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;; features.conf -- Asterisk Call Features configuration |
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;; |
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[general] |
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;parkext = 700 |
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;parkpos = 701-720 |
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;context = parkedcalls |
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</file> |
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<file name="jabber.conf"> |
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;; |
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;; jabber.conf -- Asterisk Jabber configuration |
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;; |
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[general] |
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;debug = yes |
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;autoprune = yes |
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;autoregister = yes |
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;[asterisk] |
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;type = client |
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;serverhost = jabber.example.com |
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;username = asterisk@example.com/asterisk |
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;secret = asterisk |
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;priority = 1 |
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;port = 5222 |
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;usetls = no |
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;usesasl = no |
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;buddy = buddy@example.com |
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;status = available |
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;timeout = 100 |
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</file>
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