From 503ebba846b30a4d9e493a7e8578f9c40cf89bf4 Mon Sep 17 00:00:00 2001 From: lmadsen Date: Wed, 21 Jan 2009 13:19:16 +0000 Subject: Creating tag for asterisk-1.4.23 (in the right location this time too!) git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.23@169668 f38db490-d61c-443f-a65b-d21fe96a405b --- 1.4.23-rc4/configs/adsi.conf.sample | 8 + 1.4.23-rc4/configs/adtranvofr.conf.sample | 39 ++ 1.4.23-rc4/configs/agents.conf.sample | 107 ++++ 1.4.23-rc4/configs/alarmreceiver.conf.sample | 80 +++ 1.4.23-rc4/configs/alsa.conf.sample | 62 +++ 1.4.23-rc4/configs/amd.conf.sample | 18 + 1.4.23-rc4/configs/asterisk.adsi | 159 ++++++ 1.4.23-rc4/configs/cdr.conf.sample | 148 ++++++ 1.4.23-rc4/configs/cdr_custom.conf.sample | 10 + 1.4.23-rc4/configs/cdr_manager.conf.sample | 6 + 1.4.23-rc4/configs/cdr_odbc.conf.sample | 12 + 1.4.23-rc4/configs/cdr_pgsql.conf.sample | 9 + 1.4.23-rc4/configs/cdr_tds.conf.sample | 11 + 1.4.23-rc4/configs/chan_dahdi.conf.sample | 675 ++++++++++++++++++++++++ 1.4.23-rc4/configs/codecs.conf.sample | 65 +++ 1.4.23-rc4/configs/dnsmgr.conf.sample | 5 + 1.4.23-rc4/configs/dundi.conf.sample | 239 +++++++++ 1.4.23-rc4/configs/enum.conf.sample | 22 + 1.4.23-rc4/configs/extconfig.conf.sample | 59 +++ 1.4.23-rc4/configs/extensions.ael.sample | 448 ++++++++++++++++ 1.4.23-rc4/configs/extensions.conf.sample | 614 ++++++++++++++++++++++ 1.4.23-rc4/configs/features.conf.sample | 100 ++++ 1.4.23-rc4/configs/festival.conf.sample | 35 ++ 1.4.23-rc4/configs/followme.conf.sample | 86 ++++ 1.4.23-rc4/configs/func_odbc.conf.sample | 41 ++ 1.4.23-rc4/configs/gtalk.conf.sample | 19 + 1.4.23-rc4/configs/h323.conf.sample | 193 +++++++ 1.4.23-rc4/configs/http.conf.sample | 40 ++ 1.4.23-rc4/configs/iax.conf.sample | 416 +++++++++++++++ 1.4.23-rc4/configs/iaxprov.conf.sample | 81 +++ 1.4.23-rc4/configs/indications.conf.sample | 733 +++++++++++++++++++++++++++ 1.4.23-rc4/configs/jabber.conf.sample | 18 + 1.4.23-rc4/configs/logger.conf.sample | 69 +++ 1.4.23-rc4/configs/manager.conf.sample | 56 ++ 1.4.23-rc4/configs/meetme.conf.sample | 26 + 1.4.23-rc4/configs/mgcp.conf.sample | 104 ++++ 1.4.23-rc4/configs/misdn.conf.sample | 438 ++++++++++++++++ 1.4.23-rc4/configs/modules.conf.sample | 35 ++ 1.4.23-rc4/configs/musiconhold.conf.sample | 66 +++ 1.4.23-rc4/configs/muted.conf.sample | 39 ++ 1.4.23-rc4/configs/osp.conf.sample | 72 +++ 1.4.23-rc4/configs/oss.conf.sample | 75 +++ 1.4.23-rc4/configs/phone.conf.sample | 49 ++ 1.4.23-rc4/configs/privacy.conf.sample | 3 + 1.4.23-rc4/configs/queues.conf.sample | 311 ++++++++++++ 1.4.23-rc4/configs/res_odbc.conf.sample | 49 ++ 1.4.23-rc4/configs/res_pgsql.conf.sample | 14 + 1.4.23-rc4/configs/res_snmp.conf.sample | 10 + 1.4.23-rc4/configs/rpt.conf.sample | 193 +++++++ 1.4.23-rc4/configs/rtp.conf.sample | 22 + 1.4.23-rc4/configs/say.conf.sample | 171 +++++++ 1.4.23-rc4/configs/sip.conf.sample | 685 +++++++++++++++++++++++++ 1.4.23-rc4/configs/sip_notify.conf.sample | 22 + 1.4.23-rc4/configs/skinny.conf.sample | 96 ++++ 1.4.23-rc4/configs/sla.conf.sample | 140 +++++ 1.4.23-rc4/configs/smdi.conf.sample | 75 +++ 1.4.23-rc4/configs/telcordia-1.adsi | 83 +++ 1.4.23-rc4/configs/udptl.conf.sample | 30 ++ 1.4.23-rc4/configs/users.conf.sample | 79 +++ 1.4.23-rc4/configs/voicemail.conf.sample | 248 +++++++++ 1.4.23-rc4/configs/vpb.conf.sample | 108 ++++ 61 files changed, 7926 insertions(+) create mode 100644 1.4.23-rc4/configs/adsi.conf.sample create mode 100644 1.4.23-rc4/configs/adtranvofr.conf.sample create mode 100644 1.4.23-rc4/configs/agents.conf.sample create mode 100644 1.4.23-rc4/configs/alarmreceiver.conf.sample create mode 100644 1.4.23-rc4/configs/alsa.conf.sample create mode 100644 1.4.23-rc4/configs/amd.conf.sample create mode 100644 1.4.23-rc4/configs/asterisk.adsi create mode 100644 1.4.23-rc4/configs/cdr.conf.sample create mode 100644 1.4.23-rc4/configs/cdr_custom.conf.sample create mode 100644 1.4.23-rc4/configs/cdr_manager.conf.sample create mode 100644 1.4.23-rc4/configs/cdr_odbc.conf.sample create mode 100644 1.4.23-rc4/configs/cdr_pgsql.conf.sample create mode 100644 1.4.23-rc4/configs/cdr_tds.conf.sample create mode 100644 1.4.23-rc4/configs/chan_dahdi.conf.sample create mode 100644 1.4.23-rc4/configs/codecs.conf.sample create mode 100644 1.4.23-rc4/configs/dnsmgr.conf.sample create mode 100644 1.4.23-rc4/configs/dundi.conf.sample create mode 100644 1.4.23-rc4/configs/enum.conf.sample create mode 100644 1.4.23-rc4/configs/extconfig.conf.sample create mode 100644 1.4.23-rc4/configs/extensions.ael.sample create mode 100644 1.4.23-rc4/configs/extensions.conf.sample create mode 100644 1.4.23-rc4/configs/features.conf.sample create mode 100644 1.4.23-rc4/configs/festival.conf.sample create mode 100644 1.4.23-rc4/configs/followme.conf.sample create mode 100644 1.4.23-rc4/configs/func_odbc.conf.sample create mode 100644 1.4.23-rc4/configs/gtalk.conf.sample create mode 100644 1.4.23-rc4/configs/h323.conf.sample create mode 100644 1.4.23-rc4/configs/http.conf.sample create mode 100644 1.4.23-rc4/configs/iax.conf.sample create mode 100644 1.4.23-rc4/configs/iaxprov.conf.sample create mode 100644 1.4.23-rc4/configs/indications.conf.sample create mode 100644 1.4.23-rc4/configs/jabber.conf.sample create mode 100644 1.4.23-rc4/configs/logger.conf.sample create mode 100644 1.4.23-rc4/configs/manager.conf.sample create mode 100644 1.4.23-rc4/configs/meetme.conf.sample create mode 100644 1.4.23-rc4/configs/mgcp.conf.sample create mode 100644 1.4.23-rc4/configs/misdn.conf.sample create mode 100644 1.4.23-rc4/configs/modules.conf.sample create mode 100644 1.4.23-rc4/configs/musiconhold.conf.sample create mode 100644 1.4.23-rc4/configs/muted.conf.sample create mode 100644 1.4.23-rc4/configs/osp.conf.sample create mode 100644 1.4.23-rc4/configs/oss.conf.sample create mode 100644 1.4.23-rc4/configs/phone.conf.sample create mode 100644 1.4.23-rc4/configs/privacy.conf.sample create mode 100644 1.4.23-rc4/configs/queues.conf.sample create mode 100644 1.4.23-rc4/configs/res_odbc.conf.sample create mode 100644 1.4.23-rc4/configs/res_pgsql.conf.sample create mode 100644 1.4.23-rc4/configs/res_snmp.conf.sample create mode 100644 1.4.23-rc4/configs/rpt.conf.sample create mode 100644 1.4.23-rc4/configs/rtp.conf.sample create mode 100644 1.4.23-rc4/configs/say.conf.sample create mode 100644 1.4.23-rc4/configs/sip.conf.sample create mode 100644 1.4.23-rc4/configs/sip_notify.conf.sample create mode 100644 1.4.23-rc4/configs/skinny.conf.sample create mode 100644 1.4.23-rc4/configs/sla.conf.sample create mode 100644 1.4.23-rc4/configs/smdi.conf.sample create mode 100644 1.4.23-rc4/configs/telcordia-1.adsi create mode 100644 1.4.23-rc4/configs/udptl.conf.sample create mode 100644 1.4.23-rc4/configs/users.conf.sample create mode 100644 1.4.23-rc4/configs/voicemail.conf.sample create mode 100644 1.4.23-rc4/configs/vpb.conf.sample (limited to '1.4.23-rc4/configs') diff --git a/1.4.23-rc4/configs/adsi.conf.sample b/1.4.23-rc4/configs/adsi.conf.sample new file mode 100644 index 000000000..0f36f80da --- /dev/null +++ b/1.4.23-rc4/configs/adsi.conf.sample @@ -0,0 +1,8 @@ +; +; Sample ADSI Configuration file +; +[intro] +alignment = center +greeting => Welcome to the +greeting => Asterisk +greeting => Open Source PBX diff --git a/1.4.23-rc4/configs/adtranvofr.conf.sample b/1.4.23-rc4/configs/adtranvofr.conf.sample new file mode 100644 index 000000000..dc7bcfc7c --- /dev/null +++ b/1.4.23-rc4/configs/adtranvofr.conf.sample @@ -0,0 +1,39 @@ +; +; Voice over Frame Relay (Adtran style) +; +; Configuration file + +[interfaces] +; +; Default language +; +;language=en +; +; Lines for which we are the user termination. They accept incoming +; and outgoing calls. We use the default context on the first 8 lines +; used by internal phones. +; +context=default +;user => voice00 +;user => voice01 +;user => voice02 +;user => voice03 +;user => voice04 +;user => voice05 +;user => voice06 +;user => voice07 +; Calls on 16 and 17 come from the outside world, so they get +; a little bit special treatment +context=remote +;user => voice16 +;user => voice17 +; +; Next we have lines which we only accept calls on, and typically +; do not send outgoing calls on (i.e. these are where we are the +; network termination) +; +;network => voice08 +;network => voice09 +;network => voice10 +;network => voice11 +;network => voice12 diff --git a/1.4.23-rc4/configs/agents.conf.sample b/1.4.23-rc4/configs/agents.conf.sample new file mode 100644 index 000000000..8c8889e24 --- /dev/null +++ b/1.4.23-rc4/configs/agents.conf.sample @@ -0,0 +1,107 @@ +; +; Agent configuration +; + +[general] +; +; Define whether callbacklogins should be stored in astdb for +; persistence. Persistent logins will be reloaded after +; Asterisk restarts. +; +persistentagents=yes + +; Enable or disable a single extension from logging in as multiple agents. +; The default value is "yes". +;multiplelogin=yes + +[agents] +; +; Define maxlogintries to allow agent to try max logins before +; failed. +; default to 3 +; +;maxlogintries=5 +; +; +; Define autologoff times if appropriate. This is how long +; the phone has to ring with no answer before the agent is +; automatically logged off (in seconds) +; +;autologoff=15 +; +; Define autologoffunavail to have agents automatically logged +; out when the extension that they are at returns a CHANUNAVAIL +; status when a call is attempted to be sent there. +; Default is "no". +; +;autologoffunavail=yes +; +; Define ackcall to require an acknowledgement by '#' when +; an agent logs in using agentcallbacklogin. Default is "no". +; Can also be set to "always", which will also require AgentLogin +; agents to acknowledge calls by pressing '#'. +; +;ackcall=no +; +; Define endcall to allow an agent to hangup a call by '*'. +; Default is "yes". Set this to "no" to ignore '*'. +; +;endcall=yes +; +; Define wrapuptime. This is the minimum amount of time when +; after disconnecting before the caller can receive a new call +; note this is in milliseconds. +; +;wrapuptime=5000 +; +; Define the default musiconhold for agents +; musiconhold => music_class +; +;musiconhold => default +; +; Define the default good bye sound file for agents +; default to vm-goodbye +; +;goodbye => goodbye_file +; +; Define updatecdr. This is whether or not to change the source +; channel in the CDR record for this call to agent/agent_id so +; that we know which agent generates the call +; +;updatecdr=no +; +; Group memberships for agents (may change in mid-file) +; +;group=3 +;group=1,2 +;group= +; +; -------------------------------------------------- +; This section is devoted to recording agent's calls +; The keywords are global to the chan_agent channel driver +; +; Enable recording calls addressed to agents. It's turned off by default. +;recordagentcalls=yes +; +; The format to be used to record the calls: wav, gsm, wav49. +; By default its "wav". +;recordformat=gsm +; +; The text to be added to the name of the recording. Allows forming a url link. +;urlprefix=http://localhost/calls/ +; +; The optional directory to save the conversations in. The default is +; /var/spool/asterisk/monitor +;savecallsin=/var/calls +; +; An optional custom beep sound file to play to always-connected agents. +;custom_beep=beep +; +; -------------------------------------------------- +; +; This section contains the agent definitions, in the form: +; +; agent => agentid,agentpassword,name +; +;agent => 1001,4321,Mark Spencer +;agent => 1002,4321,Will Meadows diff --git a/1.4.23-rc4/configs/alarmreceiver.conf.sample b/1.4.23-rc4/configs/alarmreceiver.conf.sample new file mode 100644 index 000000000..bf767dea3 --- /dev/null +++ b/1.4.23-rc4/configs/alarmreceiver.conf.sample @@ -0,0 +1,80 @@ +; +; alarmreceiver.conf +; +; Sample configuration file for the Asterisk alarm receiver application. +; + + +[general] + +; +; Specify a timestamp format for the metadata section of the event files +; Default is %a %b %d, %Y @ %H:%M:%S %Z + +timestampformat = %a %b %d, %Y @ %H:%M:%S %Z + +; +; Specify a command to execute when the caller hangs up +; +; Default is none +; + +;eventcmd = yourprogram -yourargs ... + +; +; Specify a spool directory for the event files. This setting is required +; if you want the app to be useful. Event files written to the spool +; directory will be of the template event-XXXXXX, where XXXXXX is a random +; and unique alphanumeric string. +; +; Default is none, and the events will be dropped on the floor. +; + +eventspooldir = /tmp + +; +; The alarmreceiver app can either log the events one-at-a-time to individual +; files in the spool directory, or it can store them until the caller +; disconnects and write them all to one file. +; +; The default setting for logindividualevents is no. +; + +logindividualevents = no + +; +; The timeout for receiving the first DTMF digit is adjustable from 1000 msec. +; to 10000 msec. The default is 2000 msec. Note: if you wish to test the +; receiver by entering digits manually, set this to a reasonable time out +; like 10000 milliseconds. + +fdtimeout = 2000 + +; +; The timeout for receiving subsequent DTMF digits is adjustable from +; 110 msec. to 4000 msec. The default is 200 msec. Note: if you wish to test +; the receiver by entering digits manually, set this to a reasonable time out +; like 4000 milliseconds. +; + +sdtimeout = 200 + +; +; The loudness of the ACK and Kissoff tones is adjustable from 100 to 8192. +; The default is 8192. This shouldn't need to be messed with, but is included +; just in case there are problems with signal levels. +; + +loudness = 8192 + +; +; The db-family setting allows the user to capture statistics on the number of +; calls, and the errors the alarm receiver sees. The default is for no +; db-family name to be defined and the database logging to be turned off. +; + +;db-family = yourfamily: + +; +; End of alarmreceiver.conf +; diff --git a/1.4.23-rc4/configs/alsa.conf.sample b/1.4.23-rc4/configs/alsa.conf.sample new file mode 100644 index 000000000..f55030618 --- /dev/null +++ b/1.4.23-rc4/configs/alsa.conf.sample @@ -0,0 +1,62 @@ +; +; Open Sound System Console Driver Configuration File +; +[general] +; +; Automatically answer incoming calls on the console? Choose yes if +; for example you want to use this as an intercom. +; +autoanswer=yes +; +; Default context (is overridden with @context syntax) +; +context=local +; +; Default extension to call +; +extension=s +; +; Default language +; +;language=en +; +; Default Music on Hold class to use when this channel is placed on hold in +; the case that the music class is not set on the channel with +; Set(CHANNEL(musicclass)=whatever) in the dialplan and the peer channel +; putting this one on hold did not suggest a class to use. +; +;mohinterpret=default +; +; Silence suppression can be enabled when sound is over a certain threshold. +; The value for the threshold should probably be between 500 and 2000 or so, +; but your mileage may vary. Use the echo test to evaluate the best setting. +;silencesuppression = yes +;silencethreshold = 1000 +; +; To set which ALSA device to use, change this parameter +;input_device=hw:0,0 +;output_device=hw:0,0 + +;------------------------------ JITTER BUFFER CONFIGURATION -------------------------- +; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an + ; ALSA channel. Defaults to "no". An enabled jitterbuffer will + ; be used only if the sending side can create and the receiving + ; side can not accept jitter. The ALSA channel can't accept jitter, + ; thus an enabled jitterbuffer on the receive ALSA side will always + ; be used if the sending side can create jitter. + +; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds. + +; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is + ; resynchronized. Useful to improve the quality of the voice, with + ; big jumps in/broken timestamps, usually sent from exotic devices + ; and programs. Defaults to 1000. + +; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP + ; channel. Two implementations are currently available - "fixed" + ; (with size always equals to jbmax-size) and "adaptive" (with + ; variable size, actually the new jb of IAX2). Defaults to fixed. + +; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no". +;----------------------------------------------------------------------------------- + diff --git a/1.4.23-rc4/configs/amd.conf.sample b/1.4.23-rc4/configs/amd.conf.sample new file mode 100644 index 000000000..ce4808a0c --- /dev/null +++ b/1.4.23-rc4/configs/amd.conf.sample @@ -0,0 +1,18 @@ +; +; Answering Machine Detection Configuration +; + +[general] +initial_silence = 2500 ; Maximum silence duration before the greeting. + ; If exceeded then MACHINE. +greeting = 1500 ; Maximum length of a greeting. If exceeded then MACHINE. +after_greeting_silence = 800 ; Silence after detecting a greeting. + ; If exceeded then HUMAN +total_analysis_time = 5000 ; Maximum time allowed for the algorithm to decide + ; on a HUMAN or MACHINE +min_word_length = 100 ; Minimum duration of Voice to considered as a word +between_words_silence = 50 ; Minimum duration of silence after a word to consider + ; the audio what follows as a new word +maximum_number_of_words = 3 ; Maximum number of words in the greeting. + ; If exceeded then MACHINE +silence_threshold = 256 diff --git a/1.4.23-rc4/configs/asterisk.adsi b/1.4.23-rc4/configs/asterisk.adsi new file mode 100644 index 000000000..a275502ac --- /dev/null +++ b/1.4.23-rc4/configs/asterisk.adsi @@ -0,0 +1,159 @@ +; +; Asterisk default ADSI script +; +; +; Begin with the preamble requirements +; +DESCRIPTION "Asterisk PBX" ; Name of vendor +VERSION 0x00 ; Version of stuff +;SECURITY "_AST" ; Security code +SECURITY 0X9BDBF7AC ; Security code +FDN 0x0000000F ; Descriptor number + +; +; Flags +; +FLAG "nocallwaiting" + +; +; Predefined strings +; +DISPLAY "titles" IS "** Asterisk PBX **" +DISPLAY "talkingto" IS "Call active." JUSTIFY LEFT +DISPLAY "callname" IS "$Call1p" JUSTIFY LEFT +DISPLAY "callnum" IS "$Call1s" JUSTIFY LEFT +DISPLAY "incoming" IS "Incoming call!" JUSTIFY LEFT +DISPLAY "ringing" IS "Calling... " JUSTIFY LEFT +DISPLAY "callended" IS "Call ended." JUSTIFY LEFT +DISPLAY "missedcall" IS "Missed call." JUSTIFY LEFT +DISPLAY "busy" IS "Busy." JUSTIFY LEFT +DISPLAY "reorder" IS "Reorder." JUSTIFY LEFT +DISPLAY "cwdisabled" IS "Callwait disabled" +DISPLAY "empty" IS "asdf" + +; +; Begin soft key definitions +; +KEY "callfwd" IS "CallFwd" OR "Call Forward" + OFFHOOK + VOICEMODE + WAITDIALTONE + SENDDTMF "*60" + GOTO "offHook" +ENDKEY + +KEY "vmail_OH" IS "VMail" OR "Voicemail" + OFFHOOK + VOICEMODE + WAITDIALTONE + SENDDTMF "8500" +ENDKEY + +KEY "vmail" IS "VMail" OR "Voicemail" + SENDDTMF "8500" +ENDKEY + +KEY "backspace" IS "BackSpc" OR "Backspace" + BACKSPACE +ENDKEY + +KEY "cwdisable" IS "CWDsble" OR "Disable Call Wait" + SENDDTMF "*70" + SETFLAG "nocallwaiting" + SHOWDISPLAY "cwdisabled" AT 4 + TIMERCLEAR + TIMERSTART 1 +ENDKEY + +KEY "cidblock" IS "CIDBlk" OR "Block Callerid" + SENDDTMF "*67" + SETFLAG "nocallwaiting" +ENDKEY + +; +; Begin main subroutine +; + +SUB "main" IS + IFEVENT NEARANSWER THEN + CLEAR + SHOWDISPLAY "titles" AT 1 NOUPDATE + SHOWDISPLAY "talkingto" AT 2 NOUPDATE + SHOWDISPLAY "callname" AT 3 + SHOWDISPLAY "callnum" AT 4 + GOTO "stableCall" + ENDIF + IFEVENT OFFHOOK THEN + CLEAR + CLEARFLAG "nocallwaiting" + CLEARDISPLAY + SHOWDISPLAY "titles" AT 1 + SHOWKEYS "vmail" + SHOWKEYS "cidblock" + SHOWKEYS "cwdisable" UNLESS "nocallwaiting" + GOTO "offHook" + ENDIF + IFEVENT IDLE THEN + CLEAR + SHOWDISPLAY "titles" AT 1 + SHOWKEYS "vmail_OH" + ENDIF + IFEVENT CALLERID THEN + CLEAR +; SHOWDISPLAY "titles" AT 1 NOUPDATE +; SHOWDISPLAY "incoming" AT 2 NOUPDATE + SHOWDISPLAY "callname" AT 3 NOUPDATE + SHOWDISPLAY "callnum" AT 4 + ENDIF + IFEVENT RING THEN + CLEAR + SHOWDISPLAY "titles" AT 1 NOUPDATE + SHOWDISPLAY "incoming" AT 2 + ENDIF + IFEVENT ENDOFRING THEN + SHOWDISPLAY "missedcall" AT 2 + CLEAR + SHOWDISPLAY "titles" AT 1 + SHOWKEYS "vmail_OH" + ENDIF + IFEVENT TIMER THEN + CLEAR + SHOWDISPLAY "empty" AT 4 + ENDIF +ENDSUB + +SUB "offHook" IS + IFEVENT FARRING THEN + CLEAR + SHOWDISPLAY "titles" AT 1 NOUPDATE + SHOWDISPLAY "ringing" AT 2 NOUPDATE + SHOWDISPLAY "callname" at 3 NOUPDATE + SHOWDISPLAY "callnum" at 4 + ENDIF + IFEVENT FARANSWER THEN + CLEAR + SHOWDISPLAY "talkingto" AT 2 + GOTO "stableCall" + ENDIF + IFEVENT BUSY THEN + CLEAR + SHOWDISPLAY "titles" AT 1 NOUPDATE + SHOWDISPLAY "busy" AT 2 NOUPDATE + SHOWDISPLAY "callname" at 3 NOUPDATE + SHOWDISPLAY "callnum" at 4 + ENDIF + IFEVENT REORDER THEN + CLEAR + SHOWDISPLAY "titles" AT 1 NOUPDATE + SHOWDISPLAY "reorder" AT 2 NOUPDATE + SHOWDISPLAY "callname" at 3 NOUPDATE + SHOWDISPLAY "callnum" at 4 + ENDIF +ENDSUB + +SUB "stableCall" IS + IFEVENT REORDER THEN + SHOWDISPLAY "callended" AT 2 + ENDIF +ENDSUB + diff --git a/1.4.23-rc4/configs/cdr.conf.sample b/1.4.23-rc4/configs/cdr.conf.sample new file mode 100644 index 000000000..693b28092 --- /dev/null +++ b/1.4.23-rc4/configs/cdr.conf.sample @@ -0,0 +1,148 @@ +; +; Asterisk Call Detail Record engine configuration +; +; CDR is Call Detail Record, which provides logging services via a variety of +; pluggable backend modules. Detailed call information can be recorded to +; databases, files, etc. Useful for billing, fraud prevention, compliance with +; Sarbanes-Oxley aka The Enron Act, QOS evaluations, and more. +; + +[general] + +; Define whether or not to use CDR logging. Setting this to "no" will override +; any loading of backend CDR modules. Default is "yes". +;enable=yes + +; Define whether or not to log unanswered calls. Setting this to "yes" will +; report every attempt to ring a phone in dialing attempts, when it was not +; answered. For example, if you try to dial 3 extensions, and this option is "yes", +; you will get 3 CDR's, one for each phone that was rung. Default is "no". Some +; find this information horribly useless. Others find it very valuable. Note, in "yes" +; mode, you will see one CDR, with one of the call targets on one side, and the originating +; channel on the other, and then one CDR for each channel attempted. This may seem +; redundant, but cannot be helped. +;unanswered = no + +; Define the CDR batch mode, where instead of posting the CDR at the end of +; every call, the data will be stored in a buffer to help alleviate load on the +; asterisk server. Default is "no". +; +; WARNING WARNING WARNING +; Use of batch mode may result in data loss after unsafe asterisk termination +; ie. software crash, power failure, kill -9, etc. +; WARNING WARNING WARNING +; +;batch=no + +; Define the maximum number of CDRs to accumulate in the buffer before posting +; them to the backend engines. 'batch' must be set to 'yes'. Default is 100. +;size=100 + +; Define the maximum time to accumulate CDRs in the buffer before posting them +; to the backend engines. If this time limit is reached, then it will post the +; records, regardless of the value defined for 'size'. 'batch' must be set to +; 'yes'. Note that time is in seconds. Default is 300 (5 minutes). +;time=300 + +; The CDR engine uses the internal asterisk scheduler to determine when to post +; records. Posting can either occur inside the scheduler thread, or a new +; thread can be spawned for the submission of every batch. For small batches, +; it might be acceptable to just use the scheduler thread, so set this to "yes". +; For large batches, say anything over size=10, a new thread is recommended, so +; set this to "no". Default is "no". +;scheduleronly=no + +; When shutting down asterisk, you can block until the CDRs are submitted. If +; you don't, then data will likely be lost. You can always check the size of +; the CDR batch buffer with the CLI "cdr status" command. To enable blocking on +; submission of CDR data during asterisk shutdown, set this to "yes". Default +; is "yes". +;safeshutdown=yes + +; Normally, CDR's are not closed out until after all extensions are finished +; executing. By enabling this option, the CDR will be ended before executing +; the "h" extension so that CDR values such as "end" and "billsec" may be +; retrieved inside of of this extension. +;endbeforehexten=no + +; +; +; CHOOSING A CDR "BACKEND" (what kind of output to generate) +; +; To choose a backend, you have to make sure either the right category is +; defined in this file, or that the appropriate config file exists, and has the +; proper definitions in it. If there are any problems, usually, the entry will +; silently ignored, and you get no output. +; +; Also, please note that you can generate CDR records in as many formats as you +; wish. If you configure 5 different CDR formats, then each event will be logged +; in 5 different places! In the example config files, all formats are commented +; out except for the cdr-csv format. +; +; Here are all the possible back ends: +; +; csv, custom, manager, odbc, pgsql, radius, sqlite, tds +; (also, mysql is available via the asterisk-addons, due to licensing +; requirements) +; (please note, also, that other backends can be created, by creating +; a new backend module in the source cdr/ directory!) +; +; Some of the modules required to provide these backends will not build or install +; unless some dependency requirements are met. Examples of this are pgsql, odbc, +; etc. If you are not getting output as you would expect, the first thing to do +; is to run the command "make menuselect", and check what modules are available, +; by looking in the "2. Call Detail Recording" option in the main menu. If your +; backend is marked with XXX, you know that the "configure" command could not find +; the required libraries for that option. +; +; To get CDRs to be logged to the plain-jane /var/log/asterisk/cdr-csv/Master.csv +; file, define the [csv] category in this file. No database necessary. The example +; config files are set up to provide this kind of output by default. +; +; To get custom csv CDR records, make sure the cdr_custom.conf file +; is present, and contains the proper [mappings] section. The advantage to +; using this backend, is that you can define which fields to output, and in +; what order. By default, the example configs are set up to mimic the cdr-csv +; output. If you don't make any changes to the mappings, you are basically generating +; the same thing as cdr-csv, but expending more CPU cycles to do so! +; +; To get manager events generated, make sure the cdr_manager.conf file exists, +; and the [general] section is defined, with the single variable 'enabled = yes'. +; +; For odbc, make sure all the proper libs are installed, that "make menuselect" +; shows that the modules are available, and the cdr_odbc.conf file exists, and +; has a [global] section with the proper variables defined. +; +; For pgsql, make sure all the proper libs are installed, that "make menuselect" +; shows that the modules are available, and the cdr_pgsql.conf file exists, and +; has a [global] section with the proper variables defined. +; +; For logging to radius databases, make sure all the proper libs are installed, that +; "make menuselect" shows that the modules are available, and the [radius] +; category is defined in this file, and in that section, make sure the 'radiuscfg' +; variable is properly pointing to an existing radiusclient.conf file. +; +; For logging to sqlite databases, make sure the 'cdr.db' file exists in the log directory, +; which is usually /var/log/asterisk. Of course, the proper libraries should be available +; during the 'configure' operation. +; +; For tds logging, make sure the proper libraries are available during the 'configure' +; phase, and that cdr_tds.conf exists and is properly set up with a [global] category. +; +; Also, remember, that if you wish to log CDR info to a database, you will have to define +; a specific table in that databse to make things work! See the doc directory for more details +; on how to create this table in each database. +; + +[csv] +usegmtime=yes ; log date/time in GMT. Default is "no" +loguniqueid=yes ; log uniqueid. Default is "no" +loguserfield=yes ; log user field. Default is "no" + +;[radius] +;usegmtime=yes ; log date/time in GMT +;loguniqueid=yes ; log uniqueid +;loguserfield=yes ; log user field +; Set this to the location of the radiusclient-ng configuration file +; The default is /etc/radiusclient-ng/radiusclient.conf +;radiuscfg => /usr/local/etc/radiusclient-ng/radiusclient.conf diff --git a/1.4.23-rc4/configs/cdr_custom.conf.sample b/1.4.23-rc4/configs/cdr_custom.conf.sample new file mode 100644 index 000000000..8bc2cb34e --- /dev/null +++ b/1.4.23-rc4/configs/cdr_custom.conf.sample @@ -0,0 +1,10 @@ +; +; Mappings for custom config file +; +; to get your csv output in a format tailored to your liking, uncomment the following +; and look for the output in the cdr-custom/Master.csv file (usually in /var/log/asterisk). +; +; +;[mappings] +;Master.csv => "${CDR(clid)}","${CDR(src)}","${CDR(dst)}","${CDR(dcontext)}","${CDR(channel)}","${CDR(dstchannel)}","${CDR(lastapp)}","${CDR(lastdata)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}" + diff --git a/1.4.23-rc4/configs/cdr_manager.conf.sample b/1.4.23-rc4/configs/cdr_manager.conf.sample new file mode 100644 index 000000000..1d7984ba4 --- /dev/null +++ b/1.4.23-rc4/configs/cdr_manager.conf.sample @@ -0,0 +1,6 @@ +; +; Asterisk Call Management CDR +; +[general] +enabled = no + diff --git a/1.4.23-rc4/configs/cdr_odbc.conf.sample b/1.4.23-rc4/configs/cdr_odbc.conf.sample new file mode 100644 index 000000000..6245e37eb --- /dev/null +++ b/1.4.23-rc4/configs/cdr_odbc.conf.sample @@ -0,0 +1,12 @@ +; +; cdr_odbc.conf +; + +;[global] +;dsn=MySQL-test +;username=username +;password=password +;loguniqueid=yes +;dispositionstring=yes +;table=cdr ;"cdr" is default table name +;usegmtime=no ; set to "yes" to log in GMT diff --git a/1.4.23-rc4/configs/cdr_pgsql.conf.sample b/1.4.23-rc4/configs/cdr_pgsql.conf.sample new file mode 100644 index 000000000..0784c7b08 --- /dev/null +++ b/1.4.23-rc4/configs/cdr_pgsql.conf.sample @@ -0,0 +1,9 @@ +; Sample Asterisk config file for CDR logging to PostgresSQL + +[global] +;hostname=localhost +;port=5432 +;dbname=asterisk +;password=password +;user=postgres +;table=cdr ;SQL table where CDRs will be inserted diff --git a/1.4.23-rc4/configs/cdr_tds.conf.sample b/1.4.23-rc4/configs/cdr_tds.conf.sample new file mode 100644 index 000000000..d8c7d075c --- /dev/null +++ b/1.4.23-rc4/configs/cdr_tds.conf.sample @@ -0,0 +1,11 @@ +; Sample Asterisk config file for CDR logging to FreeTDS + +;[global] +;hostname=fs.malico.loc +;port=1433 +;dbname=MalicoHN +;user=mangUsr +;password= +;charset=BIG5 +;table=cdr + diff --git a/1.4.23-rc4/configs/chan_dahdi.conf.sample b/1.4.23-rc4/configs/chan_dahdi.conf.sample new file mode 100644 index 000000000..2dbbb41d0 --- /dev/null +++ b/1.4.23-rc4/configs/chan_dahdi.conf.sample @@ -0,0 +1,675 @@ +; +; DAHDI telephony interface +; +; Configuration file +; +; You need to restart Asterisk to re-configure the DAHDI channels +; CLI> reload chan_dahdi.so +; will reload the configuration file, +; but not all configuration options are +; re-configured during a reload. + + + +[trunkgroups] +; +; Trunk groups are used for NFAS or GR-303 connections. +; +; Group: Defines a trunk group. +; trunkgroup => ,[,...] +; +; trunkgroup is the numerical trunk group to create +; dchannel is the DAHDI channel which will have the +; d-channel for the trunk. +; backup1 is an optional list of backup d-channels. +; +;trunkgroup => 1,24,48 +;trunkgroup => 1,24 +; +; Spanmap: Associates a span with a trunk group +; spanmap => ,[,] +; +; dahdispan is the DAHDI span number to associate +; trunkgroup is the trunkgroup (specified above) for the mapping +; logicalspan is the logical span number within the trunk group to use. +; if unspecified, no logical span number is used. +; +;spanmap => 1,1,1 +;spanmap => 2,1,2 +;spanmap => 3,1,3 +;spanmap => 4,1,4 + +[channels] +; +; Default language +; +;language=en +; +; Default context +; +context=default +; +; Switchtype: Only used for PRI. +; +; national: National ISDN 2 (default) +; dms100: Nortel DMS100 +; 4ess: AT&T 4ESS +; 5ess: Lucent 5ESS +; euroisdn: EuroISDN (also known as ETSI NET/5; Cisco calls this "primary-net5") +; ni1: Old National ISDN 1 +; qsig: Q.SIG +; +switchtype=national +; +; Some switches (AT&T especially) require network specific facility IE +; supported values are currently 'none', 'sdn', 'megacom', 'tollfreemegacom', 'accunet' +; +;nsf=none +; +; PRI Dialplan: Only RARELY used for PRI. +; +; unknown: Unknown +; private: Private ISDN +; local: Local ISDN +; national: National ISDN +; international: International ISDN +; dynamic: Dynamically selects the appropriate dialplan +; +;pridialplan=national +; +; PRI Local Dialplan: Only RARELY used for PRI (sets the calling number's numbering plan) +; +; unknown: Unknown +; private: Private ISDN +; local: Local ISDN +; national: National ISDN +; international: International ISDN +; dynamic: Dynamically selects the appropriate dialplan +; +;prilocaldialplan=national +; +; PRI callerid prefixes based on the given TON/NPI (dialplan) +; This is especially needed for euroisdn E1-PRIs +; +; sample 1 for Germany +;internationalprefix = 00 +;nationalprefix = 0 +;localprefix = 0711 +;privateprefix = 07115678 +;unknownprefix = +; +; sample 2 for Germany +;internationalprefix = + +;nationalprefix = +49 +;localprefix = +49711 +;privateprefix = +497115678 +;unknownprefix = +; +; PRI resetinterval: sets the time in seconds between restart of unused +; channels, defaults to 3600; minimum 60 seconds. Some PBXs don't like +; channel restarts. so set the interval to a very long interval e.g. 100000000 +; or 'never' to disable *entirely*. +; +;resetinterval = 3600 +; +; Overlap dialing mode (sending overlap digits) +; +;overlapdial=yes +; +; Allow inband audio (progress) when a call is RELEASEd by the far end of a PRI +; +;inbanddisconnect=yes +; +; PRI Out of band indications. +; Enable this to report Busy and Congestion on a PRI using out-of-band +; notification. Inband indication, as used by Asterisk doesn't seem to work +; with all telcos. +; +; outofband: Signal Busy/Congestion out of band with RELEASE/DISCONNECT +; inband: Signal Busy/Congestion using in-band tones +; +; priindication = outofband +; +; If you need to override the existing channels selection routine and force all +; PRI channels to be marked as exclusively selected, set this to yes. +; priexclusive = yes +; +; ISDN Timers +; All of the ISDN timers and counters that are used are configurable. Specify +; the timer name, and its value (in ms for timers). +; K: Layer 2 max number of outstanding unacknowledged I frames (default 7) +; N200: Layer 2 max number of retransmissions of a frame (default 3) +; T200: Layer 2 max time before retransmission of a frame (default 1000 ms) +; T203: Layer 2 max time without frames being exchanged (default 10000 ms) +; T305: Wait for DISCONNECT acknowledge (default 30000 ms) +; T308: Wait for RELEASE acknowledge (default 4000 ms) +; T309: Maintain active calls on Layer 2 disconnection (default -1, Asterisk clears calls) +; EuroISDN: 6000 to 12000 ms, according to (N200 + 1) x T200 + 2s +; May vary in other ISDN standards (Q.931 1993 : 90000 ms) +; T313: Wait for CONNECT acknowledge, CPE side only (default 3000 ms) +; +; pritimer => t200,1000 +; pritimer => t313,4000 +; +; To enable transmission of facility-based ISDN supplementary services (such +; as caller name from CPE over facility), enable this option. +; facilityenable = yes +; +; +; Signalling method (default is fxs). Valid values: +; em: E & M +; em_w: E & M Wink +; featd: Feature Group D (The fake, Adtran style, DTMF) +; featdmf: Feature Group D (The real thing, MF (domestic, US)) +; featdmf_ta: Feature Group D (The real thing, MF (domestic, US)) through +; a Tandem Access point +; featb: Feature Group B (MF (domestic, US)) +; fgccama Feature Group C-CAMA (DP DNIS, MF ANI) +; fgccamamf Feature Group C-CAMA MF (MF DNIS, MF ANI) +; fxs_ls: FXS (Loop Start) +; fxs_gs: FXS (Ground Start) +; fxs_ks: FXS (Kewl Start) +; fxo_ls: FXO (Loop Start) +; fxo_gs: FXO (Ground Start) +; fxo_ks: FXO (Kewl Start) +; pri_cpe: PRI signalling, CPE side +; pri_net: PRI signalling, Network side +; gr303fxoks_net: GR-303 Signalling, FXO Loopstart, Network side +; gr303fxsks_cpe: GR-303 Signalling, FXS Loopstart, CPE side +; sf: SF (Inband Tone) Signalling +; sf_w: SF Wink +; sf_featd: SF Feature Group D (The fake, Adtran style, DTMF) +; sf_featdmf: SF Feature Group D (The real thing, MF (domestic, US)) +; sf_featb: SF Feature Group B (MF (domestic, US)) +; e911: E911 (MF) style signalling +; +; The following are used for Radio interfaces: +; fxs_rx: Receive audio/COR on an FXS kewlstart interface (FXO at the +; channel bank) +; fxs_tx: Transmit audio/PTT on an FXS loopstart interface (FXO at the +; channel bank) +; fxo_rx: Receive audio/COR on an FXO loopstart interface (FXS at the +; channel bank) +; fxo_tx: Transmit audio/PTT on an FXO groundstart interface (FXS at +; the channel bank) +; em_rx: Receive audio/COR on an E&M interface (1-way) +; em_tx: Transmit audio/PTT on an E&M interface (1-way) +; em_txrx: Receive audio/COR AND Transmit audio/PTT on an E&M interface +; (2-way) +; em_rxtx: Same as em_txrx (for our dyslexic friends) +; sf_rx: Receive audio/COR on an SF interface (1-way) +; sf_tx: Transmit audio/PTT on an SF interface (1-way) +; sf_txrx: Receive audio/COR AND Transmit audio/PTT on an SF interface +; (2-way) +; sf_rxtx: Same as sf_txrx (for our dyslexic friends) +; +signalling=fxo_ls +; +; If you have an outbound signalling format that is different from format +; specified above (but compatible), you can specify outbound signalling format, +; (see below). The 'signalling' format specified will be the inbound signalling +; format. If you only specify 'signalling', then it will be the format for +; both inbound and outbound. +; +; signalling=featdmf +; outsignalling=featb +; +; For Feature Group D Tandem access, to set the default CIC and OZZ use these +; parameters: +;defaultozz=0000 +;defaultcic=303 +; +; A variety of timing parameters can be specified as well +; Including: +; prewink: Pre-wink time (default 50ms) +; preflash: Pre-flash time (default 50ms) +; wink: Wink time (default 150ms) +; flash: Flash time (default 750ms) +; start: Start time (default 1500ms) +; rxwink: Receiver wink time (default 300ms) +; rxflash: Receiver flashtime (default 1250ms) +; debounce: Debounce timing (default 600ms) +; +rxwink=300 ; Atlas seems to use long (250ms) winks +; +; How long generated tones (DTMF and MF) will be played on the channel +; (in milliseconds) +;toneduration=100 +; +; Whether or not to do distinctive ring detection on FXO lines +; +;usedistinctiveringdetection=yes +;distinctiveringaftercid=yes ; enable dring detection after callerid for those countries like Australia + ; where the ring cadence is changed *after* the callerid spill. +; +; Whether or not to use caller ID +; +usecallerid=yes +; +; Type of caller ID signalling in use +; bell = bell202 as used in US +; v23 = v23 as used in the UK +; v23_jp = v23 as used in Japan +; dtmf = DTMF as used in Denmark, Sweden and Netherlands +; smdi = Use SMDI for callerid. Requires SMDI to be enabled (usesmdi). +; +;cidsignalling=bell +; +; What signals the start of caller ID +; ring = a ring signals the start +; polarity = polarity reversal signals the start +; +;cidstart=ring +; +; Whether or not to hide outgoing caller ID (Override with *67 or *82) +; +hidecallerid=no +; +; Whether or not to enable call waiting on internal extensions +; With this set to 'yes', busy extensions will hear the call-waiting +; tone, and can use hook-flash to switch between callers. The Dial() +; app will not return the "BUSY" result for extensions. +; +callwaiting=yes +; +; Whether or not restrict outgoing caller ID (will be sent as ANI only, not +; available for the user) +; Mostly use with FXS ports +; +;restrictcid=no +; +; Whether or not use the caller ID presentation for the outgoing call that the +; calling switch is sending. +; See doc/callingpres.txt +; +usecallingpres=yes +; +; Some countries (UK) have ring tones with different ring tones (ring-ring), +; which means the callerid needs to be set later on, and not just after +; the first ring, as per the default. +; +;sendcalleridafter=1 +; +; +; Support Caller*ID on Call Waiting +; +callwaitingcallerid=yes +; +; Support three-way calling +; +threewaycalling=yes +; +; For FXS ports (either direct analog or over T1/E1): +; Support flash-hook call transfer (requires three way calling) +; Also enables call parking (overrides the 'canpark' parameter) +; +; For digital ports using ISDN PRI protocols: +; Support switch-side transfer (called 2BCT, RLT or other names) +; This setting must be enabled on both ports involved, and the +; 'facilityenable' setting must also be enabled to allow sending +; the transfer to the ISDN switch, since it sent in a FACILITY +; message. +; +transfer=yes +; +; Allow call parking +; ('canpark=no' is overridden by 'transfer=yes') +; +canpark=yes +; +; Support call forward variable +; +cancallforward=yes +; +; Whether or not to support Call Return (*69) +; +callreturn=yes +; +; Stutter dialtone support: If a mailbox is specified without a voicemail +; context, then when voicemail is received in a mailbox in the default +; voicemail context in voicemail.conf, taking the phone off hook will cause a +; stutter dialtone instead of a normal one. +; +; If a mailbox is specified *with* a voicemail context, the same will result +; if voicemail received in mailbox in the specified voicemail context. +; +; for default voicemail context, the example below is fine: +; +;mailbox=1234 +; +; for any other voicemail context, the following will produce the stutter tone: +; +;mailbox=1234@context +; +; Enable echo cancellation +; Use either "yes", "no", or a power of two from 32 to 256 if you wish to +; actually set the number of taps of cancellation. +; +; Note that when setting the number of taps, the number 256 does not translate +; to 256 ms of echo cancellation. echocancel=256 means 256 / 8 = 32 ms. +; +; Note that if any of your DAHDI cards have hardware echo cancellers, +; then this setting only turns them on and off; numeric settings will +; be treated as "yes". There are no special settings required for +; hardware echo cancellers; when present and enabled in their kernel +; modules, they take precedence over the software echo canceller compiled +; into DAHDI automatically. +; +echocancel=yes +; +; Generally, it is not necessary (and in fact undesirable) to echo cancel when +; the circuit path is entirely TDM. You may, however, change this behavior +; by enabling the echo cancel during pure TDM bridging below. +; +echocancelwhenbridged=yes +; +; In some cases, the echo canceller doesn't train quickly enough and there +; is echo at the beginning of the call. Enabling echo training will cause +; asterisk to briefly mute the channel, send an impulse, and use the impulse +; response to pre-train the echo canceller so it can start out with a much +; closer idea of the actual echo. Value may be "yes", "no", or a number of +; milliseconds to delay before training (default = 400) +; +; WARNING: In some cases this option can make echo worse! If you are +; trying to debug an echo problem, it is worth checking to see if your echo +; is better with the option set to yes or no. Use whatever setting gives +; the best results. +; +; Note that these parameters do not apply to hardware echo cancellers. +; +;echotraining=yes +;echotraining=800 +; +; If you are having trouble with DTMF detection, you can relax the DTMF +; detection parameters. Relaxing them may make the DTMF detector more likely +; to have "talkoff" where DTMF is detected when it shouldn't be. +; +;relaxdtmf=yes +; +; You may also set the default receive and transmit gains (in dB) +; +rxgain=0.0 +txgain=0.0 +; +; Logical groups can be assigned to allow outgoing rollover. Groups range +; from 0 to 63, and multiple groups can be specified. +; +group=1 +; +; Ring groups (a.k.a. call groups) and pickup groups. If a phone is ringing +; and it is a member of a group which is one of your pickup groups, then +; you can answer it by picking up and dialling *8#. For simple offices, just +; make these both the same. Groups range from 0 to 63. +; +callgroup=1 +pickupgroup=1 + +; +; Specify whether the channel should be answered immediately or if the simple +; switch should provide dialtone, read digits, etc. +; Note: If immediate=yes the dialplan execution will always start at extension +; 's' priority 1 regardless of the dialed number! +; +immediate=no +; +; Specify whether flash-hook transfers to 'busy' channels should complete or +; return to the caller performing the transfer (default is yes). +; +;transfertobusy=no +; +; CallerID can be set to "asreceived" or a specific number if you want to +; override it. Note that "asreceived" only applies to trunk interfaces. +; +;callerid=2564286000 +; +; AMA flags affects the recording of Call Detail Records. If specified +; it may be 'default', 'omit', 'billing', or 'documentation'. +; +;amaflags=default +; +; Channels may be associated with an account code to ease +; billing +; +;accountcode=lss0101 +; +; ADSI (Analog Display Services Interface) can be enabled on a per-channel +; basis if you have (or may have) ADSI compatible CPE equipment +; +;adsi=yes +; +; SMDI (Simplified Message Desk Interface) can be enabled on a per-channel +; basis if you would like that channel to behave like an SMDI message desk. +; The SMDI port specified should have already been defined in smdi.conf. The +; default port is /dev/ttyS0. +; +;usesmdi=yes +;smdiport=/dev/ttyS0 +; +; On trunk interfaces (FXS) and E&M interfaces (E&M, Wink, Feature Group D +; etc, it can be useful to perform busy detection either in an effort to +; detect hangup or for detecting busies. This enables listening for +; the beep-beep busy pattern. +; +;busydetect=yes +; +; If busydetect is enabled, it is also possible to specify how many busy tones +; to wait for before hanging up. The default is 4, but better results can be +; achieved if set to 6 or even 8. Mind that the higher the number, the more +; time that will be needed to hangup a channel, but lowers the probability +; that you will get random hangups. +; +;busycount=4 +; +; If busydetect is enabled, it is also possible to specify the cadence of your +; busy signal. In many countries, it is 500msec on, 500msec off. Without +; busypattern specified, we'll accept any regular sound-silence pattern that +; repeats times as a busy signal. If you specify busypattern, +; then we'll further check the length of the sound (tone) and silence, which +; will further reduce the chance of a false positive. +; +;busypattern=500,500 +; +; NOTE: In the Asterisk Makefile you'll find further options to tweak the busy +; detector. If your country has a busy tone with the same length tone and +; silence (as many countries do), consider defining the +; -DBUSYDETECT_COMPARE_TONE_AND_SILENCE option. +; +; Use a polarity reversal to mark when a outgoing call is answered by the +; remote party. +; +;answeronpolarityswitch=yes +; +; In some countries, a polarity reversal is used to signal the disconnect of a +; phone line. If the hanguponpolarityswitch option is selected, the call will +; be considered "hung up" on a polarity reversal. +; +;hanguponpolarityswitch=yes +; +; On trunk interfaces (FXS) it can be useful to attempt to follow the progress +; of a call through RINGING, BUSY, and ANSWERING. If turned on, call +; progress attempts to determine answer, busy, and ringing on phone lines. +; This feature is HIGHLY EXPERIMENTAL and can easily detect false answers, +; so don't count on it being very accurate. +; +; Few zones are supported at the time of this writing, but may be selected +; with "progzone" +; +; This feature can also easily detect false hangups. The symptoms of this is +; being disconnected in the middle of a call for no reason. +; +;callprogress=yes +;progzone=us +; +; FXO (FXS signalled) devices must have a timeout to determine if there was a +; hangup before the line was answered. This value can be tweaked to shorten +; how long it takes before DAHDI considers a non-ringing line to have hungup. +; +;ringtimeout=8000 +; +; For FXO (FXS signalled) devices, whether to use pulse dial instead of DTMF +; +;pulsedial=yes +; +; For fax detection, uncomment one of the following lines. The default is *OFF* +; +;faxdetect=both +;faxdetect=incoming +;faxdetect=outgoing +;faxdetect=no +; +; This option specifies a preference for which music on hold class this channel +; should listen to when put on hold if the music class has not been set on the +; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer +; channel putting this one on hold did not suggest a music class. +; +; If this option is set to "passthrough", then the hold message will always be +; passed through as signalling instead of generating hold music locally. This +; setting is only valid when used on a channel that uses digital signalling. +; +; This option may be specified globally, or on a per-channel basis. +; +;mohinterpret=default +; +; This option specifies which music on hold class to suggest to the peer channel +; when this channel places the peer on hold. It may be specified globally or on +; a per-channel. +; +;mohsuggest=default +; +; PRI channels can have an idle extension and a minunused number. So long as +; at least "minunused" channels are idle, chan_dahdi will try to call "idledial" +; on them, and then dump them into the PBX in the "idleext" extension (which +; is of the form exten@context). When channels are needed the "idle" calls +; are disconnected (so long as there are at least "minidle" calls still +; running, of course) to make more channels available. The primary use of +; this is to create a dynamic service, where idle channels are bundled through +; multilink PPP, thus more efficiently utilizing combined voice/data services +; than conventional fixed mappings/muxings. +; +;idledial=6999 +;idleext=6999@dialout +;minunused=2 +;minidle=1 +; +; Configure jitter buffers in DAHDI (each one is 20ms, default is 4) +; +;jitterbuffers=4 +; +;------------------------------ JITTER BUFFER CONFIGURATION -------------------------- +; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a + ; DAHDI channel. Defaults to "no". An enabled jitterbuffer will + ; be used only if the sending side can create and the receiving + ; side can not accept jitter. The DAHDI channel can't accept jitter, + ; thus an enabled jitterbuffer on the receive DAHDI side will always + ; be used if the sending side can create jitter. + +; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds. + +; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is + ; resynchronized. Useful to improve the quality of the voice, with + ; big jumps in/broken timestamps, usually sent from exotic devices + ; and programs. Defaults to 1000. + +; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a DAHDI + ; channel. Two implementations are currently available - "fixed" + ; (with size always equals to jbmax-size) and "adaptive" (with + ; variable size, actually the new jb of IAX2). Defaults to fixed. + +; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no". +;----------------------------------------------------------------------------------- +; +; You can define your own custom ring cadences here. You can define up to 8 +; pairs. If the silence is negative, it indicates where the callerid spill is +; to be placed. Also, if you define any custom cadences, the default cadences +; will be turned off. +; +; Syntax is: cadence=ring,silence[,ring,silence[...]] +; +; These are the default cadences: +; +;cadence=125,125,2000,-4000 +;cadence=250,250,500,1000,250,250,500,-4000 +;cadence=125,125,125,125,125,-4000 +;cadence=1000,500,2500,-5000 +; +; Each channel consists of the channel number or range. It inherits the +; parameters that were specified above its declaration. +; +; For GR-303, CRV's are created like channels except they must start with the +; trunk group followed by a colon, e.g.: +; +; crv => 1:1 +; crv => 2:1-2,5-8 +; +; +;callerid="Green Phone"<(256) 428-6121> +;channel => 1 +;callerid="Black Phone"<(256) 428-6122> +;channel => 2 +;callerid="CallerID Phone" <(256) 428-6123> +;callerid="CallerID Phone" <(630) 372-1564> +;callerid="CallerID Phone" <(256) 704-4666> +;channel => 3 +;callerid="Pac Tel Phone" <(256) 428-6124> +;channel => 4 +;callerid="Uniden Dead" <(256) 428-6125> +;channel => 5 +;callerid="Cortelco 2500" <(256) 428-6126> +;channel => 6 +;callerid="Main TA 750" <(256) 428-6127> +;channel => 44 +; +; For example, maybe we have some other channels which start out in a +; different context and use E & M signalling instead. +; +;context=remote +;signaling=em +;channel => 15 +;channel => 16 + +;signalling=em_w +; +; All those in group 0 I'll use for outgoing calls +; +; Strip most significant digit (9) before sending +; +;stripmsd=1 +;callerid=asreceived +;group=0 +;signalling=fxs_ls +;channel => 45 + +;signalling=fxo_ls +;group=1 +;callerid="Joe Schmoe" <(256) 428-6131> +;channel => 25 +;callerid="Megan May" <(256) 428-6132> +;channel => 26 +;callerid="Suzy Queue" <(256) 428-6233> +;channel => 27 +;callerid="Larry Moe" <(256) 428-6234> +;channel => 28 +; +; Sample PRI (CPE) config: Specify the switchtype, the signalling as either +; pri_cpe or pri_net for CPE or Network termination, and generally you will +; want to create a single "group" for all channels of the PRI. +; +; switchtype = national +; signalling = pri_cpe +; group = 2 +; channel => 1-23 + +; + +; Used for distinctive ring support for x100p. +; You can see the dringX patterns is to set any one of the dringXcontext fields +; and they will be printed on the console when an inbound call comes in. +; +;dring1=95,0,0 +;dring1context=internal1 +;dring2=325,95,0 +;dring2context=internal2 +; If no pattern is matched here is where we go. +;context=default +;channel => 1 + diff --git a/1.4.23-rc4/configs/codecs.conf.sample b/1.4.23-rc4/configs/codecs.conf.sample new file mode 100644 index 000000000..c8caeab60 --- /dev/null +++ b/1.4.23-rc4/configs/codecs.conf.sample @@ -0,0 +1,65 @@ +[speex] +; CBR encoding quality [0..10] +; used only when vbr = false +quality => 3 + +; codec complexity [0..10] +; tradeoff between cpu/quality +complexity => 2 + +; perceptual enhancement [true / false] +; improves clarity of decoded speech +enhancement => true + +; voice activity detection [true / false] +; reduces bitrate when no voice detected, used only for CBR +; (implicit in VBR/ABR) +vad => true + +; variable bit rate [true / false] +; uses bit rate proportionate to voice complexity +vbr => true + +; available bit rate [bps, 0 = off] +; encoding quality modulated to match this target bit rate +; not recommended with dtx or pp_vad - may cause bandwidth spikes +abr => 0 + +; VBR encoding quality [0-10] +; floating-point values allowed +vbr_quality => 4 + +; discontinuous transmission [true / false] +; stops transmitting completely when silence is detected +; pp_vad is far more effective but more CPU intensive +dtx => false + +; preprocessor configuration +; these options only affect Speex v1.1.8 or newer + +; enable preprocessor [true / false] +; allows dsp functionality below but incurs CPU overhead +preprocess => false + +; preproc voice activity detection [true / false] +; more advanced equivalent of DTX, based on voice frequencies +pp_vad => false + +; preproc automatic gain control [true / false] +pp_agc => false +pp_agc_level => 8000 + +; preproc denoiser [true / false] +pp_denoise => false + +; preproc dereverb [true / false] +pp_dereverb => false +pp_dereverb_decay => 0.4 +pp_dereverb_level => 0.3 + + +[plc] +; for all codecs which do not support native PLC +; this determines whether to perform generic PLC +; there is a minor performance penalty for this +genericplc => true diff --git a/1.4.23-rc4/configs/dnsmgr.conf.sample b/1.4.23-rc4/configs/dnsmgr.conf.sample new file mode 100644 index 000000000..e34dbcf0a --- /dev/null +++ b/1.4.23-rc4/configs/dnsmgr.conf.sample @@ -0,0 +1,5 @@ +[general] +;enable=yes ; enable creation of managed DNS lookups + ; default is 'no' +;refreshinterval=1200 ; refresh managed DNS lookups every seconds + ; default is 300 (5 minutes) \ No newline at end of file diff --git a/1.4.23-rc4/configs/dundi.conf.sample b/1.4.23-rc4/configs/dundi.conf.sample new file mode 100644 index 000000000..a1e999726 --- /dev/null +++ b/1.4.23-rc4/configs/dundi.conf.sample @@ -0,0 +1,239 @@ +; +; DUNDi configuration file +; +; For more information about DUNDi, see http://www.dundi.com +; +; +[general] +; +; The "general" section contains general parameters relating +; to the operation of the dundi client and server. +; +; The first part should be your complete contact information +; should someone else in your peer group need to contact you. +; +;department=Your Department +;organization=Your Company, Inc. +;locality=Your City +;stateprov=ST +;country=US +;email=your@email.com +;phone=+12565551212 +; +; +; Specify bind address and port number. Default is +; 4520 +; +;bindaddr=0.0.0.0 +;port=4520 +; +; Our entity identifier (Should generally be the MAC address of the +; machine it's running on. Defaults to the first eth address, but you +; can override it here, as long as you set it to the MAC of *something* +; you own!) +; +;entityid=00:07:E9:3B:76:60 +; +; Peers shall cache our query responses for the specified time, +; given in seconds. Default is 3600. +; +;cachetime=3600 +; +; This defines the max depth in which to search the DUNDi system. +; Note that the maximum time that we will wait for a response is +; (2000 + 200 * ttl) ms. +; +ttl=32 +; +; If we don't get ACK to our DPDISCOVER within 2000ms, and autokill is set +; to yes, then we cancel the whole thing (that's enough time for one +; retransmission only). This is used to keep things from stalling for a long +; time for a host that is not available, but would be ill advised for bad +; connections. In addition to 'yes' or 'no' you can also specify a number +; of milliseconds. See 'qualify' for individual peers to turn on for just +; a specific peer. +; +autokill=yes +; +; pbx_dundi creates a rotating key called "secret", under the family +; 'secretpath'. The default family is dundi (resulting in +; the key being held at dundi/secret). +; +;secretpath=dundi +; +; The 'storehistory' option (also changeable at runtime with +; 'dundi store history' and 'dundi no store history') will +; cause the DUNDi engine to keep track of the last several +; queries and the amount of time each query took to execute +; for the purpose of tracking slow nodes. This option is +; off by default due to performance impacts. +; +;storehistory=yes + +[mappings] +; +; The "mappings" section maps DUNDi contexts +; to contexts on the local asterisk system. Remember +; that numbers that are made available under the e164 +; DUNDi context are regulated by the DUNDi General Peering +; Agreement (GPA) if you are a member of the DUNDi E.164 +; Peering System. +; +; dundi_context => local_context,weight,tech,dest[,options]] +; +; 'dundi_context' is the name of the context being requested +; within the DUNDi request +; +; 'local_context' is the name of the context on the local system +; in which numbers can be looked up for which responses shall be given. +; +; 'weight' is the weight to use for the responses provided from this +; mapping. The number must be >= 0 and < 60000. Since it is totally +; valid to receive multiple responses to a query, responses received +; with a lower weight are tried first. Note that the weight has a +; special meaning in the e164 context - see the GPA for more details. +; +; 'tech' is the technology to use (IAX, SIP, H323) +; +; 'dest' is the destination to supply for reaching that number. The +; following variables can be used in the destination string and will +; be automatically substituted: +; ${NUMBER}: The number being requested +; ${IPADDR}: The IP address to connect to +; ${SECRET}: The current rotating secret key to be used +; +; Further options may include: +; +; nounsolicited: No unsolicited calls of any type permitted via this +; route +; nocomunsolicit: No commercial unsolicited calls permitted via +; this route +; residential: This number is known to be a residence +; commercial: This number is known to be a business +; mobile: This number is known to be a mobile phone +; nocomunsolicit: No commercial unsolicited calls permitted via +; this route +; nopartial: Do not search for partial matches +; +; There *must* exist an entry in mappings for DUNDi to respond +; to any request, although it may be empty. +; +;e164 => dundi-e164-canonical,0,IAX2,dundi:${SECRET}@${IPADDR}/${NUMBER},nounsolicited,nocomunsolicit,nopartial +;e164 => dundi-e164-customers,100,IAX2,dundi:${SECRET}@${IPADDR}/${NUMBER},nounsolicited,nocomunsolicit,nopartial +;e164 => dundi-e164-via-pstn,400,IAX2,dundi:${SECRET}@${IPADDR}/${NUMBER},nounsolicited,nocomunsolicit,nopartial + +;digexten => default,0,IAX2,guest@lappy/${NUMBER} +;asdf => + + +; +; +; The remaining sections represent the peers +; that we fundamentally trust. The section name +; represents the name and optionally at a specific +; DUNDi context if you want the trust to be established +; for only a specific DUNDi context. +; +; inkey - What key they will be authenticating to us with +; +; outkey - What key we use to authenticate to them +; +; host - What their host is +; +; order - What search order to use. May be 'primary', 'secondary', +; 'tertiary' or 'quartiary'. In large systems, it is beneficial +; to only query one up-stream host in order to maximize caching +; value. Adding one with primary and one with secondary gives you +; redundancy without sacrificing performance. +; +; include - Includes this peer when searching a particular context +; for lookup (set "all" to perform all lookups with that +; host. This is also the context in which peers are permitted +; to precache. +; +; noinclude - Disincludes this peer when searching a particular context +; for lookup (set "all" to perform no lookups with that +; host. +; +; permit - Permits this peer to search a given DUNDi context on +; the local system. Set "all" to permit this host to +; lookup all contexts. This is also a context for which +; we will create/forward PRECACHE commands. +; +; deny - Denies this peer to search a given DUNDi context on +; the local system. Set "all" to deny this host to +; lookup all contexts. +; +; model - inbound, outbound, or symmetric for whether we receive +; requests only, transmit requests only, or do both. +; +; precache - Utilize/Permit precaching with this peer (to pre +; cache means to provide an answer when no request +; was made and is used so that machines with few +; routes can push those routes up a to a higher level). +; outgoing means we send precache routes to this peer, +; incoming means we permit this peer to send us +; precache routes. symmetric means we do both. +; +; Note: You cannot mix symmetric/outbound model with symmetric/inbound +; precache, nor can you mix symmetric/inbound model with symmetric/outbound +; precache. +; +; +; The '*' peer is special and matches an unspecified entity +; + +; +; Sample Primary e164 DUNDi peer +; +;[00:50:8B:F3:75:BB] +;model = symmetric +;host = 64.215.96.114 +;inkey = digium +;outkey = misery +;include = e164 +;permit = e164 +;qualify = yes + +; +; Sample Secondary e164 DUNDi peer +; +;[00:A0:C9:96:92:84] +;model = symmetric +;host = misery.digium.com +;inkey = misery +;outkey = ourkey +;include = e164 +;permit = e164 +;qualify = yes +;order = secondary + +; +; Sample "push mode" downstream host +; +;[00:0C:76:96:75:28] +;model = inbound +;host = dynamic +;precache = inbound +;inkey = littleguy +;outkey = ourkey +;include = e164 ; In this case used only for precaching +;permit = e164 +;qualify = yes + +; +; Sample "push mode" upstream host +; +;[00:07:E9:3B:76:60] +;model = outbound +;precache = outbound +;host = 216.207.245.34 +;register = yes +;inkey = dhcp34 +;permit = all ; In this case used only for precaching +;include = all +;qualify = yes +;outkey=foo + +;[*] +; diff --git a/1.4.23-rc4/configs/enum.conf.sample b/1.4.23-rc4/configs/enum.conf.sample new file mode 100644 index 000000000..39c723175 --- /dev/null +++ b/1.4.23-rc4/configs/enum.conf.sample @@ -0,0 +1,22 @@ +; +; ENUM Configuration for resolving phone numbers over DNS +; +; Sample config for Asterisk +; This file is reloaded at "module reload enum" in the CLI +; +[general] +; +; The search list for domains may be customized. Domains are searched +; in the order they are listed here. +; +search => e164.arpa +; +; If you'd like to use the E.164.org public ENUM registry in addition +; to the official e164.arpa one, uncomment the following line +; +;search => e164.org +; +; As there are more H323 drivers available you have to select to which +; drive a H323 URI will map. Default is "H323". +; +h323driver => H323 diff --git a/1.4.23-rc4/configs/extconfig.conf.sample b/1.4.23-rc4/configs/extconfig.conf.sample new file mode 100644 index 000000000..fa9462b63 --- /dev/null +++ b/1.4.23-rc4/configs/extconfig.conf.sample @@ -0,0 +1,59 @@ +; +; Static and realtime external configuration +; engine configuration +; +; Please read doc/extconfig.txt for basic table +; formatting information. +; +[settings] +; +; Static configuration files: +; +; file.conf => driver,database[,table] +; +; maps a particular configuration file to the given +; database driver, database and table (or uses the +; name of the file as the table if not specified) +; +;uncomment to load queues.conf via the odbc engine. +; +;queues.conf => odbc,asterisk,ast_config +; +; The following files CANNOT be loaded from Realtime storage: +; asterisk.conf +; extconfig.conf (this file) +; logger.conf +; +; Additionally, the following files cannot be loaded from +; Realtime storage unless the storage driver is loaded +; early using 'preload' statements in modules.conf: +; manager.conf +; cdr.conf +; rtp.conf +; +; +; Realtime configuration engine +; +; maps a particular family of realtime +; configuration to a given database driver, +; database and table (or uses the name of +; the family if the table is not specified +; +;example => odbc,asterisk,alttable +; +; "odbc" is shown in the examples below, but is not the only valid realtime +; engine. There is: +; odbc ... res_config_odbc +; pgsql ... res_config_pgsql +; mysql ... res_config_mysql (available from asterisk-addons) +; +;iaxusers => odbc,asterisk +;iaxpeers => odbc,asterisk +;sipusers => odbc,asterisk +;sippeers => odbc,asterisk +;voicemail => odbc,asterisk +;extensions => odbc,asterisk +;queues => odbc,asterisk +;queue_members => odbc,asterisk +;meetme => mysql,conferences + diff --git a/1.4.23-rc4/configs/extensions.ael.sample b/1.4.23-rc4/configs/extensions.ael.sample new file mode 100644 index 000000000..ab8cdd854 --- /dev/null +++ b/1.4.23-rc4/configs/extensions.ael.sample @@ -0,0 +1,448 @@ +// +// Example AEL config file +// +// +// Static extension configuration file, used by +// the pbx_ael module. This is where you configure all your +// inbound and outbound calls in Asterisk. +// +// This configuration file is reloaded +// - With the "ael reload" command in the CLI +// - With the "reload" command (that reloads everything) in the CLI + +// The "Globals" category contains global variables that can be referenced +// in the dialplan by using the GLOBAL dialplan function: +// ${GLOBAL(VARIABLE)} +// ${${GLOBAL(VARIABLE)}} or ${text${GLOBAL(VARIABLE)}} or any hybrid +// Unix/Linux environmental variables are reached with the ENV dialplan +// function: ${ENV(VARIABLE)} +// + +globals { + CONSOLE="Console/dsp"; // Console interface for demo + //CONSOLE=Zap/1 + //CONSOLE=Phone/phone0 + IAXINFO=guest; // IAXtel username/password + //IAXINFO="myuser:mypass"; + TRUNK="Zap/g2"; // Trunk interface + // + // Note the 'g2' in the TRUNK variable above. It specifies which group (defined + // in chan_dahdi.conf) to dial, i.e. group 2, and how to choose a channel to use in + // the specified group. The four possible options are: + // + // g: select the lowest-numbered non-busy Zap channel + // (aka. ascending sequential hunt group). + // G: select the highest-numbered non-busy Zap channel + // (aka. descending sequential hunt group). + // r: use a round-robin search, starting at the next highest channel than last + // time (aka. ascending rotary hunt group). + // R: use a round-robin search, starting at the next lowest channel than last + // time (aka. descending rotary hunt group). + // + TRUNKMSD=1; // MSD digits to strip (usually 1 or 0) + //TRUNK=IAX2/user:pass@provider +}; + +// +// Any category other than "General" and "Globals" represent +// extension contexts, which are collections of extensions. +// +// Extension names may be numbers, letters, or combinations +// thereof. If an extension name is prefixed by a '_' +// character, it is interpreted as a pattern rather than a +// literal. In patterns, some characters have special meanings: +// +// X - any digit from 0-9 +// Z - any digit from 1-9 +// N - any digit from 2-9 +// [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9) +// . - wildcard, matches anything remaining (e.g. _9011. matches +// anything starting with 9011 excluding 9011 itself) +// ! - wildcard, causes the matching process to complete as soon as +// it can unambiguously determine that no other matches are possible +// +// For example the extension _NXXXXXX would match normal 7 digit dialings, +// while _1NXXNXXXXXX would represent an area code plus phone number +// preceded by a one. +// +// Each step of an extension is ordered by priority, which must +// always start with 1 to be considered a valid extension. The priority +// "next" or "n" means the previous priority plus one, regardless of whether +// the previous priority was associated with the current extension or not. +// The priority "same" or "s" means the same as the previously specified +// priority, again regardless of whether the previous entry was for the +// same extension. Priorities may be immediately followed by a plus sign +// and another integer to add that amount (most useful with 's' or 'n'). +// Priorities may then also have an alias, or label, in +// parenthesis after their name which can be used in goto situations +// +// Contexts contain several lines, one for each step of each +// extension, which can take one of two forms as listed below, +// with the first form being preferred. One may include another +// context in the current one as well, optionally with a +// date and time. Included contexts are included in the order +// they are listed. +// +//context name { +// exten-name => { +// application(arg1,arg2,...); +// +// Timing list for includes is +// +//