diff options
| author | seanbright <seanbright@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-05-28 14:39:21 +0000 |
|---|---|---|
| committer | seanbright <seanbright@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-05-28 14:39:21 +0000 |
| commit | a22b4735e5a6b8745a4915a260995886c56c7ffe (patch) | |
| tree | 2e8e77235c0fb39f0551db5e6012057fc8c580d0 /configs | |
| parent | 7f7cfd42e9d0e77e3d1ea18732595c1794bf46ea (diff) | |
Remove a bunch of trailing whitespace in preparation for reformatting/cleanup.
Let's try that again, this time removing trailing whitespace and not leading
whitespace. I can't believe no one noticed.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@197535 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'configs')
57 files changed, 1730 insertions, 1730 deletions
diff --git a/configs/agents.conf.sample b/configs/agents.conf.sample index a006d275a..3ac313322 100644 --- a/configs/agents.conf.sample +++ b/configs/agents.conf.sample @@ -32,14 +32,14 @@ persistentagents=yes ; 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". +; Default is "no". ; ;autologoffunavail=yes ; ; Define ackcall to require a DTMF acknowledgement 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. Use the acceptdtmf option to +; agents to acknowledge calls. Use the acceptdtmf option to ; configure what DTMF key press should be used to acknowledge the ; call. The default is '#'. ; @@ -70,14 +70,14 @@ persistentagents=yes ; ;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 +; 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= @@ -85,7 +85,7 @@ persistentagents=yes ; -------------------------------------------------- ; 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 ; @@ -100,7 +100,7 @@ persistentagents=yes ; /var/spool/asterisk/monitor ;savecallsin=/var/calls ; -; An optional custom beep sound file to play to always-connected agents. +; An optional custom beep sound file to play to always-connected agents. ;custom_beep=beep ; ; -------------------------------------------------- diff --git a/configs/ais.conf.sample b/configs/ais.conf.sample index f0bccc639..a4428891f 100644 --- a/configs/ais.conf.sample +++ b/configs/ais.conf.sample @@ -1,5 +1,5 @@ ; -; Sample configuration file for res_ais +; Sample configuration file for res_ais ; * SAForum AIS (Application Interface Specification) ; ; More information on the AIS specification is available from the SAForum. @@ -76,7 +76,7 @@ ; ; This example would be used for a node that has phones directly registered ; to it, but does not have direct access to voicemail. So, this node wants -; to be informed about MWI state changes on other voicemail server nodes, but +; to be informed about MWI state changes on other voicemail server nodes, but ; is not capable of publishing any state changes. ; ; [mwi] diff --git a/configs/alarmreceiver.conf.sample b/configs/alarmreceiver.conf.sample index 0ad23f8fc..796470181 100644 --- a/configs/alarmreceiver.conf.sample +++ b/configs/alarmreceiver.conf.sample @@ -7,7 +7,7 @@ [general] -; +; ; Specify a timestamp format for the metadata section of the event files ; Default is %a %b %d, %Y @ %H:%M:%S %Z @@ -32,7 +32,7 @@ timestampformat = %a %b %d, %Y @ %H:%M:%S %Z 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. @@ -46,7 +46,7 @@ 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. +; like 10000 milliseconds. fdtimeout = 2000 @@ -54,7 +54,7 @@ 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. +; like 4000 milliseconds. ; sdtimeout = 200 diff --git a/configs/alsa.conf.sample b/configs/alsa.conf.sample index 33c5a3fa8..f55030618 100644 --- a/configs/alsa.conf.sample +++ b/configs/alsa.conf.sample @@ -39,23 +39,23 @@ extension=s ;------------------------------ 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. + ; 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. + ; 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. + ; 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/configs/amd.conf.sample b/configs/amd.conf.sample index e25c18e18..ce4808a0c 100644 --- a/configs/amd.conf.sample +++ b/configs/amd.conf.sample @@ -4,15 +4,15 @@ [general] initial_silence = 2500 ; Maximum silence duration before the greeting. -; If exceeded then MACHINE. + ; 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 + ; If exceeded then HUMAN total_analysis_time = 5000 ; Maximum time allowed for the algorithm to decide -; on a HUMAN or MACHINE + ; 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 + ; the audio what follows as a new word maximum_number_of_words = 3 ; Maximum number of words in the greeting. -; If exceeded then MACHINE + ; If exceeded then MACHINE silence_threshold = 256 diff --git a/configs/asterisk.adsi b/configs/asterisk.adsi index 396de2c75..a58952589 100644 --- a/configs/asterisk.adsi +++ b/configs/asterisk.adsi @@ -35,39 +35,39 @@ DISPLAY "empty" IS "asdf" ; Begin soft key definitions ; KEY "callfwd" IS "CallFwd" OR "Call Forward" -OFFHOOK -VOICEMODE -WAITDIALTONE -SENDDTMF "*60" -GOTO "offHook" + OFFHOOK + VOICEMODE + WAITDIALTONE + SENDDTMF "*60" + GOTO "offHook" ENDKEY KEY "vmail_OH" IS "VMail" OR "Voicemail" -OFFHOOK -VOICEMODE -WAITDIALTONE -SENDDTMF "8500" + OFFHOOK + VOICEMODE + WAITDIALTONE + SENDDTMF "8500" ENDKEY KEY "vmail" IS "VMail" OR "Voicemail" -SENDDTMF "8500" + SENDDTMF "8500" ENDKEY KEY "backspace" IS "BackSpc" OR "Backspace" -BACKSPACE + BACKSPACE ENDKEY KEY "cwdisable" IS "CWDsble" OR "Disable Call Wait" -SENDDTMF "*70" -SETFLAG "nocallwaiting" -SHOWDISPLAY "cwdisabled" AT 4 -TIMERCLEAR -TIMERSTART 1 + SENDDTMF "*70" + SETFLAG "nocallwaiting" + SHOWDISPLAY "cwdisabled" AT 4 + TIMERCLEAR + TIMERSTART 1 ENDKEY KEY "cidblock" IS "CIDBlk" OR "Block Callerid" -SENDDTMF "*67" -SETFLAG "nocallwaiting" + SENDDTMF "*67" + SETFLAG "nocallwaiting" ENDKEY ; @@ -75,85 +75,85 @@ ENDKEY ; 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 + 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 + 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 + 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 + IFEVENT REORDER THEN + SHOWDISPLAY "callended" AT 2 + ENDIF ENDSUB diff --git a/configs/cdr.conf.sample b/configs/cdr.conf.sample index 195f88f32..0c0413163 100644 --- a/configs/cdr.conf.sample +++ b/configs/cdr.conf.sample @@ -14,12 +14,12 @@ ;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 +; 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 +; channel on the other, and then one CDR for each channel attempted. This may seem ; redundant, but cannot be helped. ;unanswered = no @@ -67,7 +67,7 @@ ; Normally, the 'billsec' field logged to the backends (text files or databases) ; is simply the end time (hangup time) minus the answer time in seconds. Internally, -; asterisk stores the time in terms of microseconds and seconds. By setting +; asterisk stores the time in terms of microseconds and seconds. By setting ; initiatedseconds to 'yes', you can force asterisk to report any seconds ; that were initiated (a sort of round up method). Technically, this is ; when the microsecond part of the end time is greater than the microsecond @@ -78,19 +78,19 @@ ; ; 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 +; 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 +; +; 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 +; 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 @@ -104,7 +104,7 @@ ; 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 +; 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. ; @@ -126,7 +126,7 @@ ; 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 +; 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. @@ -135,7 +135,7 @@ ; 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' +; 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 diff --git a/configs/chan_dahdi.conf.sample b/configs/chan_dahdi.conf.sample index 76771fb3e..6d9847d2a 100644 --- a/configs/chan_dahdi.conf.sample +++ b/configs/chan_dahdi.conf.sample @@ -6,7 +6,7 @@ ; will reload the configuration file, but not all configuration options ; are re-configured during a reload (signalling, as well as PRI and ; SS7-related settings cannot be changed on a reload). -; +; ; This file documents many configuration variables. Normally unless you know ; what a variable means or that it should be changed, there's no reason to ; un-comment those lines. @@ -21,11 +21,11 @@ ; ; Trunk groups are used for NFAS or GR-303 connections. ; -; Group: Defines a trunk group. +; Group: Defines a trunk group. ; trunkgroup => <trunkgroup>,<dchannel>[,<backup1>...] ; ; trunkgroup is the numerical trunk group to create -; dchannel is the DAHDI channel which will have the +; dchannel is the DAHDI channel which will have the ; d-channel for the trunk. ; backup1 is an optional list of backup d-channels. ; @@ -85,7 +85,7 @@ ; example, if you set 'national', you will be unable to dial local or ; international numbers. ; -; PRI Local Dialplan: Only RARELY used for PRI (sets the calling number's +; PRI Local Dialplan: Only RARELY used for PRI (sets the calling number's ; numbering plan). In North America, the typical use is sending the 10 digit ; callerID number and setting the prilocaldialplan to 'national' (the default). ; Only VERY rarely will you need to change this. @@ -98,12 +98,12 @@ ; national: National ISDN ; international: International ISDN ; dynamic: Dynamically selects the appropriate dialplan -; redundant: Same as dynamic, except that the underlying number is not +; redundant: Same as dynamic, except that the underlying number is not ; changed (not common) ; ;pridialplan=unknown ;prilocaldialplan=national -; +; ; pridialplan may be also set at dialtime, by prefixing the dialled number with ; one of the following letters: ; U - Unknown @@ -133,27 +133,27 @@ ; ; PRI caller ID prefixes based on the given TON/NPI (dialplan) ; This is especially needed for EuroISDN E1-PRIs -; +; ; None of the prefix settings can be changed on reload. ; -; sample 1 for Germany +; sample 1 for Germany ;internationalprefix = 00 ;nationalprefix = 0 ;localprefix = 0711 ;privateprefix = 07115678 -;unknownprefix = +;unknownprefix = ; -; sample 2 for Germany +; sample 2 for Germany ;internationalprefix = + ;nationalprefix = +49 ;localprefix = +49711 ;privateprefix = +497115678 -;unknownprefix = +;unknownprefix = ; ; PRI resetinterval: sets the time in seconds between restart of unused ; B channels; defaults to 'never'. ; -;resetinterval = 3600 +;resetinterval = 3600 ; ; Overlap dialing mode (sending overlap digits) ; Cannot be changed on a reload. @@ -168,7 +168,7 @@ ; 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 (default) ; @@ -206,7 +206,7 @@ ; 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, +; 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) @@ -284,11 +284,11 @@ ; (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. -; -; outsignalling can only be one of: +; +; outsignalling can only be one of: ; em, em_e1, em_w, sf, sf_w, sf_featd, sf_featdmf, sf_featb, featd, ; featdmf, featdmf_ta, e911, fgccama, fgccamamf -; +; ; outsignalling cannot be changed on a reload. ; ;signalling=featdmf @@ -318,9 +318,9 @@ ; None of them will update on a reload. ; ; How long generated tones (DTMF and MF) will be played on the channel -; (in milliseconds). +; (in milliseconds). ; -; This is a global, rather than a per-channel setting. It will not be +; This is a global, rather than a per-channel setting. It will not be ; updated on a reload. ; ;toneduration=100 @@ -354,7 +354,7 @@ usecallerid=yes ; What signals the start of caller ID ; ring = a ring signals the start (default) ; polarity = polarity reversal signals the start -; polarity_IN = polarity reversal signals the start, for India, +; polarity_IN = polarity reversal signals the start, for India, ; for dtmf dialtone detection; using DTMF. ; (see doc/India-CID.txt) ; @@ -381,7 +381,7 @@ usecallerid=yes ; fsk,rpas - the FXO line is monitored for MWI FSK spills preceded ; by a ring pulse alert signal. ; neon - The fxo line is monitored for the presence of NEON pulses -; indicating MWI. +; indicating MWI. ; When detected, an internal Asterisk MWI event is generated so that any other ; part of Asterisk that cares about MWI state changes is notified, just as if ; the state change came from app_voicemail. @@ -432,7 +432,7 @@ usecallingpres=yes ; ; Some countries (UK) have ring tones with different ring tones (ring-ring), ; which means the caller ID needs to be set later on, and not just after -; the first ring, as per the default (1). +; the first ring, as per the default (1). ; ;sendcalleridafter = 2 ; @@ -472,10 +472,10 @@ cancallforward=yes ; 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 +; 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. +; 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. @@ -486,9 +486,9 @@ callreturn=yes ; ; for any other voicemail context, the following will produce the stutter tone: ; -;mailbox=1234@context +;mailbox=1234@context ; -; Enable echo cancellation +; Enable echo cancellation ; Use either "yes", "no", or a power of two from 32 to 256 if you wish to ; actually set the number of taps of cancellation. ; @@ -552,7 +552,7 @@ echocancelwhenbridged=yes ; ; There are several independent gain settings: ; rxgain: gain for the rx (receive - into Asterisk) channel. Default: 0.0 -; txgain: gain for the tx (transmit - out of Asterisk Asterisk) channel. +; txgain: gain for the tx (transmit - out of Asterisk Asterisk) channel. ; Default: 0.0 ; cid_rxgain: set the gain just for the caller ID sounds Asterisk ; emits. Default: 5.0 . @@ -581,9 +581,9 @@ pickupgroup=1 ; Channel variable to be set for all calls from this channel ;setvar=CHANNEL=42 ;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will -; cause the given audio file to -; be played upon completion of -; an attended transfer. + ; cause the given audio file to + ; be played upon completion of + ; an attended transfer. ; ; Specify whether the channel should be answered immediately or if the simple @@ -600,10 +600,10 @@ pickupgroup=1 ; ; caller ID can be set to "asreceived" or a specific number if you want to ; override it. Note that "asreceived" only applies to trunk interfaces. -; fullname sets just the +; fullname sets just the ; ; fullname: sets just the name part. -; cid_number: sets just the number part: +; cid_number: sets just the number part: ; ;callerid = 123456 ; @@ -642,7 +642,7 @@ pickupgroup=1 ;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 +; 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. ; @@ -685,8 +685,8 @@ pickupgroup=1 ; ;hanguponpolarityswitch=yes ; -; polarityonanswerdelay: minimal time period (ms) between the answer -; polarity switch and hangup polarity switch. +; polarityonanswerdelay: minimal time period (ms) between the answer +; polarity switch and hangup polarity switch. ; (default: 600ms) ; ; On trunk interfaces (FXS) it can be useful to attempt to follow the progress @@ -699,7 +699,7 @@ pickupgroup=1 ; with "progzone". ; ; progzone also affects the pattern used for buzydetect (unless -; busypattern is set explicitly). The possible values are: +; busypattern is set explicitly). The possible values are: ; us (default) ; ca (alias for 'us') ; cr (Costa Rica) @@ -741,7 +741,7 @@ pickupgroup=1 ;faxdetect=no ; ; When 'faxdetect' is used, one could use 'faxbuffers' to configure the DAHDI -; transmit buffer policy. The default is *OFF*. When this configuration +; transmit buffer policy. The default is *OFF*. When this configuration ; option is used, the faxbuffer policy will be used for the life of the call ; after a fax tone is detected. The faxbuffer policy is reverted after the ; call is torn down. The sample below will result in 6 buffers and a full @@ -792,23 +792,23 @@ pickupgroup=1 ; ;------------------------------ 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. + ; 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. + ; 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. + ; 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". ;----------------------------------------------------------------------------------- @@ -834,7 +834,7 @@ pickupgroup=1 ; 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.: +; trunk group followed by a colon, e.g.: ; ; crv => 1:1 ; crv => 2:1-2,5-8 @@ -908,15 +908,15 @@ pickupgroup=1 ; A range of -1 will force it to always match. ; Anything lower than -1 would presumably cause it to never match. ; -;dring1=95,0,0 -;dring1context=internal1 +;dring1=95,0,0 +;dring1context=internal1 ;dring1range=10 -;dring2=325,95,0 -;dring2context=internal2 +;dring2=325,95,0 +;dring2context=internal2 ;dring2range=10 ; If no pattern is matched here is where we go. ;context=default -;channel => 1 +;channel => 1 ; ---------------- Options for use with signalling=ss7 ----------------- ; None of them can be changed by a reload. @@ -945,12 +945,12 @@ pickupgroup=1 ; ;ss7_calling_nai=dynamic ; -; -; sample 1 for Germany +; +; sample 1 for Germany ;ss7_internationalprefix = 00 ;ss7_nationalprefix = 0 -;ss7_subscriberprefix = -;ss7_unknownprefix = +;ss7_subscriberprefix = +;ss7_unknownprefix = ; ; This option is used to disable automatic sending of ACM when the call is started @@ -1056,7 +1056,7 @@ pickupgroup=1 ; 'stack' is for very verbose output of the channel and context call stack, only useful ; if you are debugging a crash or want to learn how the library works. The stack logging ; will be only enabled if the openr2 library was compiled with -DOR2_TRACE_STACKS -; You can mix up values, like: loglevel=error,debug,mf to log just error, debug and +; You can mix up values, like: loglevel=error,debug,mf to log just error, debug and ; multi frequency messages ; 'all' is a special value to log all the activity ; 'nothing' is a clean-up value, in case you want to not log any activity for @@ -1110,20 +1110,20 @@ pickupgroup=1 ; You most likely dont need this feature. Default is yes. ; When this is set to yes, all calls that are offered (incoming calls) which -; DNIS is valid (exists in extensions.conf) and pass collect call validation +; DNIS is valid (exists in extensions.conf) and pass collect call validation ; will be accepted with a Group B tone (either call with charge or not, depending on mfcr2_charge_calls) ; with this set to 'no' then the call will NOT be accepted on offered, and the call will start its ; execution in extensions.conf without being accepted until the channel is answered (either with Answer() or -; any other application resulting in the channel being answered). +; any other application resulting in the channel being answered). ; This can be set to 'no' if your telco or PBX needs the hangup cause to be set accurately ; when this option is set to no you must explicitly accept the call with DAHDIAcceptR2Call -; or implicitly through the Answer() application. +; or implicitly through the Answer() application. ; mfcr2_accept_on_offer=yes ; WARNING: advanced users only! I really mean it ; this parameter is commented by default because ; YOU DON'T NEED IT UNLESS YOU REALLY GROK MFC/R2 -; READ COMMENTS on doc/r2proto.conf in openr2 package +; READ COMMENTS on doc/r2proto.conf in openr2 package ; for more info ; mfcr2_advanced_protocol_file=/path/to/r2proto.conf @@ -1171,7 +1171,7 @@ pickupgroup=1 ; chan_dahdi.conf and [general] in users.conf - one section's configuration ; does not affect another one's. ; -; Instead of letting common configuration values "slide through" you can +; Instead of letting common configuration values "slide through" you can ; use configuration templates to easily keep the common part in one ; place and override where needed. ; diff --git a/configs/cli_aliases.conf.sample b/configs/cli_aliases.conf.sample index cc1e2e6d3..1d9cd9107 100644 --- a/configs/cli_aliases.conf.sample +++ b/configs/cli_aliases.conf.sample @@ -13,8 +13,8 @@ template = friendly ; By default, include friendly aliases ;template = asterisk12 ; Asterisk 1.2 style syntax ;template = asterisk14 ; Asterisk 1.4 style syntax ;template = individual_custom ; see [individual_custom] example below which -; includes a list of aliases from an external -; file + ; includes a list of aliases from an external + ; file ; Because the Asterisk CLI syntax follows a "module verb argument" syntax, @@ -70,7 +70,7 @@ pri intense debug span=pri set debug 2 span ; by Asterisk. If you wish to use the provided templates, simply define the ; context name which does not utilize the '_tpl' at the end. For example, ; if you would like to use the Asterisk 1.2 style syntax, define in the -; [general] section +; [general] section [asterisk12_tpl](!) show channeltypes=core show channeltypes @@ -92,7 +92,7 @@ show file formats=core show file formats show applications=core show applications show functions=core show functions show switches=core show switches -show hints=core show hints +show hints=core show hints show globals=core show globals show function=core show function show application=core show application @@ -102,7 +102,7 @@ show codecs=core show codecs show audio codecs=core show audio codecs show video codecs=core show video codecs show image codecs=core show image codecs -show codec=core show codec +show codec=core show codec moh classes show=moh show classes moh files show=moh show files agi no debug=agi debug off diff --git a/configs/cli_permissions.conf.sample b/configs/cli_permissions.conf.sample index 7cbad88f3..4a6973f50 100644 --- a/configs/cli_permissions.conf.sample +++ b/configs/cli_permissions.conf.sample @@ -23,7 +23,7 @@ [general] default_perm=permit ; To leave asterisk working as normal -; we should set this parameter to 'permit' + ; we should set this parameter to 'permit' ; ; Follows the per-users permissions configs. ; diff --git a/configs/console.conf.sample b/configs/console.conf.sample index d7e586a6b..9bd502696 100644 --- a/configs/console.conf.sample +++ b/configs/console.conf.sample @@ -5,7 +5,7 @@ [general] ; Set this option to "yes" to enable automatically answering calls on the -; console. This is very useful if the console is used as an intercom. +; console. This is very useful if the console is used as an intercom. ; The default value is "no". ; ;autoanswer = no @@ -21,7 +21,7 @@ ;extension = s ; Set the default CallerID for created channels. -; +; ;callerid = MyName Here <(256) 428-6000> ; Set the default language for created channels. @@ -34,7 +34,7 @@ ; The default is "no". ; ;overridecontext = no ; if 'no', the last @ will start the context -; if 'yes' the whole string is an extension. + ; if 'yes' the whole string is an extension. ; Default Music on Hold class to use when this channel is placed on hold in @@ -46,23 +46,23 @@ ;------------------------------ JITTER BUFFER CONFIGURATION -------------------------- ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an -; Console 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 Console channel can't accept jitter, -; thus an enabled jitterbuffer on the receive Console side will always -; be used if the sending side can create jitter. + ; Console 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 Console channel can't accept jitter, + ; thus an enabled jitterbuffer on the receive Console 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. + ; 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 Console -; 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. + ; 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". ;----------------------------------------------------------------------------------- @@ -76,8 +76,8 @@ [default] input_device = default ; When configuring an input device and output device, output_device = default ; use the name that you see when you run the "console -; list available" CLI command. If you say "default", the -; system default input and output devices will be used. + ; list available" CLI command. If you say "default", the + ; system default input and output devices will be used. autoanswer = no context = default extension = s @@ -86,5 +86,5 @@ language = en overridecontext = no mohinterpret = default active = yes ; This option should only be set for one console. -; It means that it is the active console to be -; used from the Asterisk CLI. + ; It means that it is the active console to be + ; used from the Asterisk CLI. diff --git a/configs/dnsmgr.conf.sample b/configs/dnsmgr.conf.sample index a2939dc10..e34dbcf0a 100644 --- a/configs/dnsmgr.conf.sample +++ b/configs/dnsmgr.conf.sample @@ -1,5 +1,5 @@ [general] ;enable=yes ; enable creation of managed DNS lookups -; default is 'no' + ; default is 'no' ;refreshinterval=1200 ; refresh managed DNS lookups every <n> seconds -; default is 300 (5 minutes)
\ No newline at end of file + ; default is 300 (5 minutes)
\ No newline at end of file diff --git a/configs/dundi.conf.sample b/configs/dundi.conf.sample index 3eb1bd320..1b6a174c0 100644 --- a/configs/dundi.conf.sample +++ b/configs/dundi.conf.sample @@ -1,6 +1,6 @@ ; ; DUNDi configuration file -; +; ; For more information about DUNDi, see http://www.dundi.com ; ; @@ -50,9 +50,9 @@ 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 +; 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 +; 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. @@ -60,7 +60,7 @@ ttl=32 autokill=yes ; ; pbx_dundi creates a rotating key called "secret", under the family -; 'secretpath'. The default family is dundi (resulting in +; 'secretpath'. The default family is dundi (resulting in ; the key being held at dundi/secret). ; ;secretpath=dundi @@ -78,8 +78,8 @@ autokill=yes ; ; 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 +; 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. ; @@ -108,14 +108,14 @@ autokill=yes ; ; Further options may include: ; -; nounsolicited: No unsolicited calls of any type permitted via this +; nounsolicited: No unsolicited calls of any type permitted via this ; route -; nocomunsolicit: No commercial unsolicited calls permitted via +; 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 +; nocomunsolicit: No commercial unsolicited calls permitted via ; this route ; nopartial: Do not search for partial matches ; @@ -163,7 +163,7 @@ autokill=yes ; ; host - What their host is ; -; order - What search order to use. May be 'primary', 'secondary', +; 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 @@ -187,7 +187,7 @@ autokill=yes ; the local system. Set "all" to deny this host to ; lookup all contexts. ; -; model - inbound, outbound, or symmetric for whether we receive +; 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 @@ -241,7 +241,7 @@ autokill=yes ;inkey = littleguy ;outkey = ourkey ;include = e164 ; In this case used only for precaching -;permit = e164 +;permit = e164 ;qualify = yes ; @@ -254,7 +254,7 @@ autokill=yes ;register = yes ;inkey = dhcp34 ;permit = all ; In this case used only for precaching -;include = all +;include = all ;qualify = yes ;outkey=foo diff --git a/configs/extconfig.conf.sample b/configs/extconfig.conf.sample index 2f1554f63..542bedb52 100644 --- a/configs/extconfig.conf.sample +++ b/configs/extconfig.conf.sample @@ -7,7 +7,7 @@ ; [settings] ; -; Static configuration files: +; Static configuration files: ; ; file.conf => driver,database[,table] ; diff --git a/configs/extensions.ael.sample b/configs/extensions.ael.sample index c7720290a..69f441d1e 100644 --- a/configs/extensions.ael.sample +++ b/configs/extensions.ael.sample @@ -3,49 +3,49 @@ // // // 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 +// 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)} // ${${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=DAHDI/1 -//CONSOLE=Phone/phone0 -IAXINFO=guest; // IAXtel username/password -//IAXINFO="myuser:mypass"; -TRUNK="DAHDI/G2"; // Trunk interface -// -// Note the 'G2' in the TRUNK variable above. It specifies which group (defined -// in 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 DAHDI channel -// (aka. ascending sequential hunt group). -// G: select the highest-numbered non-busy DAHDI 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. + CONSOLE="Console/dsp"; // Console interface for demo + //CONSOLE=DAHDI/1 + //CONSOLE=Phone/phone0 + IAXINFO=guest; // IAXtel username/password + //IAXINFO="myuser:mypass"; + TRUNK="DAHDI/G2"; // Trunk interface + // + // Note the 'G2' in the TRUNK variable above. It specifies which group (defined + // in 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 DAHDI channel + // (aka. ascending sequential hunt group). + // G: select the highest-numbered non-busy DAHDI 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 '_' @@ -56,12 +56,12 @@ TRUNKMSD=1; // MSD digits to strip (usually 1 or 0) // 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 +// . - 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, +// 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. // @@ -72,8 +72,8 @@ TRUNKMSD=1; // MSD digits to strip (usually 1 or 0) // 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 +// 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 @@ -87,11 +87,11 @@ TRUNKMSD=1; // MSD digits to strip (usually 1 or 0) // exten-name => { // application(arg1,arg2,...); // -// Timing list for includes is +// Timing list for includes is // // <time range>|<days of week>|<days of month>|<months> // -// includes { +// includes { // daytime|9:00-17:00|mon-fri|*|*; // }; // @@ -110,73 +110,73 @@ TRUNKMSD=1; // MSD digits to strip (usually 1 or 0) // // context ael-dundi-e164-canonical { -// -// List canonical entries here -// -// 12564286000 => &ael-std-exten(6000,IAX2/foo); -// _125642860XX => Dial(IAX2/otherbox/${EXTEN:7}); + // + // List canonical entries here + // + // 12564286000 => &ael-std-exten(6000,IAX2/foo); + // _125642860XX => Dial(IAX2/otherbox/${EXTEN:7}); }; context ael-dundi-e164-customers { -// -// If you are an ITSP or Reseller, list your customers here. -// -//_12564286000 => Dial(SIP/customer1); -//_12564286001 => Dial(IAX2/customer2); + // + // If you are an ITSP or Reseller, list your customers here. + // + //_12564286000 => Dial(SIP/customer1); + //_12564286001 => Dial(IAX2/customer2); }; context ael-dundi-e164-via-pstn { -// -// If you are freely delivering calls to the PSTN, list them here -// -//_1256428XXXX => Dial(DAHDI/G2/${EXTEN:7}); // Expose all of 256-428 -//_1256325XXXX => Dial(DAHDI/G2/${EXTEN:7}); // Ditto for 256-325 + // + // If you are freely delivering calls to the PSTN, list them here + // + //_1256428XXXX => Dial(DAHDI/G2/${EXTEN:7}); // Expose all of 256-428 + //_1256325XXXX => Dial(DAHDI/G2/${EXTEN:7}); // Ditto for 256-325 }; context ael-dundi-e164-local { -// -// Context to put your dundi IAX2 or SIP user in for -// full access -// -includes { -ael-dundi-e164-canonical; -ael-dundi-e164-customers; -ael-dundi-e164-via-pstn; -}; + // + // Context to put your dundi IAX2 or SIP user in for + // full access + // + includes { + ael-dundi-e164-canonical; + ael-dundi-e164-customers; + ael-dundi-e164-via-pstn; + }; }; context ael-dundi-e164-switch { -// -// Just a wrapper for the switch -// + // + // Just a wrapper for the switch + // -switches { -DUNDi/e164; -}; + switches { + DUNDi/e164; + }; }; context ael-dundi-e164-lookup { -// -// Locally to lookup, try looking for a local E.164 solution -// then try DUNDi if we don't have one. -// -includes { -ael-dundi-e164-local; -ael-dundi-e164-switch; -}; -// + // + // Locally to lookup, try looking for a local E.164 solution + // then try DUNDi if we don't have one. + // + includes { + ael-dundi-e164-local; + ael-dundi-e164-switch; + }; + // }; // -// DUNDi can also be implemented as a Macro instead of using -// the Local channel driver. +// DUNDi can also be implemented as a Macro instead of using +// the Local channel driver. // macro ael-dundi-e164(exten) { // // ARG1 is the extension to Dial // -goto ${exten}|1; -return; + goto ${exten}|1; + return; }; // @@ -186,7 +186,7 @@ return; // up, please go to www.gnophone.com or www.iaxtel.com // context ael-iaxtel700 { -_91700XXXXXXX => Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel); + _91700XXXXXXX => Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel); }; // @@ -196,99 +196,99 @@ _91700XXXXXXX => Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel); // to be on-line or else dialing can be severly delayed. // context ael-iaxprovider { -switches { -// IAX2/user:[key]@myserver/mycontext; -}; + switches { + // IAX2/user:[key]@myserver/mycontext; + }; }; context ael-trunkint { -// -// International long distance through trunk -// -includes { -ael-dundi-e164-lookup; -}; -_9011. => { -&ael-dundi-e164(${EXTEN:4}); -Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}); -}; + // + // International long distance through trunk + // + includes { + ael-dundi-e164-lookup; + }; + _9011. => { + &ael-dundi-e164(${EXTEN:4}); + Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}); + }; }; context ael-trunkld { -// -// Long distance context accessed through trunk -// -includes { -ael-dundi-e164-lookup; -}; -_91NXXNXXXXXX => { -&ael-dundi-e164(${EXTEN:1}); -Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}); -}; + // + // Long distance context accessed through trunk + // + includes { + ael-dundi-e164-lookup; + }; + _91NXXNXXXXXX => { + &ael-dundi-e164(${EXTEN:1}); + Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}); + }; }; context ael-trunklocal { -// -// Local seven-digit dialing accessed through trunk interface -// -_9NXXXXXX => { -Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}); -}; + // + // Local seven-digit dialing accessed through trunk interface + // + _9NXXXXXX => { + Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}); + }; }; context ael-trunktollfree { -// -// Long distance context accessed through trunk interface -// + // + // Long distance context accessed through trunk interface + // -_91800NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}); -_91888NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}); -_91877NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}); -_91866NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}); + _91800NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}); + _91888NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}); + _91877NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}); + _91866NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}); }; context ael-international { -// -// Master context for international long distance -// -ignorepat => 9; -includes { -ael-longdistance; -ael-trunkint; -}; + // + // Master context for international long distance + // + ignorepat => 9; + includes { + ael-longdistance; + ael-trunkint; + }; }; context ael-longdistance { -// -// Master context for long distance -// -ignorepat => 9; -includes { -ael-local; -ael-trunkld; -}; + // + // Master context for long distance + // + ignorepat => 9; + includes { + ael-local; + ael-trunkld; + }; }; context ael-local { -// -// Master context for local, toll-free, and iaxtel calls only -// -ignorepat => 9; -includes { -ael-default; -ael-trunklocal; -ael-iaxtel700; -ael-trunktollfree; -ael-iaxprovider; -}; + // + // Master context for local, toll-free, and iaxtel calls only + // + ignorepat => 9; + includes { + ael-default; + ael-trunklocal; + ael-iaxtel700; + ael-trunktollfree; + ael-iaxprovider; + }; }; // // You can use an alternative switch type as well, to resolve -// extensions that are not known here, for example with remote +// extensions that are not known here, for example with remote // IAX switching you transparently get access to the remote // Asterisk PBX -// +// // switch => IAX2/user:password@bigserver/local // // An "lswitch" is like a switch but is literal, in that @@ -306,69 +306,69 @@ ael-iaxprovider; macro ael-std-exten-ael( ext , dev ) { -Dial(${dev}/${ext},20); -switch(${DIALSTATUS}) { -case BUSY: -Voicemail(${ext},b); -break; -default: -Voicemail(${ext},u); -}; -catch a { -VoiceMailMain(${ext}); -return; -}; -return; + Dial(${dev}/${ext},20); + switch(${DIALSTATUS}) { + case BUSY: + Voicemail(${ext},b); + break; + default: + Voicemail(${ext},u); + }; + catch a { + VoiceMailMain(${ext}); + return; + }; + return; }; context ael-demo { -s => { -Wait(1); -Answer(); -Set(TIMEOUT(digit)=5); -Set(TIMEOUT(response)=10); + s => { + Wait(1); + Answer(); + Set(TIMEOUT(digit)=5); + Set(TIMEOUT(response)=10); restart: -Background(demo-congrats); + Background(demo-congrats); instructions: -for (x=0; ${x} < 3; x=${x} + 1) { -Background(demo-instruct); -WaitExten(); -}; -}; -2 => { -Background(demo-moreinfo); -goto s|instructions; -}; -3 => { -Set(LANGUAGE()=fr); -goto s|restart; -}; -1000 => { -goto ael-default|s|1; -}; -500 => { -Playback(demo-abouttotry); -Dial(IAX2/guest@misery.digium.com/s@default); -Playback(demo-nogo); -goto s|instructions; -}; -600 => { -Playback(demo-echotest); -Echo(); -Playback(demo-echodone); -goto s|instructions; -}; -_1234 => &ael-std-exten-ael(${EXTEN}, "IAX2"); -8500 => { -VoicemailMain(); -goto s|instructions; -}; -# => { -Playback(demo-thanks); -Hangup(); -}; -t => goto #|1; -i => Playback(invalid); + for (x=0; ${x} < 3; x=${x} + 1) { + Background(demo-instruct); + WaitExten(); + }; + }; + 2 => { + Background(demo-moreinfo); + goto s|instructions; + }; + 3 => { + Set(LANGUAGE()=fr); + goto s|restart; + }; + 1000 => { + goto ael-default|s|1; + }; + 500 => { + Playback(demo-abouttotry); + Dial(IAX2/guest@misery.digium.com/s@default); + Playback(demo-nogo); + goto s|instructions; + }; + 600 => { + Playback(demo-echotest); + Echo(); + Playback(demo-echodone); + goto s|instructions; + }; + _1234 => &ael-std-exten-ael(${EXTEN}, "IAX2"); + 8500 => { + VoicemailMain(); + goto s|instructions; + }; + # => { + Playback(demo-thanks); + Hangup(); + }; + t => goto #|1; + i => Playback(invalid); }; @@ -380,12 +380,12 @@ i => Playback(invalid); context ael-default { -// By default we include the demo. In a production system, you +// By default we include the demo. In a production system, you // probably don't want to have the demo there. -includes { -ael-demo; -}; + includes { + ael-demo; + }; // // Extensions like the two below can be used for FWD, Nikotel, sipgate etc. // Note that you must have a [sipprovider] section in sip.conf whereas @@ -443,6 +443,6 @@ ael-demo; // friendly Asterisk CLI prompt. // // 'show application <command>' will show details of how you -// use that particular application in this file, the dial plan. +// use that particular application in this file, the dial plan. // } diff --git a/configs/extensions.conf.sample b/configs/extensions.conf.sample index 230576d45..db93f12e3 100644 --- a/configs/extensions.conf.sample +++ b/configs/extensions.conf.sample @@ -1,21 +1,21 @@ ; extensions.conf - the Asterisk dial plan ; ; Static extension configuration file, used by -; the pbx_config module. This is where you configure all your -; inbound and outbound calls in Asterisk. -; -; This configuration file is reloaded +; the pbx_config module. This is where you configure all your +; inbound and outbound calls in Asterisk. +; +; This configuration file is reloaded ; - With the "dialplan reload" command in the CLI ; - With the "reload" command (that reloads everything) in the CLI ; -; The "General" category is for certain variables. +; The "General" category is for certain variables. ; [general] ; ; If static is set to no, or omitted, then the pbx_config will rewrite ; this file when extensions are modified. Remember that all comments -; made in the file will be lost when that happens. +; made in the file will be lost when that happens. ; ; XXX Not yet implemented XXX ; @@ -30,8 +30,8 @@ writeprotect=no ; things to do, it will terminate the call with BUSY, CONGESTION ; or HANGUP depending on Asterisk's best guess. This is the default. ; -; If autofallthrough is not set, then if an extension runs out of -; things to do, Asterisk will wait for a new extension to be dialed +; If autofallthrough is not set, then if an extension runs out of +; things to do, Asterisk will wait for a new extension to be dialed ; (this is the original behavior of Asterisk 1.0 and earlier). ; ;autofallthrough=no @@ -41,7 +41,7 @@ writeprotect=no ; If extenpatternmatchnew is set (true, yes, etc), then a new algorithm that uses ; a Trie to find the best matching pattern is used. In dialplans ; with more than about 20-40 extensions in a single context, this -; new algorithm can provide a noticeable speedup. +; new algorithm can provide a noticeable speedup. ; With 50 extensions, the speedup is 1.32x ; with 88 extensions, the speedup is 2.23x ; with 138 extensions, the speedup is 3.44x @@ -49,15 +49,15 @@ writeprotect=no ; with 438 extensions, the speedup is 10.4x ; With 1000 extensions, the speedup is ~25x ; with 10,000 extensions, the speedup is 374x -; Basically, the new algorithm provides a flat response +; Basically, the new algorithm provides a flat response ; time, no matter the number of extensions. ; -; By default, the old pattern matcher is used. +; By default, the old pattern matcher is used. ; ; ****This is a new feature! ********************* -; The new pattern matcher is for the brave, the bold, and +; The new pattern matcher is for the brave, the bold, and ; the desperate. If you have large dialplans (more than about 50 extensions -; in a context), and/or high call volume, you might consider setting +; in a context), and/or high call volume, you might consider setting ; this value to "yes" !! ; Please, if you try this out, and are forced to return to the ; old pattern matcher, please report your reasons in a bug report @@ -69,7 +69,7 @@ writeprotect=no ; ;extenpatternmatchnew=no ; -; If clearglobalvars is set, global variables will be cleared +; If clearglobalvars is set, global variables will be cleared ; and reparsed on a dialplan reload, or Asterisk reload. ; ; If clearglobalvars is not set, then global variables will persist @@ -108,7 +108,7 @@ clearglobalvars=no ;#include "filename.conf" ; ; You can execute a program or script that produces config files, and they -; will be inserted where you insert the #exec command. The #exec command +; will be inserted where you insert the #exec command. The #exec command ; works on all asterisk configuration files. However, you will need to ; activate them within asterisk.conf with the "execincludes" option. They ; are otherwise considered a security risk. @@ -153,8 +153,8 @@ TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) ; yourself a ton of grief. ; WARNING WARNING WARNING WARNING ; -; Any category other than "General" and "Globals" represent -; extension contexts, which are collections of extensions. +; 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 '_' @@ -165,12 +165,12 @@ TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) ; 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 +; . - 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, +; 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. ; @@ -197,7 +197,7 @@ TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) ;[context] ;exten => someexten,{priority|label{+|-}offset}[(alias)],application(arg1,arg2,...) ; -; Timing list for includes is +; Timing list for includes is ; ; <time range>,<days of week>,<days of month>,<months>[,<timezone>] ; @@ -246,7 +246,7 @@ TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) ; ; If you are freely delivering calls to the PSTN, list them here ; -;exten => _1256428XXXX,1,Dial(DAHDI/G2/${EXTEN:7}) ; Expose all of 256-428 +;exten => _1256428XXXX,1,Dial(DAHDI/G2/${EXTEN:7}) ; Expose all of 256-428 ;exten => _1256325XXXX,1,Dial(DAHDI/G2/${EXTEN:7}) ; Ditto for 256-325 [dundi-e164-local] @@ -272,15 +272,15 @@ switch => DUNDi/e164 include => dundi-e164-local include => dundi-e164-switch ; -; DUNDi can also be implemented as a Macro instead of using -; the Local channel driver. +; DUNDi can also be implemented as a Macro instead of using +; the Local channel driver. ; [macro-dundi-e164] ; ; ARG1 is the extension to Dial ; ; Extension "s" is not a wildcard extension that matches "anything". -; In macros, it is the start extension. In most other cases, +; In macros, it is the start extension. In most other cases, ; you have to goto "s" to execute that extension. ; ; For wildcard matches, see above - all pattern matches start with @@ -367,10 +367,10 @@ include => iaxprovider include => parkedcalls ; ; You can use an alternative switch type as well, to resolve -; extensions that are not known here, for example with remote +; extensions that are not known here, for example with remote ; IAX switching you transparently get access to the remote ; Asterisk PBX -; +; ; switch => IAX2/user:password@bigserver/local ; ; An "lswitch" is like a switch but is literal, in that @@ -388,7 +388,7 @@ include => parkedcalls [macro-trunkdial] ; -; Standard trunk dial macro (hangs up on a dialstatus that should +; Standard trunk dial macro (hangs up on a dialstatus that should ; terminate call) ; ${ARG1} - What to dial ; @@ -430,7 +430,7 @@ exten => stdexten-NOANSWER,n,NoOp(Finish stdexten NOANSWER) exten => stdexten-NOANSWER,n,Return() ; If they press #, return to start exten => stdexten-BUSY,1,Voicemail(${mbx},b) -; If busy, send to voicemail w/ busy announce + ; If busy, send to voicemail w/ busy announce exten => stdexten-BUSY,n,NoOp(Finish stdexten BUSY) exten => stdexten-BUSY,n,Return() ; If they press #, return to start @@ -458,8 +458,8 @@ exten => _X.,n,Set(LOCAL(tortcntx)=${ARG4}) exten => _X.,n,Set(LOCAL(cntx)=${ARG5}) exten => _X.,n,Set(LOCAL(mbx)="${ext}"$["${cntx}" ? "@${cntx}" :: ""]) -exten => _X.,n,Dial(${dev},20,p) ; Ring the interface, 20 seconds maximum, call screening -; option (or use P for databased call _X.creening) +exten => _X.,n,Dial(${dev},20,p) ; Ring the interface, 20 seconds maximum, call screening + ; option (or use P for databased call _X.creening) exten => _X.,n,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER) exten => stdexten-NOANSWER,1,Voicemail(${mbx},u) ; If unavailable, send to voicemail w/ unavail announce @@ -520,8 +520,8 @@ exten => 1000,1,Goto(default,s,1) ; We also create an example user, 1234, who is on the console and has ; voicemail, etc. ; -exten => 1234,1,Playback(transfer,skip) ; "Please hold while..." -; (but skip if channel is not up) +exten => 1234,1,Playback(transfer,skip) ; "Please hold while..." + ; (but skip if channel is not up) exten => 1234,n,Gosub(stdexten(1234,${GLOBAL(CONSOLE)})) exten => 1234,n,Goto(default,s,1) ; exited Voicemail @@ -583,7 +583,7 @@ exten => 8500,n,Goto(s,6) ; ; The page context calls up the page macro that sets variables needed for auto-answer -; It is in is own context to make calling it from the Page() application as simple as +; It is in is own context to make calling it from the Page() application as simple as ; Local/{peername}@page ; [page] @@ -610,7 +610,7 @@ exten => _X.,1,Macro(page,SIP/${EXTEN}) [default] ; -; By default we include the demo. In a production system, you +; By default we include the demo. In a production system, you ; probably don't want to have the demo there. ; include => demo @@ -640,11 +640,11 @@ include => demo ;exten => 6394,1,Dial(Local/6275/n) ; this will dial ${MARK} ;exten => 6275,1,Gosub(stdexten(6275,${MARK})) -; assuming ${MARK} is something like DAHDI/2 + ; assuming ${MARK} is something like DAHDI/2 ;exten => 6275,n,Goto(default,s,1) ; exited Voicemail ;exten => mark,1,Goto(6275,1) ; alias mark to 6275 ;exten => 6536,1,Gosub(stdexten(6236,${WIL})) -; Ditto for wil + ; Ditto for wil ;exten => 6536,n,Goto(default,s,1) ; exited Voicemail ;exten => wil,1,Goto(6236,1) @@ -723,7 +723,7 @@ include => demo ; friendly Asterisk CLI prompt. ; ; "core show application <command>" will show details of how you -; use that particular application in this file, the dial plan. +; use that particular application in this file, the dial plan. ; "core show functions" will list all dialplan functions ; "core show function <COMMAND>" will show you more information about ; one function. Remember that function names are UPPER CASE. diff --git a/configs/extensions.lua.sample b/configs/extensions.lua.sample index 0bbb3aef1..df32ec705 100644 --- a/configs/extensions.lua.sample +++ b/configs/extensions.lua.sample @@ -22,20 +22,20 @@ TRUNKMSD = 1 -- 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 +-- . - 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. --- +-- -- If your extension has special characters in it such as '.' and '!' you must -- explicitly make it a string in the tabale definition: -- @@ -44,7 +44,7 @@ TRUNKMSD = 1 -- -- There are no priorities. All extensions to asterisk appear to have a single -- priority as if they consist of a single priority. --- +-- -- Each context is defined as a table in the extensions table. The -- context names should be strings. -- @@ -52,7 +52,7 @@ TRUNKMSD = 1 -- extension. This extension should be set to a table containing a list -- of context names. Do not put references to tables in the includes -- table. --- +-- -- include = {"a", "b", "c"}; -- -- Channel variables can be accessed thorugh the global 'channel' table. @@ -79,7 +79,7 @@ TRUNKMSD = 1 -- Also notice the absence of the following constructs from the examples above: -- channel.func_name(1,2,3) = "value" -- this will NOT work -- value = channel.func_name(1,2,3) -- this will NOT work as expected --- +-- -- -- Dialplan applications can be accessed through the global 'app' table. -- @@ -97,103 +97,103 @@ TRUNKMSD = 1 -- function outgoing_local(c, e) -app.dial("DAHDI/1/" .. e, "", "") + app.dial("DAHDI/1/" .. e, "", "") end function demo_instruct() -app.background("demo-instruct") -app.waitexten() + app.background("demo-instruct") + app.waitexten() end function demo_congrats() -app.background("demo-congrats") -demo_instruct() + app.background("demo-congrats") + demo_instruct() end -- Answer the chanel and play the demo sound files function demo_start(context, exten) -app.wait(1) -app.answer() + app.wait(1) + app.answer() -channel.TIMEOUT("digit"):set(5) -channel.TIMEOUT("response"):set(10) --- app.set("TIMEOUT(digit)=5") --- app.set("TIMEOUT(response)=10") + channel.TIMEOUT("digit"):set(5) + channel.TIMEOUT("response"):set(10) + -- app.set("TIMEOUT(digit)=5") + -- app.set("TIMEOUT(response)=10") -demo_congrats(context, exten) + demo_congrats(context, exten) end function demo_hangup() -app.playback("demo-thanks") -app.hangup() + app.playback("demo-thanks") + app.hangup() end extensions = { -demo = { -s = demo_start; - -["2"] = function() -app.background("demo-moreinfo") -demo_instruct() -end; -["3"] = function () -channel.LANGUAGE():set("fr") -- set the language to french -demo_congrats() -end; - -["1000"] = function() -app.goto("default", "s", 1) -end; - -["1234"] = function() -app.playback("transfer", "skip") --- do a dial here -end; - -["1235"] = function() -app.voicemail("1234", "u") -end; - -["1236"] = function() -app.dial("Console/dsp") -app.voicemail(1234, "b") -end; - -["#"] = demo_hangup; -t = demo_hangup; -i = function() -app.playback("invalid") -demo_instruct() -end; - -["500"] = function() -app.playback("demo-abouttotry") -app.dial("IAX2/guest@misery.digium.com/s@default") -app.playback("demo-nogo") -demo_instruct() -end; - -["600"] = function() -app.playback("demo-echotest") -app.echo() -app.playback("demo-echodone") -demo_instruct() -end; - -["8500"] = function() -app.voicemailmain() -demo_instruct() -end; - -}; - -default = { --- by default, do the demo -include = {"demo"}; -}; - -["local"] = { -["_NXXXXXX"] = outgoing_local; -}; + demo = { + s = demo_start; + + ["2"] = function() + app.background("demo-moreinfo") + demo_instruct() + end; + ["3"] = function () + channel.LANGUAGE():set("fr") -- set the language to french + demo_congrats() + end; + + ["1000"] = function() + app.goto("default", "s", 1) + end; + + ["1234"] = function() + app.playback("transfer", "skip") + -- do a dial here + end; + + ["1235"] = function() + app.voicemail("1234", "u") + end; + + ["1236"] = function() + app.dial("Console/dsp") + app.voicemail(1234, "b") + end; + + ["#"] = demo_hangup; + t = demo_hangup; + i = function() + app.playback("invalid") + demo_instruct() + end; + + ["500"] = function() + app.playback("demo-abouttotry") + app.dial("IAX2/guest@misery.digium.com/s@default") + app.playback("demo-nogo") + demo_instruct() + end; + + ["600"] = function() + app.playback("demo-echotest") + app.echo() + app.playback("demo-echodone") + demo_instruct() + end; + + ["8500"] = function() + app.voicemailmain() + demo_instruct() + end; + + }; + + default = { + -- by default, do the demo + include = {"demo"}; + }; + + ["local"] = { + ["_NXXXXXX"] = outgoing_local; + }; } diff --git a/configs/extensions_minivm.conf.sample b/configs/extensions_minivm.conf.sample index 75f87c165..2f9d24637 100644 --- a/configs/extensions_minivm.conf.sample +++ b/configs/extensions_minivm.conf.sample @@ -1,4 +1,4 @@ -; MINI-VOICEMAIL dialplan example +; MINI-VOICEMAIL dialplan example ; --------------------------------------------------------------------------------------- ; ASTERISK_FILE_VERSION(__FILE__, "$Revision$") ; @@ -10,7 +10,7 @@ ; A macro to test the MINIVMACCOUNT dialplan function ; Currently, accountcode and pincode is not used in the application ; They where added to be used in dialplan scripting -; +; ; [macro-minivmfunctest] exten => s,1,set(account=${ARGV1}) diff --git a/configs/features.conf.sample b/configs/features.conf.sample index 83aa69643..23732d824 100644 --- a/configs/features.conf.sample +++ b/configs/features.conf.sample @@ -5,52 +5,52 @@ [general] parkext => 700 ; What extension to dial to park (all parking lots) parkpos => 701-720 ; What extensions to park calls on. (defafult parking lot) -; These needs to be numeric, as Asterisk starts from the start position -; and increments with one for the next parked call. + ; These needs to be numeric, as Asterisk starts from the start position + ; and increments with one for the next parked call. context => parkedcalls ; Which context parked calls are in (default parking lot) ;parkinghints = no ; Add hints priorities automatically for parking slots (default is no). -;parkingtime => 45 ; Number of seconds a call can be parked for -; (default is 45 seconds) +;parkingtime => 45 ; Number of seconds a call can be parked for + ; (default is 45 seconds) ;comebacktoorigin = yes ; Whether to return to the original calling extension upon parking -; timeout or to send the call to context 'parkedcallstimeout' at -; extension 's', priority '1' (default is yes). -;courtesytone = beep ; Sound file to play to the parked caller -; when someone dials a parked call -; or the Touch Monitor is activated/deactivated. + ; timeout or to send the call to context 'parkedcallstimeout' at + ; extension 's', priority '1' (default is yes). +;courtesytone = beep ; Sound file to play to the parked caller + ; when someone dials a parked call + ; or the Touch Monitor is activated/deactivated. ;parkedplay = caller ; Who to play the courtesy tone to when picking up a parked call -; one of: parked, caller, both (default is caller) + ; one of: parked, caller, both (default is caller) ;parkedcalltransfers = caller ; Enables or disables DTMF based transfers when picking up a parked call. -; one of: callee, caller, both, no (default is no) + ; one of: callee, caller, both, no (default is no) ;parkedcallreparking = caller ; Enables or disables DTMF based parking when picking up a parked call. -; one of: callee, caller, both, no (default is no) + ; one of: callee, caller, both, no (default is no) ;parkedcallhangup = caller ; Enables or disables DTMF based hangups when picking up a parked call. -; one of: callee, caller, both, no (default is no) + ; one of: callee, caller, both, no (default is no) ;parkedcallrecording = caller ; Enables or disables DTMF based one-touch recording when picking up a parked call. -; one of: callee, caller, both, no (default is no) + ; one of: callee, caller, both, no (default is no) ;adsipark = yes ; if you want ADSI parking announcements -;findslot => next ; Continue to the 'next' free parking space. -; Defaults to 'first' available +;findslot => next ; Continue to the 'next' free parking space. + ; Defaults to 'first' available ;parkedmusicclass=default ; This is the MOH class to use for the parked channel -; as long as the class is not set on the channel directly -; using Set(CHANNEL(musicclass)=whatever) in the dialplan + ; as long as the class is not set on the channel directly + ; using Set(CHANNEL(musicclass)=whatever) in the dialplan ;transferdigittimeout => 3 ; Number of seconds to wait between digits when transferring a call -; (default is 3 seconds) + ; (default is 3 seconds) ;xfersound = beep ; to indicate an attended transfer is complete ;xferfailsound = beeperr ; to indicate a failed transfer ;pickupexten = *8 ; Configure the pickup extension. (default is *8) ;pickupsound = beep ; to indicate a successful pickup (default: no sound) ;pickupfailsound = beeperr ; to indicate that the pickup failed (default: no sound) -;featuredigittimeout = 1000 ; Max time (ms) between digits for -; feature activation (default is 1000 ms) +;featuredigittimeout = 1000 ; Max time (ms) between digits for + ; feature activation (default is 1000 ms) ;atxfernoanswertimeout = 15 ; Timeout for answer on attended transfer default is 15 seconds. ;atxferdropcall = no ; If someone does an attended transfer, then hangs up before the transferred -; caller is connected, then by default, the system will try to call back the -; person that did the transfer. If this is set to "yes", the callback will -; not be attempted and the transfer will just fail. + ; caller is connected, then by default, the system will try to call back the + ; person that did the transfer. If this is set to "yes", the callback will + ; not be attempted and the transfer will just fail. ;atxferloopdelay = 10 ; Number of seconds to sleep between retries (if atxferdropcall = no) ;atxfercallbackretries = 2 ; Number of times to attempt to send the call back to the transferer. -; By default, this is 2. + ; By default, this is 2. ; Note that the DTMF features listed below only work when two channels have answered and are bridged together. ; They can not be used while the remote party is ringing or in progress. If you require this feature you can use diff --git a/configs/festival.conf.sample b/configs/festival.conf.sample index 774f1a16c..e91821719 100644 --- a/configs/festival.conf.sample +++ b/configs/festival.conf.sample @@ -15,9 +15,9 @@ ; ;usecache=yes ; -; If usecache=yes, a directory to store waveform cache files. +; If usecache=yes, a directory to store waveform cache files. ; The cache is never cleared (yet), so you must take care of cleaning it -; yourself (just delete any or all files from the cache). +; yourself (just delete any or all files from the cache). ; THIS DIRECTORY *MUST* EXIST and must be writable from the asterisk process. ; Defaults to /tmp/ ; @@ -25,10 +25,10 @@ ; ; Festival command to send to the server. ; Defaults to: (tts_textasterisk "%s" 'file)(quit)\n -; %s is replaced by the desired text to say. The command MUST end with a -; (quit) directive, or the cache handling mechanism will hang. Do not -; forget the \n at the end. -; +; %s is replaced by the desired text to say. The command MUST end with a +; (quit) directive, or the cache handling mechanism will hang. Do not +; forget the \n at the end. +; ;festivalcommand=(tts_textasterisk "%s" 'file)(quit)\n ; ; diff --git a/configs/followme.conf.sample b/configs/followme.conf.sample index a8a9955bb..b11836a5c 100644 --- a/configs/followme.conf.sample +++ b/configs/followme.conf.sample @@ -29,7 +29,7 @@ pls_hold_prompt=>followme/pls-hold-while-try status_prompt=>followme/status ; The global default for 'The party you're calling isn't at their desk' message. ; -sorry_prompt=>followme/sorry +sorry_prompt=>followme/sorry ; The global default for 'I'm sorry, but we were unable to locate your party' message. ; ; @@ -41,9 +41,9 @@ context=>default number=>01233456,25 ; The a follow-me number to call. The format is: ; number=> <number to call[&2nd #[&3rd #]]> [, <timeout value in seconds> [, <order in follow-me>] ] -; You can specify as many of these numbers as you like. They will be dialed in the +; You can specify as many of these numbers as you like. They will be dialed in the ; order that you specify them in the config file OR as specified with the order field -; on the number prompt. As you can see from the example, forked dialing of multiple +; on the number prompt. As you can see from the example, forked dialing of multiple ; numbers in the same step is supported with this application if you'd like to dial ; multiple numbers in the same followme step. ; It's also important to note that the timeout value is not the same @@ -79,7 +79,7 @@ status_prompt=>followme/status ; The 'The party you're calling isn't at their desk' message prompt. ; Default is the global default. ; -sorry_prompt=>followme/sorry +sorry_prompt=>followme/sorry ; The 'I'm sorry, but we were unable to locate your party' message prompt. Default ; is the global default. diff --git a/configs/func_odbc.conf.sample b/configs/func_odbc.conf.sample index 2b67e5396..1bc11be2e 100644 --- a/configs/func_odbc.conf.sample +++ b/configs/func_odbc.conf.sample @@ -76,10 +76,10 @@ readsql=${ARG1} ; ODBC_ANTIGF - A blacklist. [ANTIGF] dsn=mysql1,mysql2 ; Use mysql1 as the primary handle, but fall back to mysql2 -; if mysql1 is down. Supports up to 5 comma-separated -; DSNs. "dsn" may also be specified as "readhandle" and -; "writehandle", if it is important to separate reads and -; writes to different databases. + ; if mysql1 is down. Supports up to 5 comma-separated + ; DSNs. "dsn" may also be specified as "readhandle" and + ; "writehandle", if it is important to separate reads and + ; writes to different databases. readsql=SELECT COUNT(*) FROM exgirlfriends WHERE callerid='${SQL_ESC(${ARG1})}' syntax=<callerid> synopsis=Check if a specified callerid is contained in the ex-gf database diff --git a/configs/gtalk.conf.sample b/configs/gtalk.conf.sample index f3dd3f830..8873d0678 100644 --- a/configs/gtalk.conf.sample +++ b/configs/gtalk.conf.sample @@ -2,19 +2,19 @@ ;context=default ;;Context to dump call into ;bindaddr=0.0.0.0 ;;Address to bind to ;allowguest=yes ;;Allow calls from people not in -;;list of peers + ;;list of peers ; ;[guest] ;;special account for options on guest account -;disallow=all +;disallow=all ;allow=ulaw ;context=guest ; ;[ogorman] -;username=ogorman@gmail.com ;;username of the peer your -;;calling or accepting calls from +;username=ogorman@gmail.com ;;username of the peer your + ;;calling or accepting calls from ;disallow=all ;allow=ulaw -;context=default +;context=default ;connection=asterisk ;;client or component in jabber.conf -;;for the call to leave on. + ;;for the call to leave on. ; diff --git a/configs/h323.conf.sample b/configs/h323.conf.sample index c2e5db328..bdeb6b320 100644 --- a/configs/h323.conf.sample +++ b/configs/h323.conf.sample @@ -44,7 +44,7 @@ port = 1720 ; or ;dtmfmode=cisco:121 ; -; Set the gatekeeper +; Set the gatekeeper ; DISCOVER - Find the Gk address using multicast ; DISABLE - Disable the use of a GK ; <IP address> or <Host name> - The acutal IP address or hostname of your GK @@ -70,9 +70,9 @@ port = 1720 ; ;UserByAlias=no ; -; Default context gets used in siutations where you are using -; the GK routed model or no type=user was found. This gives you -; the ability to either play an invalid message or to simply not +; Default context gets used in siutations where you are using +; the GK routed model or no type=user was found. This gives you +; the ability to either play an invalid message or to simply not ; use user authentication at all. ; ;context=default @@ -122,27 +122,27 @@ port = 1720 ; ;------------------------------ JITTER BUFFER CONFIGURATION -------------------------- ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a -; H323 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 H323 channel can accept jitter, -; thus an enabled jitterbuffer on the receive H323 side will only -; be used if the sending side can create jitter and jbforce is -; also set to yes. + ; H323 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 H323 channel can accept jitter, + ; thus an enabled jitterbuffer on the receive H323 side will only + ; be used if the sending side can create jitter and jbforce is + ; also set to yes. ; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a H323 -; channel. Defaults to "no". + ; channel. Defaults to "no". ; 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, usualy sent from exotic devices -; and programs. Defaults to 1000. + ; resynchronized. Useful to improve the quality of the voice, with + ; big jumps in/broken timestamps, usualy sent from exotic devices + ; and programs. Defaults to 1000. ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a H323 -; channel. Two implementations are currenlty available - "fixed" -; (with size always equals to jbmax-size) and "adaptive" (with -; variable size, actually the new jb of IAX2). Defaults to fixed. + ; channel. Two implementations are currenlty 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". ;----------------------------------------------------------------------------------- @@ -153,7 +153,7 @@ port = 1720 ; and Gatekeeper, if there is one. ; ; Example: if someone calls time@your.asterisk.box.com -; Asterisk will send the call to the extension 'time' +; Asterisk will send the call to the extension 'time' ; in the context default ; ; [default] @@ -161,13 +161,13 @@ port = 1720 ; exten => time,2,Playback,current-time ; ; Keyword's 'prefix' and 'e164' are only make sense when -; used with a gatekeeper. You can specify either a prefix +; used with a gatekeeper. You can specify either a prefix ; or E.164 this endpoint is responsible for terminating. -; +; ; Example: The H.323 alias 'det-gw' will tell the gatekeeper ; to route any call with the prefix 1248 to this alias. Keyword ; e164 is used when you want to specifiy a full telephone -; number. So a call to the number 18102341212 would be +; number. So a call to the number 18102341212 would be ; routed to the H.323 alias 'time'. ; ;[time] @@ -182,10 +182,10 @@ port = 1720 ; ; ; Inbound H.323 calls from BillyBob would land in the incoming -; context with a maximum of 4 concurrent incoming calls -; +; context with a maximum of 4 concurrent incoming calls +; ; -; Note: If keyword 'incominglimit' are omitted Asterisk will not +; Note: If keyword 'incominglimit' are omitted Asterisk will not ; enforce any maximum number of concurrent calls. ; ;[BillyBob] diff --git a/configs/http.conf.sample b/configs/http.conf.sample index a47a2d653..017263b38 100644 --- a/configs/http.conf.sample +++ b/configs/http.conf.sample @@ -38,7 +38,7 @@ bindaddr=127.0.0.1 ;enablestatic=yes ; ; Redirect one URI to another. This is how you would set a -; default page. +; default page. ; Syntax: redirect=<from here> <to there> ; For example, if you are using the Asterisk-gui, ; it is convenient to enable the following redirect: diff --git a/configs/iax.conf.sample b/configs/iax.conf.sample index 259fe626b..562a4e6c8 100644 --- a/configs/iax.conf.sample +++ b/configs/iax.conf.sample @@ -12,11 +12,11 @@ [general] ;bindport=4569 ; bindport and bindaddr may be specified ; ; NOTE: bindport must be specified BEFORE -; bindaddr or may be specified on a specific -; bindaddr if followed by colon and port -; (e.g. bindaddr=192.168.0.1:4569) + ; bindaddr or may be specified on a specific + ; bindaddr if followed by colon and port + ; (e.g. bindaddr=192.168.0.1:4569) ;bindaddr=192.168.0.1 ; more than once to bind to multiple -; ; addresses, but the first will be the +; ; addresses, but the first will be the ; ; default ; ; Set iaxcompat to yes if you plan to use layered switches or @@ -36,7 +36,7 @@ ; ; For increased security against brute force password attacks ; enable "delayreject" which will delay the sending of authentication -; reject for REGREQ or AUTHREP if there is a password. +; reject for REGREQ or AUTHREP if there is a password. ; ;delayreject=yes ; @@ -60,7 +60,7 @@ ; ;accountcode=lss0101 ; -; You may specify a global default language for users. +; You may specify a global default language for users. ; Can be specified also on a per-user basis ; If omitted, will fallback to english ; @@ -111,7 +111,7 @@ disallow=lpc10 ; Icky sound quality... Mr. Roboto. ; ; forcejitterbuffer=yes|no: in the ideal world, when we bridge VoIP channels ; we don't want to do jitterbuffering on the switch, since the endpoints -; can each handle this. However, some endpoints may have poor jitterbuffers +; can each handle this. However, some endpoints may have poor jitterbuffers ; themselves, so this option will force * to always jitterbuffer, even in this ; case. ; @@ -166,7 +166,7 @@ forcejitterbuffer=no ; ; With a large amount of traffic on IAX2 trunks, there is a risk of bad voice quality due to ; the fact that the IAX2 trunking scheme depends on the Linux system to handle fragmentation of -; UDP packets. This may not be very efficient. +; UDP packets. This may not be very efficient. ; This setting sets the maximum transmission unit for IAX2 UDP trunking. ; default is 1240 bytes. Zero disables this functionality and let's the O/S handle fragmentation. ; @@ -177,7 +177,7 @@ forcejitterbuffer=no ; encryption = yes ; ; Force encryption insures no connection is established unless both sides support -; encryption. By turning this option on, encryption is automatically turned on as well. +; encryption. By turning this option on, encryption is automatically turned on as well. ; ; forceencryption = yes @@ -211,7 +211,7 @@ forcejitterbuffer=no ; Sample Registration for iaxtel ; ; Visit http://www.iaxtel.com to register with iaxtel. Replace "user" -; and "pass" with your username and password for iaxtel. Incoming +; and "pass" with your username and password for iaxtel. Incoming ; calls arrive at the "s" extension of "default" context. ; ;register => user:pass@iaxtel.com @@ -228,7 +228,7 @@ forcejitterbuffer=no ;register => FWDNumber:passwd@iax.fwdnet.net ; ; -; You can disable authentication debugging to reduce the amount of +; You can disable authentication debugging to reduce the amount of ; debugging traffic. ; ;authdebug=no @@ -256,7 +256,7 @@ forcejitterbuffer=no autokill=yes ; ; codecpriority controls the codec negotiation of an inbound IAX call. -; This option is inherited to all user entities. It can also be defined +; This option is inherited to all user entities. It can also be defined ; in each user entity separately which will override the setting in general. ; ; The valid values are: @@ -284,29 +284,29 @@ autokill=yes ;allowfwdownload=yes ;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list -; just like friends added from the config file only on a -; as-needed basis? (yes|no) + ; just like friends added from the config file only on a + ; as-needed basis? (yes|no) ;rtupdate=yes ; Send registry updates to database using realtime? (yes|no) -; If set to yes, when a IAX2 peer registers successfully, -; the ip address, the origination port, the registration period, -; and the username of the peer will be set to database via realtime. -; If not present, defaults to 'yes'. + ; If set to yes, when a IAX2 peer registers successfully, + ; the ip address, the origination port, the registration period, + ; and the username of the peer will be set to database via realtime. + ; If not present, defaults to 'yes'. ;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule -; as if it had just registered? (yes|no|<seconds>) -; If set to yes, when the registration expires, the friend will -; vanish from the configuration until requested again. -; If set to an integer, friends expire within this number of -; seconds instead of the registration interval. + ; as if it had just registered? (yes|no|<seconds>) + ; If set to yes, when the registration expires, the friend will + ; vanish from the configuration until requested again. + ; If set to an integer, friends expire within this number of + ; seconds instead of the registration interval. ;rtignoreregexpire=yes ; When reading a peer from Realtime, if the peer's registration -; has expired based on its registration interval, used the stored -; address information regardless. (yes|no) + ; has expired based on its registration interval, used the stored + ; address information regardless. (yes|no) ;parkinglot=edvina ; Default parkinglot for IAX peers and users -; This can also be configured per device -; Parkinglots are defined in features.conf + ; This can also be configured per device + ; Parkinglots are defined in features.conf ; Guest sections for unauthenticated connection attempts. Just specify an ; empty secret, or provide no secret section. @@ -357,7 +357,7 @@ inkeys=freeworlddialup ; across the net. "md5" uses a challenge/response md5 sum arrangement, but ; still requires both ends have plain text access to the secret. "rsa" allows ; unidirectional secret knowledge through public/private keys. If "rsa" -; authentication is used, "inkeys" is a list of acceptable public keys on the +; authentication is used, "inkeys" is a list of acceptable public keys on the ; local system that can be used to authenticate the remote peer, separated by ; the ":" character. "outkey" is a single, private key to use to authenticate ; to the other side. Public keys are named /var/lib/asterisk/keys/<name>.pub @@ -377,13 +377,13 @@ inkeys=freeworlddialup ;auth=md5,plaintext,rsa ;secret=markpasswd ;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will -; cause the given audio file to -; be played upon completion of -; an attended transfer. + ; cause the given audio file to + ; be played upon completion of + ; an attended transfer. ;dbsecret=mysecrets/place ; Secrets can be stored in astdb, too ;transfer=no ; Disable IAX native transfer -;transfer=mediaonly ; When doing IAX native transfers, transfer -; only media stream +;transfer=mediaonly ; When doing IAX native transfers, transfer + ; only media stream ;jitterbuffer=yes ; Override global setting an enable jitter buffer ; ; for this user ;maxauthreq=10 ; Set maximum number of outstanding AUTHREQs waiting for replies. Any further authentication attempts will be blocked @@ -395,10 +395,10 @@ inkeys=freeworlddialup ;language=en ; Use english as default language ;encryption=yes ; Enable IAX2 encryption. The default is no. ;keyrotate=off ; This is a compatibility option for older versions of -; ; IAX2 that do not support key rotation with encryption. -; ; This option will disable the IAX_COMMAND_RTENC message. +; ; IAX2 that do not support key rotation with encryption. +; ; This option will disable the IAX_COMMAND_RTENC message. ; ; default is on. -; ; +; ; ; ; Peers may also be specified, with a secret and ; a remote hostname. @@ -414,20 +414,20 @@ host=216.207.245.47 ;mask=255.255.255.255 ;qualify=yes ; Make sure this peer is alive ;qualifysmoothing = yes ; use an average of the last two PONG -; results to reduce falsely detected LAGGED hosts -; Default: Off + ; results to reduce falsely detected LAGGED hosts + ; Default: Off ;qualifyfreqok = 60000 ; how frequently to ping the peer when -; everything seems to be ok, in milliseconds + ; everything seems to be ok, in milliseconds ;qualifyfreqnotok = 10000 ; how frequently to ping the peer when it's -; either LAGGED or UNAVAILABLE, in milliseconds + ; either LAGGED or UNAVAILABLE, in milliseconds ;jitterbuffer=no ; Turn off jitter buffer for this peer ; ;encryption=yes ; Enable IAX2 encryption. The default is no. ;keyrotate=off ; This is a compatibility option for older versions of -; ; IAX2 that do not support key rotation with encryption. -; ; This option will disable the IAX_COMMAND_RTENC message. +; ; IAX2 that do not support key rotation with encryption. +; ; This option will disable the IAX_COMMAND_RTENC message. ; ; default is on. -; ; +; ; ; Peers can remotely register as well, so that they can be mobile. Default ; IP's can also optionally be given but are not required. Caller*ID can be ; suggested to the other side as well if it is for example a phone instead of diff --git a/configs/iaxprov.conf.sample b/configs/iaxprov.conf.sample index 06891d785..d3789dcde 100644 --- a/configs/iaxprov.conf.sample +++ b/configs/iaxprov.conf.sample @@ -7,7 +7,7 @@ ; Templates provide a group of settings from which provisioning takes place. ; A template may be based upon any template that has been specified before ; it. If the template that an entry is based on is not specified then it is -; presumed to be 'default' (unless it is the first of course). +; presumed to be 'default' (unless it is the first of course). ; ; Templates which begin with 'si-' are used for provisioning units with ; specific service identifiers. For example the entry "si-000364000126" diff --git a/configs/indications.conf.sample b/configs/indications.conf.sample index 239dcd11c..c7ab52ea0 100644 --- a/configs/indications.conf.sample +++ b/configs/indications.conf.sample @@ -516,7 +516,7 @@ callwaiting = 425/150,0/150,425/150,0/4000 dialrecall = 425/500,0/50 ; RECORDTONE - not specified record = 1400/500,0/15000 -; 950/1400/1800 3x0.33 on 1.0 off repeated 3 times +; 950/1400/1800 3x0.33 on 1.0 off repeated 3 times info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000 ; STUTTER - not specified stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425 @@ -567,7 +567,7 @@ stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/1 [sg] description = Singapore ; Singapore -; Reference: http://www.ida.gov.sg/idaweb/doc/download/I397/ida_ts_pstn1_i4r2.pdf +; Reference: http://www.ida.gov.sg/idaweb/doc/download/I397/ida_ts_pstn1_i4r2.pdf ; Frequency specs are: 425 Hz +/- 20Hz; 24 Hz +/- 2Hz; modulation depth 100%; SIT +/- 50Hz ringcadence = 400,200,400,2000 dial = 425 @@ -691,7 +691,7 @@ info = !950/330,!1400/330,!1800/330,0 stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440 [ve] -; Tone definition source for ve found on +; Tone definition source for ve found on ; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf description = Venezuela / South America ringcadence = 1000,4000 diff --git a/configs/jabber.conf.sample b/configs/jabber.conf.sample index 2990d8e91..6cfb755bd 100644 --- a/configs/jabber.conf.sample +++ b/configs/jabber.conf.sample @@ -1,14 +1,14 @@ [general] ;debug=yes ;;Turn on debugging by default. ;autoprune=yes ;;Auto remove users from buddy list. Depending on your -;;setup (ie, using your personal Gtalk account for a test) -;;you might lose your contacts list. Default is 'no'. + ;;setup (ie, using your personal Gtalk account for a test) + ;;you might lose your contacts list. Default is 'no'. ;autoregister=yes ;;Auto register users from buddy list. ;[asterisk] ;;label ;type=client ;;Client or Component connection ;serverhost=astjab.org ;;Route to server for example, -;; talk.google.com + ;; talk.google.com ;username=asterisk@astjab.org/asterisk ;;Username with optional resource. ;secret=blah ;;Password ;priority=1 ;;Resource priority @@ -17,7 +17,7 @@ ;usesasl=yes ;;Use sasl or not ;buddy=mogorman@astjab.org ;;Manual addition of buddy to list. ;status=available ;;One of: chat, available, away, -;; xaway, or dnd + ;; xaway, or dnd ;statusmessage="I am available" ;;Have custom status message for -;;Asterisk. + ;;Asterisk. ;timeout=100 ;;Timeout on the message stack. diff --git a/configs/jingle.conf.sample b/configs/jingle.conf.sample index f3dd3f830..8873d0678 100644 --- a/configs/jingle.conf.sample +++ b/configs/jingle.conf.sample @@ -2,19 +2,19 @@ ;context=default ;;Context to dump call into ;bindaddr=0.0.0.0 ;;Address to bind to ;allowguest=yes ;;Allow calls from people not in -;;list of peers + ;;list of peers ; ;[guest] ;;special account for options on guest account -;disallow=all +;disallow=all ;allow=ulaw ;context=guest ; ;[ogorman] -;username=ogorman@gmail.com ;;username of the peer your -;;calling or accepting calls from +;username=ogorman@gmail.com ;;username of the peer your + ;;calling or accepting calls from ;disallow=all ;allow=ulaw -;context=default +;context=default ;connection=asterisk ;;client or component in jabber.conf -;;for the call to leave on. + ;;for the call to leave on. ; diff --git a/configs/logger.conf.sample b/configs/logger.conf.sample index fc64078da..c9e9890a7 100644 --- a/configs/logger.conf.sample +++ b/configs/logger.conf.sample @@ -15,7 +15,7 @@ ; see strftime(3) Linux manual for format specifiers. Note that there is also ; a fractional second parameter which may be used in this field. Use %1q ; for tenths, %2q for hundredths, etc. -; +; ;dateformat=%F %T ; ISO 8601 date format ;dateformat=%F %T.%3q ; with milliseconds ; @@ -90,7 +90,7 @@ console => notice,warning,error messages => notice,warning,error ;full => notice,warning,error,debug,verbose -;syslog keyword : This special keyword logs to syslog facility +;syslog keyword : This special keyword logs to syslog facility ; ;syslog.local0 => notice,warning,error ; diff --git a/configs/manager.conf.sample b/configs/manager.conf.sample index 855b9e6bc..28a815401 100644 --- a/configs/manager.conf.sample +++ b/configs/manager.conf.sample @@ -1,6 +1,6 @@ ; ; AMI - The Asterisk Manager Interface -; +; ; Third party application call management support and PBX event supervision ; ; This configuration file is read every time someone logs in @@ -13,11 +13,11 @@ ; ---------------------------- SECURITY NOTE ------------------------------- ; Note that you should not enable the AMI on a public IP address. If needed, ; block this TCP port with iptables (or another FW software) and reach it -; with IPsec, SSH, or SSL vpn tunnel. You can also make the manager +; with IPsec, SSH, or SSL vpn tunnel. You can also make the manager ; interface available over http/https if Asterisk's http server is enabled in ; http.conf and if both "enabled" and "webenabled" are set to yes in -; this file. Both default to no. httptimeout provides the maximum -; timeout in seconds before a web based session is discarded. The +; this file. Both default to no. httptimeout provides the maximum +; timeout in seconds before a web based session is discarded. The ; default is 60 seconds. ; [general] @@ -27,9 +27,9 @@ port = 5038 ;httptimeout = 60 ; a) httptimeout sets the Max-Age of the http cookie -; b) httptimeout is the amount of time the webserver waits +; b) httptimeout is the amount of time the webserver waits ; on a action=waitevent request (actually its httptimeout-10) -; c) httptimeout is also the amount of time the webserver keeps +; c) httptimeout is also the amount of time the webserver keeps ; a http session alive after completing a successful action bindaddr = 0.0.0.0 @@ -44,8 +44,8 @@ bindaddr = 0.0.0.0 ;tlsbindaddr=0.0.0.0 ; address to bind to, default to bindaddr ;tlscertfile=/tmp/asterisk.pem ; path to the certificate. ;tlsprivatekey=/tmp/private.pem ; path to the private key, if no private given, -; if no tlsprivatekey is given, default is to search -; tlscertfile for private key. + ; if no tlsprivatekey is given, default is to search + ; tlscertfile for private key. ;tlscipher=<cipher string> ; string specifying which SSL ciphers to use or not use ; ;allowmultiplelogin = yes ; IF set to no, rejects manager logins that are already in use. @@ -58,7 +58,7 @@ bindaddr = 0.0.0.0 ;timestampevents = yes ; debug = on ; enable some debugging info in AMI messages (default off). -; Also accessible through the "manager debug" CLI command. + ; Also accessible through the "manager debug" CLI command. ;[mark] ;secret = mysecret ;deny=0.0.0.0/0.0.0.0 @@ -72,7 +72,7 @@ bindaddr = 0.0.0.0 ; ;displayconnects = yes ; Display on CLI user login/logoff ; -; Authorization for various classes +; Authorization for various classes ; ; Read authorization permits you to receive asynchronous events, in general. ; Write authorization permits you to send commands and get back responses. The diff --git a/configs/meetme.conf.sample b/configs/meetme.conf.sample index 05bcb893f..c40c4606e 100644 --- a/configs/meetme.conf.sample +++ b/configs/meetme.conf.sample @@ -5,13 +5,13 @@ [general] ;audiobuffers=32 ; The number of 20ms audio buffers to be used -; when feeding audio frames from non-DAHDI channels -; into the conference; larger numbers will allow -; for the conference to 'de-jitter' audio that arrives -; at different timing than the conference's timing -; source, but can also allow for latency in hearing -; the audio from the speaker. Minimum value is 2, -; maximum value is 32. + ; when feeding audio frames from non-DAHDI channels + ; into the conference; larger numbers will allow + ; for the conference to 'de-jitter' audio that arrives + ; at different timing than the conference's timing + ; source, but can also allow for latency in hearing + ; the audio from the speaker. Minimum value is 2, + ; maximum value is 32. ; ; Conferences may be scheduled from realtime? ;schedule=yes @@ -34,12 +34,12 @@ ; [rooms] ; -; Usage is conf => confno[,pin][,adminpin] +; Usage is conf => confno[,pin][,adminpin] ; ; Note that once a participant has called the conference, a change to the pin ; number done in this file will not take effect until there are no more users ; in the conference and it goes away. When it is created again, it will have ; the new pin number. ; -;conf => 1234 +;conf => 1234 ;conf => 2345,9938 diff --git a/configs/mgcp.conf.sample b/configs/mgcp.conf.sample index 01c8fe77c..116b66cd0 100644 --- a/configs/mgcp.conf.sample +++ b/configs/mgcp.conf.sample @@ -13,27 +13,27 @@ ;------------------------------ JITTER BUFFER CONFIGURATION -------------------------- ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a -; MGCP 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 MGCP channel can accept jitter, -; thus an enabled jitterbuffer on the receive MGCP side will only -; be used if the sending side can create jitter and jbforce is -; also set to yes. + ; MGCP 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 MGCP channel can accept jitter, + ; thus an enabled jitterbuffer on the receive MGCP side will only + ; be used if the sending side can create jitter and jbforce is + ; also set to yes. ; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a MGCP -; channel. Defaults to "no". + ; channel. Defaults to "no". ; 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. + ; 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 MGCP -; 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. + ; 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". ;----------------------------------------------------------------------------------- @@ -47,27 +47,27 @@ ;; The MGCP channel supports the following service codes: ;; # - Transfer -;; *67 - Calling Number Delivery Blocking -;; *70 - Cancel Call Waiting -;; *72 - Call Forwarding Activation -;; *73 - Call Forwarding Deactivation -;; *78 - Do Not Disturb Activation -;; *79 - Do Not Disturb Deactivation +;; *67 - Calling Number Delivery Blocking +;; *70 - Cancel Call Waiting +;; *72 - Call Forwarding Activation +;; *73 - Call Forwarding Deactivation +;; *78 - Do Not Disturb Activation +;; *79 - Do Not Disturb Deactivation ;; *8 - Call pick-up ; -; known to work with Swissvoice IP10s -;[192.168.1.20] -;context=local -;host=192.168.1.20 -;callerid = "John Doe" <123> +; known to work with Swissvoice IP10s +;[192.168.1.20] +;context=local +;host=192.168.1.20 +;callerid = "John Doe" <123> ;callgroup=0 ; in the range from 0 to 63 ;pickupgroup=0 ; in the range from 0 to 63 -;nat=no -;threewaycalling=yes +;nat=no +;threewaycalling=yes ;transfer=yes ; transfer requires threewaycalling=yes. Use FLASH to transfer ;callwaiting=yes ; this might be a cause of trouble for ip10s -;cancallforward=yes -;line => aaln/1 +;cancallforward=yes +;line => aaln/1 ; ;[dph100] @@ -79,7 +79,7 @@ ;context=local ;host=dynamic ;dtmfmode=none ; DTMF Mode can be 'none', 'rfc2833', or 'inband' or -; 'hybrid' which starts in none and moves to inband. Default is none. + ; 'hybrid' which starts in none and moves to inband. Default is none. ;slowsequence=yes ; The DPH100M does not follow MGCP standards for sequencing ;line => aaln/1 @@ -87,11 +87,11 @@ ;[192.168.1.20] ;accountcode = 1000 ; record this in cdr as account identification for billing ;amaflags = billing ; record this in cdr as flagged for 'billing', -; 'documentation', or 'omit' + ; 'documentation', or 'omit' ;context = local ;host = 192.168.1.20 -;wcardep = aaln/* ; enables wildcard endpoint and sets it to 'aaln/*' -; another common format is '*' +;wcardep = aaln/* ; enables wildcard endpoint and sets it to 'aaln/*' + ; another common format is '*' ;callerid = "Duane Cox" <123> ; now lets setup line 1 using per endpoint configuration... ;callwaiting = no ;callreturn = yes diff --git a/configs/minivm.conf.sample b/configs/minivm.conf.sample index 0e29dd96d..55a39c869 100644 --- a/configs/minivm.conf.sample +++ b/configs/minivm.conf.sample @@ -16,7 +16,7 @@ ; Change the from, body and/or subject, variables: ; MVM_NAME, MVM_DUR, MVM_MSGNUM, VM_MAILBOX, MVM_CALLERID, MVM_CIDNUM, ; MVM_CIDNAME, MVM_DATE -; +; ; In addition to these, you can set the MVM_COUNTER channel variable in the ; dial plan and use that as a counter. It will also be used in the file name ; of the media file attached to the message @@ -89,43 +89,43 @@ emaildateformat=%A, %B %d, %Y at %r ;pagersubject=New VM ${MVM_COUNTER} ;pagerbody=New ${MVM_DUR} long msg in box ${MVM_MAILBOX}\nfrom ${MVM_CALLERID}, on ${MVM_DATE} ; -; +; ;--------------Timezone definitions (used in voicemail accounts) ------------------- ; -; Users may be located in different timezones, or may have different -; message announcements for their introductory message when they enter -; the voicemail system. Set the message and the timezone each user -; hears here. Set the user into one of these zones with the tz= attribute -; in the options field of the mailbox. Of course, language substitution -; still applies here so you may have several directory trees that have -; alternate language choices. -; -; Look in /usr/share/zoneinfo/ for names of timezones. -; Look at the manual page for strftime for a quick tutorial on how the -; variable substitution is done on the values below. -; -; Supported values: +; Users may be located in different timezones, or may have different +; message announcements for their introductory message when they enter +; the voicemail system. Set the message and the timezone each user +; hears here. Set the user into one of these zones with the tz= attribute +; in the options field of the mailbox. Of course, language substitution +; still applies here so you may have several directory trees that have +; alternate language choices. +; +; Look in /usr/share/zoneinfo/ for names of timezones. +; Look at the manual page for strftime for a quick tutorial on how the +; variable substitution is done on the values below. +; +; Supported values: ; 'filename' filename of a soundfile (single ticks around the filename ; required) -; ${VAR} variable substitution -; A or a Day of week (Saturday, Sunday, ...) -; B or b or h Month name (January, February, ...) -; d or e numeric day of month (first, second, ..., thirty-first) -; Y Year -; I or l Hour, 12 hour clock -; H Hour, 24 hour clock (single digit hours preceded by "oh") -; k Hour, 24 hour clock (single digit hours NOT preceded by "oh") -; M Minute, with 00 pronounced as "o'clock" +; ${VAR} variable substitution +; A or a Day of week (Saturday, Sunday, ...) +; B or b or h Month name (January, February, ...) +; d or e numeric day of month (first, second, ..., thirty-first) +; Y Year +; I or l Hour, 12 hour clock +; H Hour, 24 hour clock (single digit hours preceded by "oh") +; k Hour, 24 hour clock (single digit hours NOT preceded by "oh") +; M Minute, with 00 pronounced as "o'clock" ; N Minute, with 00 pronounced as "hundred" (US military time) -; P or p AM or PM +; P or p AM or PM ; Q "today", "yesterday" or ABdY -; (*note: not standard strftime value) +; (*note: not standard strftime value) ; q "" (for today), "yesterday", weekday, or ABdY -; (*note: not standard strftime value) -; R 24 hour time, including minute -; +; (*note: not standard strftime value) +; R 24 hour time, including minute +; ; The message here is not used in mini-voicemail, but stays for -; backwards compatibility +; backwards compatibility [zonemessages] eastern=America/New_York|'vm-received' Q 'digits/at' IMp @@ -141,27 +141,27 @@ military=Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p' ; attachmedia = yes | no ; Add media file as attachment? ; dateformat = <formatstring> ; See above ; charset = <charset> ; Mime charset definition for e-mail messages -; locale = <locale> ; Locale for LC_TIME - to get weekdays in local language +; locale = <locale> ; Locale for LC_TIME - to get weekdays in local language ; ; See your O/S documentation for proper settings for setlocale() ; templatefile = <filename> ; File name (relative to Asterisk configuration directory, -; or absolute + ; or absolute ; messagebody = Format ; Message body definition with variables ; -[template-sv_SE_email] +[template-sv_SE_email] messagebody=Hej ${MVM_NAME}:\n\n\tDu har fått ett röstbrevlåde-meddelande från ${MVM_CALLERID}.\nLängd: ${MVM_DUR}\nMailbox ${MVM_MAILBOX}\nDatum: ${MVM_DATE}. \nMeddelandet bifogas det här brevet. Om du inte kan läsa det, kontakta intern support. \nHälsningar\n\n\t\t\t\t--Asterisk\n subject = Du har fått röstmeddelande (se bilaga) fromemail = swedish-voicemail-service@stockholm.example.com fromaddress = Asterisk Röstbrevlåda charset=iso-8859-1 -attachmedia=yes +attachmedia=yes dateformat=%A, %d %B %Y at %H:%M:%S locale=sv_SE -[template-en_US_email] +[template-en_US_email] messagebody=Dear ${MVM_NAME}:\n\n\tjust wanted to let you know you were just left a ${MVM_DUR} long message \nin mailbox ${MVM_MAILBOX} from ${MVM_CALLERID}, on ${MVM_DATE}, so you might\nwant to check it when you get a chance. Thanks!\n\n\t\t\t\t--Asterisk\n subject = New voicemail charset=ascii -attachmedia=yes +attachmedia=yes dateformat=%A, %B %d, %Y at %r ;[template-sv_SE_pager] @@ -180,12 +180,12 @@ dateformat=%A, %B %d, %Y at %r ;[template-en_US_email_southern] ;templatefile = templates/email_en_US.txt ;subject = Y'all got voicemail, honey! -;charset=ascii +;charset=ascii ;[template-en_UK_email] ;templatefile = templates/email_en_us.txt ;subject = Dear old chap, you've got an electronic communique -;charset=ascii +;charset=ascii ;----------------------- Mailbox accounts -------------------------- ;Template for mailbox definition - all options diff --git a/configs/misdn.conf.sample b/configs/misdn.conf.sample index 08fb288f3..f4ca486e9 100644 --- a/configs/misdn.conf.sample +++ b/configs/misdn.conf.sample @@ -111,26 +111,26 @@ crypt_keys=test,muh ;------------------------------ JITTER BUFFER CONFIGURATION -------------------------- ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a -; SIP 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 SIP channel can accept jitter, -; thus a jitterbuffer on the receive SIP side will be used only -; if it is forced and enabled. + ; SIP 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 SIP channel can accept jitter, + ; thus a jitterbuffer on the receive SIP side will be used only + ; if it is forced and enabled. ; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP -; channel. Defaults to "no". + ; channel. Defaults to "no". ; 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. + ; 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 jbmaxsize) and "adaptive" (with -; variable size, actually the new jb of IAX2). Defaults to fixed. + ; channel. Two implementations are currently available - "fixed" + ; (with size always equals to jbmaxsize) 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/configs/modules.conf.sample b/configs/modules.conf.sample index bdcc266cd..fdcc70a63 100644 --- a/configs/modules.conf.sample +++ b/configs/modules.conf.sample @@ -21,7 +21,7 @@ autoload=yes ; Uncomment the following if you wish to use the Speech Recognition API ;preload => res_speech.so ; -; If you want, load the GTK console right away. +; If you want, load the GTK console right away. ; noload => pbx_gtkconsole.so ;load => pbx_gtkconsole.so @@ -38,6 +38,6 @@ noload => chan_console.so ; ; Only load one timing interface. If DAHDI is available, use that as it will ; provide the best results. -; +; ;noload => res_timing_dahdi.so ;noload => res_timing_pthread.so diff --git a/configs/musiconhold.conf.sample b/configs/musiconhold.conf.sample index 39df862bf..714e26403 100644 --- a/configs/musiconhold.conf.sample +++ b/configs/musiconhold.conf.sample @@ -3,14 +3,14 @@ ; [general] ;cachertclasses=yes ; use 1 instance of moh class for all users who are using it, -; decrease consumable cpu cycles and memory -; disabled by default + ; decrease consumable cpu cycles and memory + ; disabled by default ; valid mode options: -; files -- read files from a directory in any Asterisk supported +; files -- read files from a directory in any Asterisk supported ; media format -; quietmp3 -- default +; quietmp3 -- default ; mp3 -- loud ; mp3nb -- unbuffered ; quietmp3nb -- quiet unbuffered diff --git a/configs/osp.conf.sample b/configs/osp.conf.sample index 5eccf85d5..08445bb95 100644 --- a/configs/osp.conf.sample +++ b/configs/osp.conf.sample @@ -1,17 +1,17 @@ ; ; Open Settlement Protocol Sample Configuration File ; -; This file contains configuration of OSP server providers that are used by the -; Asterisk OSP module. The section "general" is reserved for global options. -; All other sections describe specific OSP Providers. The provider "default" -; is used when no provider is otherwise specified. +; This file contains configuration of OSP server providers that are used by the +; Asterisk OSP module. The section "general" is reserved for global options. +; All other sections describe specific OSP Providers. The provider "default" +; is used when no provider is otherwise specified. ; -; The "servicepoint" and "source" parameters must be configured. For most +; The "servicepoint" and "source" parameters must be configured. For most ; implementations the other parameters in this file can be left unchanged. ; [general] ; -; Enable cryptographic acceleration hardware. +; Enable cryptographic acceleration hardware. ; The default value is no. ; ;accelerate=no @@ -23,9 +23,9 @@ ; ;securityfeatures=no ; -; Defines the status of tokens that Asterisk will validate. -; 0 - signed tokens only -; 1 - unsigned tokens only +; Defines the status of tokens that Asterisk will validate. +; 0 - signed tokens only +; 1 - unsigned tokens only ; 2 - both signed and unsigned ; The default value is 0, i.e. the Asterisk will only validate signed tokens. ; If securityfeatures are disabled, Asterisk cannot validate signed tokens. @@ -45,37 +45,37 @@ ;source=domain name or [IP address in brackets] ; ; Define path and file name of crypto files. -; The default path for crypto file is /var/lib/asterisk/keys. If no path is +; The default path for crypto file is /var/lib/asterisk/keys. If no path is ; defined, crypto files will in /var/lib/asterisk/keys directory. ; -; Specify the private key file name. -; If this parameter is unspecified or not present, the default name will be the -; osp.conf section name followed by "-privatekey.pem" (for example: +; Specify the private key file name. +; If this parameter is unspecified or not present, the default name will be the +; osp.conf section name followed by "-privatekey.pem" (for example: ; default-privatekey.pem) ; If securityfeatures are disabled, this parameter is ignored. ; ;privatekey=pkey.pem ; -; Specify the local certificate file. -; If this parameter is unspecified or not present, the default name will be the -; osp.conf section name followed by "- localcert.pem " (for example: -; default-localcert.pem) +; Specify the local certificate file. +; If this parameter is unspecified or not present, the default name will be the +; osp.conf section name followed by "- localcert.pem " (for example: +; default-localcert.pem) ; If securityfeatures are disabled, this parameter is ignored. ; ;localcert=localcert.pem ; -; Specify one or more Certificate Authority key file names. If none are listed, -; a single Certificate Authority key file name is added with the default name of -; the osp.conf section name followed by "-cacert_0.pem " (for example: +; Specify one or more Certificate Authority key file names. If none are listed, +; a single Certificate Authority key file name is added with the default name of +; the osp.conf section name followed by "-cacert_0.pem " (for example: ; default-cacert_0.pem) ; If securityfeatures are disabled, this parameter is ignored. ; ;cacert=cacert_0.pem ; -; Configure parameters for OSP communication between Asterisk OSP client and OSP -; servers. +; Configure parameters for OSP communication between Asterisk OSP client and OSP +; servers. ; -; maxconnections: Max number of simultaneous connections to the provider OSP +; maxconnections: Max number of simultaneous connections to the provider OSP ; server (default=20) ; retrydelay: Extra delay between retries (default=0) ; retrylimit: Max number of retries before giving up (default=2) @@ -86,18 +86,18 @@ ;retrylimit=2 ;timeout=500 ; -; Set the authentication policy. +; Set the authentication policy. ; 0 - NO - Accept all calls. -; 1 - YES - Accept calls with valid token or no token. Block calls with -; invalid token. -; 2 - EXCLUSIVE - Accept calls with valid token. Block calls with invalid token +; 1 - YES - Accept calls with valid token or no token. Block calls with +; invalid token. +; 2 - EXCLUSIVE - Accept calls with valid token. Block calls with invalid token ; or no token. ; Default is 1, ; If securityfeatures are disabled, Asterisk cannot validate signed tokens. ; ;authpolicy=1 ; -; Set the default destination protocol. The OSP module supports SIP, H323, and +; Set the default destination protocol. The OSP module supports SIP, H323, and ; IAX protocols. The default protocol is set to SIP. ; ;defaultprotocol=SIP diff --git a/configs/oss.conf.sample b/configs/oss.conf.sample index f0ed94ea6..d29d3ac52 100644 --- a/configs/oss.conf.sample +++ b/configs/oss.conf.sample @@ -3,75 +3,75 @@ ; [general] -; General config options, with default values shown. -; You should use one section per device, with [general] being used -; for the first device and also as a template for other devices. -; -; All but 'debug' can go also in the device-specific sections. -; -; debug = 0x0 ; misc debug flags, default is 0 - -; Set the device to use for I/O -; device = /dev/dsp - -; Optional mixer command to run upon startup (e.g. to set -; volume levels, mutes, etc. -; mixer = - -; Software mic volume booster (or attenuator), useful for sound -; cards or microphones with poor sensitivity. The volume level -; is in dB, ranging from -20.0 to +20.0 -; boost = n ; mic volume boost in dB - -; Set the callerid for outgoing calls -; callerid = John Doe <555-1234> - -; autoanswer = no ; no autoanswer on call -; autohangup = yes ; hangup when other party closes -; extension = s ; default extension to call -; context = default ; default context for outgoing calls -; language = "" ; default language - -; If you set overridecontext to 'yes', then the whole dial string -; will be interpreted as an extension, which is extremely useful -; to dial SIP, IAX and other extensions which use the '@' character. -; The default is 'no' just for backward compatibility, but the -; suggestion is to change it. -; overridecontext = no ; if 'no', the last @ will start the context -; if 'yes' the whole string is an extension. - -; low level device parameters in case you have problems with the -; device driver on your operating system. You should not touch these -; unless you know what you are doing. -; queuesize = 10 ; frames in device driver -; frags = 8 ; argument to SETFRAGMENT - -;------------------------------ JITTER BUFFER CONFIGURATION -------------------------- -; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an -; OSS 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 OSS channel can't accept jitter, -; thus an enabled jitterbuffer on the receive OSS 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 an OSS -; 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". -;----------------------------------------------------------------------------------- + ; General config options, with default values shown. + ; You should use one section per device, with [general] being used + ; for the first device and also as a template for other devices. + ; + ; All but 'debug' can go also in the device-specific sections. + ; + ; debug = 0x0 ; misc debug flags, default is 0 + + ; Set the device to use for I/O + ; device = /dev/dsp + + ; Optional mixer command to run upon startup (e.g. to set + ; volume levels, mutes, etc. + ; mixer = + + ; Software mic volume booster (or attenuator), useful for sound + ; cards or microphones with poor sensitivity. The volume level + ; is in dB, ranging from -20.0 to +20.0 + ; boost = n ; mic volume boost in dB + + ; Set the callerid for outgoing calls + ; callerid = John Doe <555-1234> + + ; autoanswer = no ; no autoanswer on call + ; autohangup = yes ; hangup when other party closes + ; extension = s ; default extension to call + ; context = default ; default context for outgoing calls + ; language = "" ; default language + + ; If you set overridecontext to 'yes', then the whole dial string + ; will be interpreted as an extension, which is extremely useful + ; to dial SIP, IAX and other extensions which use the '@' character. + ; The default is 'no' just for backward compatibility, but the + ; suggestion is to change it. + ; overridecontext = no ; if 'no', the last @ will start the context + ; if 'yes' the whole string is an extension. + + ; low level device parameters in case you have problems with the + ; device driver on your operating system. You should not touch these + ; unless you know what you are doing. + ; queuesize = 10 ; frames in device driver + ; frags = 8 ; argument to SETFRAGMENT + + ;------------------------------ JITTER BUFFER CONFIGURATION -------------------------- + ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an + ; OSS 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 OSS channel can't accept jitter, + ; thus an enabled jitterbuffer on the receive OSS 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 an OSS + ; 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". + ;----------------------------------------------------------------------------------- ; below is an entry for a second console channel ; [card1] -; device = /dev/dsp1 ; alternate device + ; device = /dev/dsp1 ; alternate device ; Below are the settings to support video. You can include them ; in your general configuration as [general](+,video) @@ -79,26 +79,26 @@ ; Section names used here are only examples. [my_video](!) ; you can just include in your config -videodevice = /dev/video0 ; uses your V4L webcam as video source -videodevice = X11 ; X11 grabber. Dragging on the local display moves the origin. -videocodec = h263 ; also h261, h263p, h264, mpeg4, ... - -; video_size is the geometry used by the encoder. -; Depending on the codec your choice is restricted. -video_size = 352x288 ; the format WIDTHxHEIGHT is also ok -video_size = cif ; sqcif, qcif, cif, qvga, vga, ... - -; You can also set the geometry used for the camera, local display and remote display. -; The local window is on the right, the remote window is on the left. -; Right clicking with the mouse on a video window increases the size, -; center-clicking reduces the size. -camera_size = cif -remote_size = cif -local_size = qcif - -bitrate = 60000 ; rate told to ffmpeg. -fps = 5 ; frames per second from the source. -; qmin = 3 ; quantizer value passed to the encoder. + videodevice = /dev/video0 ; uses your V4L webcam as video source + videodevice = X11 ; X11 grabber. Dragging on the local display moves the origin. + videocodec = h263 ; also h261, h263p, h264, mpeg4, ... + + ; video_size is the geometry used by the encoder. + ; Depending on the codec your choice is restricted. + video_size = 352x288 ; the format WIDTHxHEIGHT is also ok + video_size = cif ; sqcif, qcif, cif, qvga, vga, ... + + ; You can also set the geometry used for the camera, local display and remote display. + ; The local window is on the right, the remote window is on the left. + ; Right clicking with the mouse on a video window increases the size, + ; center-clicking reduces the size. + camera_size = cif + remote_size = cif + local_size = qcif + + bitrate = 60000 ; rate told to ffmpeg. + fps = 5 ; frames per second from the source. + ; qmin = 3 ; quantizer value passed to the encoder. ; The keypad is made of an image (in any format supported by SDL_image) ; and some configuration entries indicating the location and function of buttons. @@ -115,30 +115,30 @@ fps = 5 ; frames per second from the source. ; diameter of the ellipse. ; [my_skin](!) -keypad = /tmp/keypad.jpg -region = 1 rect 19 18 67 18 28 -region = 2 rect 84 18 133 18 28 -region = 3 rect 152 18 201 18 28 -region = 4 rect 19 60 67 60 28 -region = 5 rect 84 60 133 60 28 -region = 6 rect 152 60 201 60 28 -region = 7 rect 19 103 67 103 28 -region = 8 rect 84 103 133 103 28 -region = 9 rect 152 103 201 103 28 -region = * rect 19 146 67 146 28 -region = 0 rect 84 146 133 146 28 -region = # rect 152 146 201 146 28 -region = pickup rect 229 15 267 15 40 -region = hangup rect 230 66 270 64 40 -region = mute circle 232 141 264 141 33 -region = sendvideo circle 235 185 266 185 33 -region = autoanswer rect 228 212 275 212 50 + keypad = /tmp/keypad.jpg + region = 1 rect 19 18 67 18 28 + region = 2 rect 84 18 133 18 28 + region = 3 rect 152 18 201 18 28 + region = 4 rect 19 60 67 60 28 + region = 5 rect 84 60 133 60 28 + region = 6 rect 152 60 201 60 28 + region = 7 rect 19 103 67 103 28 + region = 8 rect 84 103 133 103 28 + region = 9 rect 152 103 201 103 28 + region = * rect 19 146 67 146 28 + region = 0 rect 84 146 133 146 28 + region = # rect 152 146 201 146 28 + region = pickup rect 229 15 267 15 40 + region = hangup rect 230 66 270 64 40 + region = mute circle 232 141 264 141 33 + region = sendvideo circle 235 185 266 185 33 + region = autoanswer rect 228 212 275 212 50 ; another skin with entries for the keypad and a small font ; to write to the message boards in the skin. [skin2](!) -keypad = /tmp/kpad2.jpg -keypad_font = /tmp/font.png + keypad = /tmp/kpad2.jpg + keypad_font = /tmp/font.png ; to add video support, uncomment this and remember to install ; the keypad and keypad_font files to the right place diff --git a/configs/phone.conf.sample b/configs/phone.conf.sample index 17204501e..3d4a7c2dd 100644 --- a/configs/phone.conf.sample +++ b/configs/phone.conf.sample @@ -6,8 +6,8 @@ [interfaces] ; ; Select a mode, either the phone jack provides dialtone, reads digits, -; then starts PBX with the given extension (dialtone mode), or -; immediately provides the PBX without reading any digits or providing +; then starts PBX with the given extension (dialtone mode), or +; immediately provides the PBX without reading any digits or providing ; any dialtone (this is the immediate mode, the default). Also, you ; can set the mode to "fxo" if you have a linejack to make it operate ; properly. If you are using a Sigma Designs board you may set this to diff --git a/configs/phoneprov.conf.sample b/configs/phoneprov.conf.sample index f3df08c49..c819d6d63 100644 --- a/configs/phoneprov.conf.sample +++ b/configs/phoneprov.conf.sample @@ -6,15 +6,15 @@ ;serveraddr=192.168.1.1 ; Override address to send to the phone to use as server address. ;serveriface=eth0 ; Same as above, except an ethernet interface. -; Useful for when the interface uses DHCP and the asterisk http -; server listens on a different IP than chan_sip. + ; Useful for when the interface uses DHCP and the asterisk http + ; server listens on a different IP than chan_sip. ;serverport=5060 ; Override port to send to the phone to use as server port. default_profile=polycom ; The default profile to use if none specified in users.conf ; You can define profiles for different phones specifying what files to register ; with the provisioning server. You can define either static files, or dynamically ; generated files that can have dynamic names and point to templates that variables -; can be substituted into. You can also set arbitrary variables for the profiles +; can be substituted into. You can also set arbitrary variables for the profiles ; templates to have access to. Example: ;[example] @@ -43,48 +43,48 @@ default_profile=polycom ; The default profile to use if none specified in users. [polycom] staticdir => configs/ ; Sub directory of AST_DATA_DIR/phoneprov that static files reside -; in. This allows a request to /phoneprov/sip.cfg to pull the file -; from /phoneprov/configs/sip.cfg + ; in. This allows a request to /phoneprov/sip.cfg to pull the file + ; from /phoneprov/configs/sip.cfg mime_type => text/xml ; Default mime type to use if one isn't specified or the -; extension isn't recognized + ; extension isn't recognized static_file => bootrom.ld,application/octet-stream ; Static files the phone will download static_file => bootrom.ver,plain/text ; static_file => filename,mime-type static_file => sip.ld,application/octet-stream static_file => sip.ver,plain/text static_file => sip.cfg static_file => custom.cfg -static_file => 2201-06642-001.bootrom.ld,application/octet-stream -static_file => 2201-06642-001.sip.ld,application/octet-stream -static_file => 2345-11000-001.bootrom.ld,application/octet-stream +static_file => 2201-06642-001.bootrom.ld,application/octet-stream +static_file => 2201-06642-001.sip.ld,application/octet-stream +static_file => 2345-11000-001.bootrom.ld,application/octet-stream static_file => 2345-11300-001.bootrom.ld,application/octet-stream static_file => 2345-11300-010.bootrom.ld,application/octet-stream static_file => 2345-11300-010.sip.ld,application/octet-stream -static_file => 2345-11402-001.bootrom.ld,application/octet-stream -static_file => 2345-11402-001.sip.ld,application/octet-stream -static_file => 2345-11500-001.bootrom.ld,application/octet-stream -static_file => 2345-11500-010.bootrom.ld,application/octet-stream -static_file => 2345-11500-020.bootrom.ld,application/octet-stream -static_file => 2345-11500-030.bootrom.ld,application/octet-stream -static_file => 2345-11500-030.sip.ld,application/octet-stream -static_file => 2345-11500-040.bootrom.ld,application/octet-stream -static_file => 2345-11500-040.sip.ld,application/octet-stream -static_file => 2345-11600-001.bootrom.ld,application/octet-stream -static_file => 2345-11600-001.sip.ld,application/octet-stream -static_file => 2345-11605-001.bootrom.ld,application/octet-stream -static_file => 2345-11605-001.sip.ld,application/octet-stream -static_file => 2345-12200-001.bootrom.ld,application/octet-stream -static_file => 2345-12200-001.sip.ld,application/octet-stream -static_file => 2345-12200-002.bootrom.ld,application/octet-stream -static_file => 2345-12200-002.sip.ld,application/octet-stream +static_file => 2345-11402-001.bootrom.ld,application/octet-stream +static_file => 2345-11402-001.sip.ld,application/octet-stream +static_file => 2345-11500-001.bootrom.ld,application/octet-stream +static_file => 2345-11500-010.bootrom.ld,application/octet-stream +static_file => 2345-11500-020.bootrom.ld,application/octet-stream +static_file => 2345-11500-030.bootrom.ld,application/octet-stream +static_file => 2345-11500-030.sip.ld,application/octet-stream +static_file => 2345-11500-040.bootrom.ld,application/octet-stream +static_file => 2345-11500-040.sip.ld,application/octet-stream +static_file => 2345-11600-001.bootrom.ld,application/octet-stream +static_file => 2345-11600-001.sip.ld,application/octet-stream +static_file => 2345-11605-001.bootrom.ld,application/octet-stream +static_file => 2345-11605-001.sip.ld,application/octet-stream +static_file => 2345-12200-001.bootrom.ld,application/octet-stream +static_file => 2345-12200-001.sip.ld,application/octet-stream +static_file => 2345-12200-002.bootrom.ld,application/octet-stream +static_file => 2345-12200-002.sip.ld,application/octet-stream static_file => 2345-12200-004.bootrom.ld,application/octet-stream static_file => 2345-12200-004.sip.ld,application/octet-stream static_file => 2345-12200-005.bootrom.ld,application/octet-stream static_file => 2345-12200-005.sip.ld,application/octet-stream static_file => 2345-12500-001.bootrom.ld,application/octet-stream -static_file => 2345-12500-001.sip.ld,application/octet-stream -static_file => 2345-12560-001.bootrom.ld,application/octet-stream -static_file => 2345-12560-001.sip.ld,application/octet-stream -static_file => 2345-12600-001.bootrom.ld,application/octet-stream +static_file => 2345-12500-001.sip.ld,application/octet-stream +static_file => 2345-12560-001.bootrom.ld,application/octet-stream +static_file => 2345-12560-001.sip.ld,application/octet-stream +static_file => 2345-12600-001.bootrom.ld,application/octet-stream static_file => 2345-12600-001.sip.ld,application/octet-stream static_file => 2345-12670-001.bootrom.ld,application/octet-stream static_file => 2345-12670-001.sip.ld,application/octet-stream @@ -112,6 +112,6 @@ static_file => SoundPointIPLocalization/Korean_Korea/SoundPointIP-dictionary.xml ${MAC}.cfg => 000000000000.cfg ; Dynamically generated files. ${MAC}-phone.cfg => 000000000000-phone.cfg ; (relative to AST_DATA_DIR/phoneprov) -config/${MAC} => polycom.xml ; Dynamic Filename => template file +config/${MAC} => polycom.xml ; Dynamic Filename => template file ${MAC}-directory.xml => 000000000000-directory.xml setvar => CUSTOM_CONFIG=/var/lib/asterisk/phoneprov/configs/custom.cfg ; Custom variable diff --git a/configs/queuerules.conf.sample b/configs/queuerules.conf.sample index 5ab794be7..ccabe2cfa 100644 --- a/configs/queuerules.conf.sample +++ b/configs/queuerules.conf.sample @@ -1,12 +1,12 @@ -; It is possible to change the value of the QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY +; It is possible to change the value of the QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY ; channel variables in mid-call by defining rules in the queue for when to do so. This can allow for -; a call to be opened to more members or potentially a different set of members. -; The advantage to changing members this way as opposed to inserting the caller into a -; different queue with more members or reinserting the caller into the same queue with a different -; QUEUE_MAX_PENALTY or QUEUE_MIN_PENALTY set is that the caller does not lose his place in the queue. +; a call to be opened to more members or potentially a different set of members. +; The advantage to changing members this way as opposed to inserting the caller into a +; different queue with more members or reinserting the caller into the same queue with a different +; QUEUE_MAX_PENALTY or QUEUE_MIN_PENALTY set is that the caller does not lose his place in the queue. ; -; Note: There is a limitation to these rules; a caller will follow the penaltychange rules for -; the queue that were defined at the time the caller entered the queue. If an update to the rules is +; Note: There is a limitation to these rules; a caller will follow the penaltychange rules for +; the queue that were defined at the time the caller entered the queue. If an update to the rules is ; made during the the caller's stay in the queue, these will not be reflected for that caller. ; ; The syntax for these rules is diff --git a/configs/queues.conf.sample b/configs/queues.conf.sample index fb45a2e9e..db3ee3515 100644 --- a/configs/queues.conf.sample +++ b/configs/queues.conf.sample @@ -13,12 +13,12 @@ persistentmembers = yes ; Keep queue statistics during a reload. Default is 'no' ; keepstats = no -; +; ; AutoFill Behavior -; The old/current behavior of the queue has a serial type behavior +; The old/current behavior of the queue has a serial type behavior ; in that the queue will make all waiting callers wait in the queue ; even if there is more than one available member ready to take -; calls until the head caller is connected with the member they +; calls until the head caller is connected with the member they ; were trying to get to. The next waiting caller in line then ; becomes the head caller, and they are then connected with the ; next available member and all available members and waiting callers @@ -26,8 +26,8 @@ keepstats = no ; autofill=yes makes sure that when the waiting callers are connecting ; with available members in a parallel fashion until there are ; no more available members or no more waiting callers. This is -; probably more along the lines of how a queue should work and -; in most cases, you will want to enable this behavior. If you +; probably more along the lines of how a queue should work and +; in most cases, you will want to enable this behavior. If you ; do not specify or comment out this option, it will default to no ; to keep backward compatibility with the old behavior. ; @@ -36,22 +36,22 @@ autofill = yes ; Monitor Type ; By setting monitor-type = MixMonitor, when specifying monitor-format ; to enable recording of queue member conversations, app_queue will -; now use the new MixMonitor application instead of Monitor so +; now use the new MixMonitor application instead of Monitor so ; the concept of "joining/mixing" the in/out files now goes away ; when this is enabled. You can set the default type for all queues ; here, and then also change monitor-type for individual queues within -; queue by using the same configuration parameter within a queue +; queue by using the same configuration parameter within a queue ; configuration block. If you do not specify or comment out this option, ; it will default to the old 'Monitor' behavior to keep backward -; compatibility. +; compatibility. ; monitor-type = MixMonitor ; -; UpdateCDR behavior. +; UpdateCDR behavior. ; This option is implemented to mimic chan_agents behavior of populating -; CDR dstchannel field of a call with an agent name, which you can set -; at the login time with AddQueueMember membername parameter. -; +; CDR dstchannel field of a call with an agent name, which you can set +; at the login time with AddQueueMember membername parameter. +; ; updatecdr = no ; @@ -134,7 +134,7 @@ shared_lastcall=no ; The member's phone is rung for 5 seconds and he does not answer. ; The retry time of 4 seconds occurs. ; The queue selects a second member to call. -; +; ; How long does that second member's phone ring? Does it ring for 5 seconds ; since the timeout set in app_queue is 5 seconds? Does it ring for 1 second since ; the caller has been in the queue for 9 seconds and is supposed to be removed after @@ -143,8 +143,8 @@ shared_lastcall=no ; rather use the time specified in the configuration file even if it means having the ; caller stay in the queue longer than the time specified in the application argument. ; For the scenario described above, timeoutpriority=conf would result in the second -; member's phone ringing for 5 seconds. By specifying "app" as the value for -; timeoutpriority, you are saying that the timeout specified as the argument to the +; member's phone ringing for 5 seconds. By specifying "app" as the value for +; timeoutpriority, you are saying that the timeout specified as the argument to the ; Queue application is more important. In the scenario above, timeoutpriority=app ; would result in the second member's phone ringing for 1 second. ; @@ -152,7 +152,7 @@ shared_lastcall=no ; and the configuration file timeout is set to 0, but the application argument timeout is ; non-zero, then the timeoutpriority is ignored and the application argument is used as ; the timeout. Furthermore, if no application argument timeout is specified, then the -; timeoutpriority option is ignored and the configuration file timeout is always used +; timeoutpriority option is ignored and the configuration file timeout is always used ; when calling queue members. ; ; In timeoutpriority=conf mode however timeout specified in config file will take higher @@ -170,8 +170,8 @@ shared_lastcall=no ;timeoutpriority = app|conf ; ;-----------------------END QUEUE TIMING OPTIONS--------------------------------- -; Weight of queue - when compared to other queues, higher weights get -; first shot at available channels when the same channel is included in +; Weight of queue - when compared to other queues, higher weights get +; first shot at available channels when the same channel is included in ; more than one queue. ; ;weight=0 @@ -196,21 +196,21 @@ shared_lastcall=no ; ;maxlen = 0 ; -; If set to yes, just prior to the caller being bridged with a queue member +; If set to yes, just prior to the caller being bridged with a queue member ; the following variables will be set ; MEMBERINTERFACE is the interface name (eg. Agent/1234) ; MEMBERNAME is the member name (eg. Joe Soap) -; MEMBERCALLS is the number of calls that interface has taken, -; MEMBERLASTCALL is the last time the member took a call. -; MEMBERPENALTY is the penalty of the member +; MEMBERCALLS is the number of calls that interface has taken, +; MEMBERLASTCALL is the last time the member took a call. +; MEMBERPENALTY is the penalty of the member ; MEMBERDYNAMIC indicates if a member is dynamic or not ; MEMBERREALTIME indicates if a member is realtime or not ; ;setinterfacevar=no ; -; If set to yes, just prior to the caller being bridged with a queue member +; If set to yes, just prior to the caller being bridged with a queue member ; the following variables will be set: -; QEHOLDTIME callers hold time +; QEHOLDTIME callers hold time ; QEORIGINALPOS original position of the caller in the queue ; ;setqueueentryvar=no @@ -220,7 +220,7 @@ shared_lastcall=no ; and just prior to the caller leaving the queue ; QUEUENAME name of the queue ; QUEUEMAX maxmimum number of calls allowed -; QUEUESTRATEGY the strategy of the queue; +; QUEUESTRATEGY the strategy of the queue; ; QUEUECALLS number of calls currently in the queue ; QUEUEHOLDTIME current average hold time ; QUEUECOMPLETED number of completed calls for the queue @@ -231,17 +231,17 @@ shared_lastcall=no ;setqueuevar=no ; ; if set, run this macro when connected to the queue member -; you can override this macro by setting the macro option on +; you can override this macro by setting the macro option on ; the queue application ; ; membermacro=somemacro -; How often to announce queue position and/or estimated +; How often to announce queue position and/or estimated ; holdtime to caller (0=off) ; Note that this value is ignored if the caller's queue ; position has changed (see min-announce-frequency) ; -;announce-frequency = 90 +;announce-frequency = 90 ; ; The absolute minimum time between the start of each ; queue position and/or estimated holdtime announcement @@ -300,26 +300,26 @@ shared_lastcall=no ; ; queue-thankyou= ; -; ("You are now first in line.") -;queue-youarenext = queue-youarenext -; ("There are") + ; ("You are now first in line.") +;queue-youarenext = queue-youarenext + ; ("There are") ;queue-thereare = queue-thereare -; ("calls waiting.") + ; ("calls waiting.") ;queue-callswaiting = queue-callswaiting -; ("The current est. holdtime is") + ; ("The current est. holdtime is") ;queue-holdtime = queue-holdtime -; ("minutes.") + ; ("minutes.") ;queue-minutes = queue-minutes -; ("seconds.") + ; ("seconds.") ;queue-seconds = queue-seconds -; ("Thank you for your patience.") + ; ("Thank you for your patience.") ;queue-thankyou = queue-thankyou -; ("Hold time") + ; ("Hold time") ;queue-reporthold = queue-reporthold -; ("All reps busy / wait for next") + ; ("All reps busy / wait for next") ;periodic-announce = queue-periodic-announce ; -; A set of periodic announcements can be defined by separating +; A set of periodic announcements can be defined by separating ; periodic announcements to reproduce by commas. For example: ;periodic-announce = queue-periodic-announce,your-call-is-important,please-wait ; @@ -358,7 +358,7 @@ shared_lastcall=no ; ; You can specify the options supplied to MixMonitor by calling ; Set(MONITOR_OPTIONS=av(<x>)V(<x>)W(<x>)) -; The 'b' option for MixMonitor (only save audio to the file while bridged) is +; The 'b' option for MixMonitor (only save audio to the file while bridged) is ; implied. ; ; You can specify a post recording command to be executed after the end of @@ -379,9 +379,9 @@ shared_lastcall=no ; whether a caller may join a queue depending on several factors of member availability. ; Similarly, then leavewhenempty option controls whether a caller may remain in a queue ; he has already joined. Both options take a comma-separated list of factors which -; contribute towards whether a caller may join/remain in the queue. The list of +; contribute towards whether a caller may join/remain in the queue. The list of ; factors which contribute to these option is as follows: -; +; ; paused: a member is not considered available if he is paused ; penalty: a member is not considered available if his penalty is less than QUEUE_MAX_PENALTY ; inuse: a member is not considered available if he is currently on a call @@ -394,14 +394,14 @@ shared_lastcall=no ; current device state. ; wrapup: A member is not considered available if he is currently in his wrapuptime after ; taking a call. -; +; ; For the "joinempty" option, when a caller attempts to enter a queue, the members of that ; queue are examined. If all members are deemed to be unavailable due to any of the conditions ; listed for the "joinempty" option, then the caller will be unable to enter the queue. For the ; "leavewhenempty" option, the state of the members of the queue are checked periodically during ; the caller's stay in the queue. If all of the members are unavailable due to any of the above ; conditions, then the caller will be removed from the queue. -; +; ; Some examples: ; ;joinempty = paused,inuse,invalid @@ -411,7 +411,7 @@ shared_lastcall=no ; ;leavewhenempty = inuse,ringing ; -; A caller will be removed from the queue if at least one member cannot be found +; A caller will be removed from the queue if at least one member cannot be found ; who is not on the phone, or whose phone is not ringing. ; ; For the sake of backwards-compatibility, the joinempty and leavewhenempty @@ -461,7 +461,7 @@ shared_lastcall=no ; ; timeoutrestart = no ; -; If you wish to implement a rule defined in queuerules.conf (see +; If you wish to implement a rule defined in queuerules.conf (see ; configs/queuerules.conf.sample from the asterisk source directory for ; more information about penalty rules) by default, you may specify this ; by setting defaultrule to the rule's name @@ -501,5 +501,5 @@ shared_lastcall=no ; ;member => Agent/@1 ; Any agent in group 1 ;member => Agent/:1,1 ; Any agent in group 1, wait for first -; available, but consider with penalty + ; available, but consider with penalty diff --git a/configs/res_odbc.conf.sample b/configs/res_odbc.conf.sample index 85bd8f45a..7e9405dde 100644 --- a/configs/res_odbc.conf.sample +++ b/configs/res_odbc.conf.sample @@ -1,4 +1,4 @@ -;;; odbc setup file +;;; odbc setup file ; ENV is a global set of environmental variables that will get set. ; Note that all environmental variables can be seen by all connections, @@ -49,11 +49,11 @@ pre-connect => yes sanitysql => select count(*) from systables ; forcecommit => no ; Default to committing uncommitted transactions? ; isolation => read_committed ; Isolation level; supported levels are: -; read_uncommitted, read_committed, repeatable_read, -; serializable. Note that not all databases support -; all isolation levels (e.g. Postgres only supports -; repeatable_read and serializable). See database -; documentation for further information. + ; read_uncommitted, read_committed, repeatable_read, + ; serializable. Note that not all databases support + ; all isolation levels (e.g. Postgres only supports + ; repeatable_read and serializable). See database + ; documentation for further information. ; ; Many databases have a default of '\' to escape special characters. MS SQL ; Server does not. diff --git a/configs/res_snmp.conf.sample b/configs/res_snmp.conf.sample index fb7bfd750..0aa042b08 100644 --- a/configs/res_snmp.conf.sample +++ b/configs/res_snmp.conf.sample @@ -15,7 +15,7 @@ [general] ; We run as a subagent per default -- to run as a full agent -; we must run as root (to be able to bind to port 161) +; we must run as root (to be able to bind to port 161) ;subagent = yes ; SNMP must be explicitly enabled to be active ;enabled = yes diff --git a/configs/rpt.conf.sample b/configs/rpt.conf.sample index 871793d65..f1c86a11b 100644 --- a/configs/rpt.conf.sample +++ b/configs/rpt.conf.sample @@ -28,13 +28,13 @@ ;funcchar = * ; function lead-in character (defaults to '*') ;endchar = # ; command mode end character (defaults to '#') ;;nobusyout=yes ; (optional) Do not busy-out reverse-patch when -; normal patch in use + ; normal patch in use ;hangtime=1000 ; squelch tail hang time (in ms) (optional) ;totime=100000 ; transmit time-out time (in ms) (optional) ;idtime=30000 ; id interval time (in ms) (optional) ;politeid=30000 ; time in milliseconds before ID timer -; expires to try and ID in the tail. -; (optional, default is 30000). + ; expires to try and ID in the tail. + ; (optional, default is 30000). ;idtalkover=|iwb6nil/rpt ; Talkover ID (optional) default is none ;unlinkedct=ct2 ; unlinked courtesy tone (optional) default is none @@ -69,13 +69,13 @@ ;funcchar = * ; function lead-in character (defaults to '*') ;endchar = # ; command mode end character (defaults to '#') ;;nobusyout=yes ; (optional) Do not busy-out reverse-patch when -; normal patch in use + ; normal patch in use ;hangtime=1000 ; squelch tail hang time (in ms) (optional) ;totime=100000 ; transmit time-out time (in ms) (optional) ;idtime=30000 ; id interval time (in ms) (optional) ;politeid=30000 ; time in milliseconds before ID timer -; expires to try and ID in the tail. -; (optional, default is 30000). + ; expires to try and ID in the tail. + ; (optional, default is 30000). ;idtalkover=|iwb6nil/rpt ; Talkover ID (optional) default is none ;unlinkedct=ct2 ; unlinked courtesy tone (optional) default is none @@ -86,9 +86,9 @@ ; specify the rxchannel and the txchannel will be assumed from the rxchannel ;txchannel = DAHDI/6 ; Tx audio/signalling channel ;functions = functions-remote -;remote = ft897 ; Set remote=y for dumb remote or -; remote=ft897 for Yaesu FT-897 or -; remote=rbi for Doug Hall RBI1 +;remote = ft897 ; Set remote=y for dumb remote or + ; remote=ft897 for Yaesu FT-897 or + ; remote=rbi for Doug Hall RBI1 ;iobase = 0x378 ; Specify IO port for parallel port (optional) ;[functions-repeater] @@ -106,7 +106,7 @@ ;6=autopatchup ; Autopatch up ;0=autopatchdn ; Autopatch down -;90=cop,1 ; System warm boot +;90=cop,1 ; System warm boot ;91=cop,2 ; System enable ;92=cop,3 ; System disable @@ -135,7 +135,7 @@ ; Single frequencies are created by setting freq1 or freq2 to zero. ; ; |m - Morse escape sequence -; +; ; Sends Morse code at the telemetry amplitude and telemetry frequency as defined in the ; [morse] section. ; @@ -150,15 +150,15 @@ ;ct1=|t(350,0,100,2048)(500,0,100,2048)(660,0,100,2048) -;ct2=|t(660,880,150,2048) -;ct3=|t(440,0,150,2048) +;ct2=|t(660,880,150,2048) +;ct3=|t(440,0,150,2048) ;ct4=|t(550,0,150,2048) ;ct5=|t(660,0,150,2048) ;ct6=|t(880,0,150,2048) ;ct7=|t(660,440,150,2048) ;ct8=|t(700,1100,150,2048) -;remotetx=|t(2000,0,75,2048)(0,0,75,0)(1600,0,75,2048); -;remotemon=|t(1600,0,75,2048) +;remotetx=|t(2000,0,75,2048)(0,0,75,0)(1600,0,75,2048); +;remotemon=|t(1600,0,75,2048) ;cmdmode=|t(900,903,200,2048) ;functcomplete=|t(1000,0,100,2048)(0,0,100,0)(1000,0,100,2048) @@ -168,7 +168,7 @@ ;speed=20 ; Approximate speed in WPM ;frequency=800 ; Morse Telemetry Frequency ;amplitude=4096 ; Morse Telemetry Amplitude -;idfrequency=330 ; Morse ID Frequency +;idfrequency=330 ; Morse ID Frequency ;idamplitude=2048 ; Morse ID Amplitude ;[nodes] diff --git a/configs/rtp.conf.sample b/configs/rtp.conf.sample index 615b6fe46..224dc2abe 100644 --- a/configs/rtp.conf.sample +++ b/configs/rtp.conf.sample @@ -18,8 +18,8 @@ rtpend=20000 ; allowed to continue (in 'samples', 1/8000 of a second) ; ;dtmftimeout=3000 -; rtcpinterval = 5000 ; Milliseconds between rtcp reports -;(min 500, max 60000, default 5000) +; rtcpinterval = 5000 ; Milliseconds between rtcp reports + ;(min 500, max 60000, default 5000) ; ; Enable strict RTP protection. This will drop RTP packets that ; do not come from the source of the RTP stream. This option is diff --git a/configs/say.conf.sample b/configs/say.conf.sample index f592b780a..163114875 100644 --- a/configs/say.conf.sample +++ b/configs/say.conf.sample @@ -1,11 +1,11 @@ -; +; ; language configuration ; [general] mode=old ; method for playing numbers and dates -; old - using asterisk core function -; new - using this configuration file + ; old - using asterisk core function + ; new - using this configuration file ; The new language routines produce strings of the form ; prefix:[format:]data @@ -66,7 +66,7 @@ mode=old ; method for playing numbers and dates ; date:M:200604172030.00-4-102 ; date:p:200604172030.00-4-102 ; -; +; ; Remember, normally X Z N are special, and the search is ; case insensitive, so you must use [X] [N] [Z] .. if you ; want exact match. @@ -75,126 +75,126 @@ mode=old ; method for playing numbers and dates ; language-independent [digit-base](!) ; base rule for digit strings -; XXX incomplete yet -_digit:[0-9] => digits/${SAY} -_digit:[-] => letters/dash -_digit:[*] => letters/star -_digit:[@] => letters/at -_digit:[0-9]. => digit:${SAY:0:1}, digit:${SAY:1} + ; XXX incomplete yet + _digit:[0-9] => digits/${SAY} + _digit:[-] => letters/dash + _digit:[*] => letters/star + _digit:[@] => letters/at + _digit:[0-9]. => digit:${SAY:0:1}, digit:${SAY:1} [date-base](!) ; base rules for dates and times -; the 'SAY' variable contains YYYYMMDDHHmm.ss-dow-doy -; these rule map the strftime attributes. -_date:Y:. => num:${SAY:0:4} ; year, 19xx -_date:[Bb]:. => digits/mon-$[${SAY:4:2}-1] ; month name, 0..11 -_date:[Aa]:. => digits/day-${SAY:16:1} ; day of week -_date:[de]:. => num:${SAY:6:2} ; day of month -_date:[hH]:. => num:${SAY:8:2} ; hour -_date:[I]:. => num:$[${SAY:8:2} % 12] ; hour 0-12 -_date:[M]:. => num:${SAY:10:2} ; minute -; XXX too bad the '?' function does not remove the quotes -; _date:[pP]:. => digits/$[ ${SAY:10:2} > 12 ? "p-m" :: "a-m"] ; am pm -_date:[pP]:. => digits/p-m ; am pm -_date:[S]:. => num:${SAY:13:2} ; seconds + ; the 'SAY' variable contains YYYYMMDDHHmm.ss-dow-doy + ; these rule map the strftime attributes. + _date:Y:. => num:${SAY:0:4} ; year, 19xx + _date:[Bb]:. => digits/mon-$[${SAY:4:2}-1] ; month name, 0..11 + _date:[Aa]:. => digits/day-${SAY:16:1} ; day of week + _date:[de]:. => num:${SAY:6:2} ; day of month + _date:[hH]:. => num:${SAY:8:2} ; hour + _date:[I]:. => num:$[${SAY:8:2} % 12] ; hour 0-12 + _date:[M]:. => num:${SAY:10:2} ; minute + ; XXX too bad the '?' function does not remove the quotes + ; _date:[pP]:. => digits/$[ ${SAY:10:2} > 12 ? "p-m" :: "a-m"] ; am pm + _date:[pP]:. => digits/p-m ; am pm + _date:[S]:. => num:${SAY:13:2} ; seconds [en-base](!) -_[n]um:0. => num:${SAY:1} -_[n]um:X => digits/${SAY} -_[n]um:1X => digits/${SAY} -_[n]um:[2-9]0 => digits/${SAY} -_[n]um:[2-9][1-9] => digits/${SAY:0:1}0, num:${SAY:1} -_[n]um:XXX => num:${SAY:0:1}, digits/hundred, num:${SAY:1} - -_[n]um:XXXX => num:${SAY:0:1}, digits/thousand, num:${SAY:1} -_[n]um:XXXXX => num:${SAY:0:2}, digits/thousand, num:${SAY:2} -_[n]um:XXXXXX => num:${SAY:0:3}, digits/thousand, num:${SAY:3} - -_[n]um:XXXXXXX => num:${SAY:0:1}, digits/million, num:${SAY:1} -_[n]um:XXXXXXXX => num:${SAY:0:2}, digits/million, num:${SAY:2} -_[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/million, num:${SAY:3} - -_[n]um:XXXXXXXXXX => num:${SAY:0:1}, digits/billion, num:${SAY:1} -_[n]um:XXXXXXXXXXX => num:${SAY:0:2}, digits/billion, num:${SAY:2} -_[n]um:XXXXXXXXXXXX => num:${SAY:0:3}, digits/billion, num:${SAY:3} - -; enumeration -_e[n]um:X => digits/h-${SAY} -_e[n]um:1X => digits/h-${SAY} -_e[n]um:[2-9]0 => digits/h-${SAY} -_e[n]um:[2-9][1-9] => num:${SAY:0:1}0, digits/h-${SAY:1} -_e[n]um:[1-9]XX => num:${SAY:0:1}, digits/hundred, enum:${SAY:1} + _[n]um:0. => num:${SAY:1} + _[n]um:X => digits/${SAY} + _[n]um:1X => digits/${SAY} + _[n]um:[2-9]0 => digits/${SAY} + _[n]um:[2-9][1-9] => digits/${SAY:0:1}0, num:${SAY:1} + _[n]um:XXX => num:${SAY:0:1}, digits/hundred, num:${SAY:1} + + _[n]um:XXXX => num:${SAY:0:1}, digits/thousand, num:${SAY:1} + _[n]um:XXXXX => num:${SAY:0:2}, digits/thousand, num:${SAY:2} + _[n]um:XXXXXX => num:${SAY:0:3}, digits/thousand, num:${SAY:3} + + _[n]um:XXXXXXX => num:${SAY:0:1}, digits/million, num:${SAY:1} + _[n]um:XXXXXXXX => num:${SAY:0:2}, digits/million, num:${SAY:2} + _[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/million, num:${SAY:3} + + _[n]um:XXXXXXXXXX => num:${SAY:0:1}, digits/billion, num:${SAY:1} + _[n]um:XXXXXXXXXXX => num:${SAY:0:2}, digits/billion, num:${SAY:2} + _[n]um:XXXXXXXXXXXX => num:${SAY:0:3}, digits/billion, num:${SAY:3} + + ; enumeration + _e[n]um:X => digits/h-${SAY} + _e[n]um:1X => digits/h-${SAY} + _e[n]um:[2-9]0 => digits/h-${SAY} + _e[n]um:[2-9][1-9] => num:${SAY:0:1}0, digits/h-${SAY:1} + _e[n]um:[1-9]XX => num:${SAY:0:1}, digits/hundred, enum:${SAY:1} [it](digit-base,date-base) -_[n]um:0. => num:${SAY:1} -_[n]um:X => digits/${SAY} -_[n]um:1X => digits/${SAY} -_[n]um:[2-9]0 => digits/${SAY} -_[n]um:[2-9][1-9] => digits/${SAY:0:1}0, num:${SAY:1} -_[n]um:1XX => digits/hundred, num:${SAY:1} -_[n]um:[2-9]XX => num:${SAY:0:1}, digits/hundred, num:${SAY:1} - -_[n]um:1XXX => digits/thousand, num:${SAY:1} -_[n]um:[2-9]XXX => num:${SAY:0:1}, digits/thousands, num:${SAY:1} -_[n]um:XXXXX => num:${SAY:0:2}, digits/thousands, num:${SAY:2} -_[n]um:XXXXXX => num:${SAY:0:3}, digits/thousands, num:${SAY:3} - -_[n]um:1XXXXXX => num:${SAY:0:1}, digits/million, num:${SAY:1} -_[n]um:[2-9]XXXXXX => num:${SAY:0:1}, digits/millions, num:${SAY:1} -_[n]um:XXXXXXXX => num:${SAY:0:2}, digits/millions, num:${SAY:2} -_[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/millions, num:${SAY:3} - -_datetime::. => date:AdBY 'digits/at' IMp:${SAY} -_date::. => date:AdBY:${SAY} -_time::. => date:IMp:${SAY} + _[n]um:0. => num:${SAY:1} + _[n]um:X => digits/${SAY} + _[n]um:1X => digits/${SAY} + _[n]um:[2-9]0 => digits/${SAY} + _[n]um:[2-9][1-9] => digits/${SAY:0:1}0, num:${SAY:1} + _[n]um:1XX => digits/hundred, num:${SAY:1} + _[n]um:[2-9]XX => num:${SAY:0:1}, digits/hundred, num:${SAY:1} + + _[n]um:1XXX => digits/thousand, num:${SAY:1} + _[n]um:[2-9]XXX => num:${SAY:0:1}, digits/thousands, num:${SAY:1} + _[n]um:XXXXX => num:${SAY:0:2}, digits/thousands, num:${SAY:2} + _[n]um:XXXXXX => num:${SAY:0:3}, digits/thousands, num:${SAY:3} + + _[n]um:1XXXXXX => num:${SAY:0:1}, digits/million, num:${SAY:1} + _[n]um:[2-9]XXXXXX => num:${SAY:0:1}, digits/millions, num:${SAY:1} + _[n]um:XXXXXXXX => num:${SAY:0:2}, digits/millions, num:${SAY:2} + _[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/millions, num:${SAY:3} + + _datetime::. => date:AdBY 'digits/at' IMp:${SAY} + _date::. => date:AdBY:${SAY} + _time::. => date:IMp:${SAY} [en](en-base,date-base,digit-base) -_datetime::. => date:AdBY 'digits/at' IMp:${SAY} -_date::. => date:AdBY:${SAY} -_time::. => date:IMp:${SAY} + _datetime::. => date:AdBY 'digits/at' IMp:${SAY} + _date::. => date:AdBY:${SAY} + _time::. => date:IMp:${SAY} [de](date-base,digit-base) -_[n]um:0. => num:${SAY:1} -_[n]um:X => digits/${SAY} -_[n]um:1X => digits/${SAY} -_[n]um:[2-9]0 => digits/${SAY} -_[n]um:[2-9][1-9] => digits/${SAY:1}-and, digits/${SAY:0:1}0 -_[n]um:1XX => digits/ein, digits/hundred, num:${SAY:1} -_[n]um:[2-9]XX => digits/${SAY:0:1}, digits/hundred, num:${SAY:1} -_[n]um:1XXX => digits/ein, digits/thousand, num:${SAY:1} -_[n]um:[2-9]XXX => digits/${SAY:0:1}, digits/thousand, num:${SAY:1} -_[n]um:XXXXX => num:${SAY:0:2}, digits/thousand, num:${SAY:2} -_[n]um:X00XXX => digits/${SAY:0:1}, digits/hundred, digits/thousand, num:${SAY:3} -_[n]um:XXXXXX => digits/${SAY:0:1}, digits/hundred, num:${SAY:1} -_[n]um:1XXXXXX => digits/eine, digits/million, num:${SAY:1} -_[n]um:[2-9]XXXXXX => digits/${SAY:0:1}, digits/millions, num:${SAY:1} -_[n]um:XXXXXXXX => num:${SAY:0:2}, digits/millions, num:${SAY:2} -_[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/millions, num:${SAY:3} - -_datetime::. => date:AdBY 'digits/at' IMp:${SAY} -_date::. => date:AdBY:${SAY} -_time::. => date:IMp:${SAY} + _[n]um:0. => num:${SAY:1} + _[n]um:X => digits/${SAY} + _[n]um:1X => digits/${SAY} + _[n]um:[2-9]0 => digits/${SAY} + _[n]um:[2-9][1-9] => digits/${SAY:1}-and, digits/${SAY:0:1}0 + _[n]um:1XX => digits/ein, digits/hundred, num:${SAY:1} + _[n]um:[2-9]XX => digits/${SAY:0:1}, digits/hundred, num:${SAY:1} + _[n]um:1XXX => digits/ein, digits/thousand, num:${SAY:1} + _[n]um:[2-9]XXX => digits/${SAY:0:1}, digits/thousand, num:${SAY:1} + _[n]um:XXXXX => num:${SAY:0:2}, digits/thousand, num:${SAY:2} + _[n]um:X00XXX => digits/${SAY:0:1}, digits/hundred, digits/thousand, num:${SAY:3} + _[n]um:XXXXXX => digits/${SAY:0:1}, digits/hundred, num:${SAY:1} + _[n]um:1XXXXXX => digits/eine, digits/million, num:${SAY:1} + _[n]um:[2-9]XXXXXX => digits/${SAY:0:1}, digits/millions, num:${SAY:1} + _[n]um:XXXXXXXX => num:${SAY:0:2}, digits/millions, num:${SAY:2} + _[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/millions, num:${SAY:3} + + _datetime::. => date:AdBY 'digits/at' IMp:${SAY} + _date::. => date:AdBY:${SAY} + _time::. => date:IMp:${SAY} [hu](digit-base,date-base) -_[n]um:0. => num:${SAY:1} -_[n]um:X => digits/${SAY} -_[n]um:1[1-9] => digits/10en, digits/${SAY:1} -_[n]um:2[1-9] => digits/20on, digits/${SAY:1} -_[n]um:[1-9]0 => digits/${SAY} -_[n]um:[3-9][1-9] => digits/${SAY:0:1}0, num:${SAY:1} -_[n]um:XXX => num:${SAY:0:1}, digits/hundred, num:${SAY:1} - -_[n]um:XXXX => num:${SAY:0:1}, digits/thousand, num:${SAY:1} -_[n]um:XXXXX => num:${SAY:0:2}, digits/thousand, num:${SAY:2} -_[n]um:XXXXXX => num:${SAY:0:3}, digits/thousand, num:${SAY:3} - -_[n]um:XXXXXXX => num:${SAY:0:1}, digits/million, num:${SAY:1} -_[n]um:XXXXXXXX => num:${SAY:0:2}, digits/million, num:${SAY:2} -_[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/million, num:${SAY:3} - -_[n]um:XXXXXXXXXX => num:${SAY:0:1}, digits/billion, num:${SAY:1} -_[n]um:XXXXXXXXXXX => num:${SAY:0:2}, digits/billion, num:${SAY:2} -_[n]um:XXXXXXXXXXXX => num:${SAY:0:3}, digits/billion, num:${SAY:3} - -_datetime::. => date:YBdA k 'ora' M 'perc':${SAY} -_date::. => date:YBdA:${SAY} -_time::. => date:k 'ora' M 'perc':${SAY} + _[n]um:0. => num:${SAY:1} + _[n]um:X => digits/${SAY} + _[n]um:1[1-9] => digits/10en, digits/${SAY:1} + _[n]um:2[1-9] => digits/20on, digits/${SAY:1} + _[n]um:[1-9]0 => digits/${SAY} + _[n]um:[3-9][1-9] => digits/${SAY:0:1}0, num:${SAY:1} + _[n]um:XXX => num:${SAY:0:1}, digits/hundred, num:${SAY:1} + + _[n]um:XXXX => num:${SAY:0:1}, digits/thousand, num:${SAY:1} + _[n]um:XXXXX => num:${SAY:0:2}, digits/thousand, num:${SAY:2} + _[n]um:XXXXXX => num:${SAY:0:3}, digits/thousand, num:${SAY:3} + + _[n]um:XXXXXXX => num:${SAY:0:1}, digits/million, num:${SAY:1} + _[n]um:XXXXXXXX => num:${SAY:0:2}, digits/million, num:${SAY:2} + _[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/million, num:${SAY:3} + + _[n]um:XXXXXXXXXX => num:${SAY:0:1}, digits/billion, num:${SAY:1} + _[n]um:XXXXXXXXXXX => num:${SAY:0:2}, digits/billion, num:${SAY:2} + _[n]um:XXXXXXXXXXXX => num:${SAY:0:3}, digits/billion, num:${SAY:3} + + _datetime::. => date:YBdA k 'ora' M 'perc':${SAY} + _date::. => date:YBdA:${SAY} + _time::. => date:k 'ora' M 'perc':${SAY} diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index 862b482d4..20467f1aa 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -3,7 +3,7 @@ ; ; SIP dial strings ;----------------------------------------------------------- -; In the dialplan (extensions.conf) you can use several +; In the dialplan (extensions.conf) you can use several ; syntaxes for dialing SIP devices. ; SIP/devicename ; SIP/username@domain (SIP uri) @@ -17,11 +17,11 @@ ; username@domain ; Call any SIP user on the Internet ; (Don't forget to enable DNS SRV records if you want to use this) -; +; ; devicename/extension ; If you define a SIP proxy as a peer below, you may call -; SIP/proxyhostname/user or SIP/user@proxyhostname -; where the proxyhostname is defined in a section below +; SIP/proxyhostname/user or SIP/user@proxyhostname +; where the proxyhostname is defined in a section below ; This syntax also works with ATA's with FXO ports ; ; SIP/username[:password[:md5secret[:authname]]]@host[:port] @@ -54,7 +54,7 @@ ; When naming devices, make sure you understand how Asterisk matches calls ; that come in. ; 1. Asterisk checks the SIP From: address username and matches against -; names of devices with type=user +; names of devices with type=user ; The name is the text between square brackets [name] ; 2. Asterisk checks the From: addres and matches the list of devices ; with a type=peer @@ -64,14 +64,14 @@ ; Don't mix extensions with the names of the devices. Devices need a unique ; name. The device name is *not* used as phone numbers. Phone numbers are ; anything you declare as an extension in the dialplan (extensions.conf). -; +; ; When setting up trunks, make sure there's no risk that any From: username -; (caller ID) will match any of your device names, because then Asterisk +; (caller ID) will match any of your device names, because then Asterisk ; might match the wrong device. ; ; Note: The parameter "username" is not the username and in most cases is ; not needed at all. Check below. In later releases, it's renamed -; to "defaultuser" which is a better name, since it is used in +; to "defaultuser" which is a better name, since it is used in ; combination with the "defaultip" setting. ;----------------------------------------------------------------------------- @@ -81,25 +81,25 @@ ; You are encouraged to use the dialplan groupcount functionality ; to enforce call limits instead of using this channel-specific method. ; -; You can still set limits per device in sip.conf or in a database by using +; You can still set limits per device in sip.conf or in a database by using ; "setvar" to set variables that can be used in the dialplan for various limits. [general] context=default ; Default context for incoming calls ;allowguest=no ; Allow or reject guest calls (default is yes) ;match_auth_username=yes ; if available, match user entry using the -; 'username' field from the authentication line -; instead of the From: field. + ; 'username' field from the authentication line + ; instead of the From: field. allowoverlap=no ; Disable overlap dialing support. (Default is yes) ;allowtransfer=no ; Disable all transfers (unless enabled in peers or users) -; Default is enabled + ; Default is enabled ;realm=mydomain.tld ; Realm for digest authentication -; defaults to "asterisk". If you set a system name in -; asterisk.conf, it defaults to that system name -; Realms MUST be globally unique according to RFC 3261 -; Set this to your host name or domain name + ; defaults to "asterisk". If you set a system name in + ; asterisk.conf, it defaults to that system name + ; Realms MUST be globally unique according to RFC 3261 + ; Set this to your host name or domain name udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all) -; Optionally add a port number, 192.168.1.1:5062 (default is port 5060) + ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060) ; ; Note that the TCP and TLS support for chan_sip is currently considered @@ -109,20 +109,20 @@ udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0 ; tcpenable=no ; Enable server for incoming TCP connections (default is no) tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces) -; Optionally add a port number, 192.168.1.1:5062 (default is port 5060) + ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060) ;tlsenable=no ; Enable server for incoming TLS (secure) connections (default is no) ;tlsbindaddr=0.0.0.0 ; IP address for TLS server to bind to (0.0.0.0) binds to all interfaces) -; Optionally add a port number, 192.168.1.1:5063 (default is port 5061) -; Remember that the IP address must match the common name (hostname) in the -; certificate, so you don't want to bind a TLS socket to multiple IP addresses. + ; Optionally add a port number, 192.168.1.1:5063 (default is port 5061) + ; Remember that the IP address must match the common name (hostname) in the + ; certificate, so you don't want to bind a TLS socket to multiple IP addresses. ;tlscertfile=</path/to/certificate.pem> ; Certificate file (*.pem only) to use for TLS connections -; default is to look for "asterisk.pem" in current directory + ; default is to look for "asterisk.pem" in current directory ;tlsprivatekey=</path/to/private.pem> ; Private key file (*.pem only) for TLS connections. -; If no tlsprivatekey is specified, tlscertfile is searched for -; for both public and private key. + ; If no tlsprivatekey is specified, tlscertfile is searched for + ; for both public and private key. ;tlscafile=</path/to/certificate> ; If the server your connecting to uses a self signed certificate @@ -130,12 +130,12 @@ tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 ; verify the authenticity of their certificate. ;tlscadir=</path/to/ca/dir> -; A directory full of CA certificates. The files must be named with -; the CA subject name hash value. -; (see man SSL_CTX_load_verify_locations for more info) +; A directory full of CA certificates. The files must be named with +; the CA subject name hash value. +; (see man SSL_CTX_load_verify_locations for more info) ;tlsdontverifyserver=[yes|no] -; If set to yes, don't verify the servers certificate when acting as +; If set to yes, don't verify the servers certificate when acting as ; a client. If you don't have the server's CA certificate you can ; set this and it will connect without requiring tlscafile to be set. ; Default is no. @@ -146,20 +146,20 @@ tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 ; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS ; ;tlsclientmethod=tlsv1 ; values include tlsv1, sslv3, sslv2. -; Specify protocol for outbound client connections. -; If left unspecified, the default is sslv2. + ; Specify protocol for outbound client connections. + ; If left unspecified, the default is sslv2. srvlookup=yes ; Enable DNS SRV lookups on outbound calls -; Note: Asterisk only uses the first host -; in SRV records -; Disabling DNS SRV lookups disables the -; ability to place SIP calls based on domain -; names to some other SIP users on the Internet + ; Note: Asterisk only uses the first host + ; in SRV records + ; Disabling DNS SRV lookups disables the + ; ability to place SIP calls based on domain + ; names to some other SIP users on the Internet -;pedantic=yes ; Enable checking of tags in headers, -; international character conversions in URIs -; and multiline formatted headers for strict -; SIP compatibility (defaults to "no") +;pedantic=yes ; Enable checking of tags in headers, + ; international character conversions in URIs + ; and multiline formatted headers for strict + ; SIP compatibility (defaults to "no") ; See qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters. ;tos_sip=cs3 ; Sets TOS for SIP packets. @@ -173,32 +173,32 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;cos_text=3 ; Sets 802.1p priority for RTP text packets. ;maxexpiry=3600 ; Maximum allowed time of incoming registrations -; and subscriptions (seconds) + ; and subscriptions (seconds) ;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60) ;defaultexpiry=120 ; Default length of incoming/outgoing registration ;mwiexpiry=3600 ; Expiry time for outgoing MWI subscriptions -;qualifyfreq=60 ; Qualification: How often to check for the -; host to be up in seconds -; Set to low value if you use low timeout for -; NAT of UDP sessions +;qualifyfreq=60 ; Qualification: How often to check for the + ; host to be up in seconds + ; Set to low value if you use low timeout for + ; NAT of UDP sessions ;qualifygap=100 ; Number of milliseconds between each group of peers being qualified ;qualifypeers=1 ; Number of peers in a group to be qualified at the same time ;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY ;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC -; fully. Enable this option to not get error messages -; when sending MWI to phones with this bug. + ; fully. Enable this option to not get error messages + ; when sending MWI to phones with this bug. ;mwi_from=asterisk ; When sending MWI NOTIFY requests, use this setting in -; the From: header as the "name" portion. Also fill the -; "user" portion of the URI in the From: header with this -; value if no fromuser is set -; Default: empty -;vmexten=voicemail ; dialplan extension to reach mailbox sets the -; Message-Account in the MWI notify message -; defaults to "asterisk" + ; the From: header as the "name" portion. Also fill the + ; "user" portion of the URI in the From: header with this + ; value if no fromuser is set + ; Default: empty +;vmexten=voicemail ; dialplan extension to reach mailbox sets the + ; Message-Account in the MWI notify message + ; defaults to "asterisk" ;preferred_codec_only=yes ; Respond to a SIP invite with the single most preferred codec -; rather than advertising all joint codec capabilities. This -; limits the other side's codec choice to exactly what we prefer. + ; rather than advertising all joint codec capabilities. This + ; limits the other side's codec choice to exactly what we prefer. ;disallow=all ; First disallow all codecs ;allow=ulaw ; Allow codecs in order of preference @@ -220,135 +220,135 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;mohsuggest=default ; ;parkinglot=plaza ; Sets the default parking lot for call parking -; This may also be set for individual users/peers -; Parkinglots are configured in features.conf + ; This may also be set for individual users/peers + ; Parkinglots are configured in features.conf ;language=en ; Default language setting for all users/peers -; This may also be set for individual users/peers + ; This may also be set for individual users/peers ;relaxdtmf=yes ; Relax dtmf handling ;trustrpid = no ; If Remote-Party-ID should be trusted ;sendrpid = yes ; If Remote-Party-ID should be sent ;sendrpid = rpid ; Use the "Remote-Party-ID" header -; to send the identity of the remote party -; This is identical to sendrpid=yes + ; to send the identity of the remote party + ; This is identical to sendrpid=yes ;sendrpid = pai ; Use the "P-Asserted-Identity" header -; to send the identity of the remote party + ; to send the identity of the remote party ;rpid_update = no ; In certain cases, the only method by which a connected line -; change may be immediately transmitted is with a SIP UPDATE request. -; If communicating with another Asterisk server, and you wish to be able -; transmit such UPDATE messages to it, then you must enable this option. -; Otherwise, we will have to wait until we can send a reinvite to -; transmit the information. + ; change may be immediately transmitted is with a SIP UPDATE request. + ; If communicating with another Asterisk server, and you wish to be able + ; transmit such UPDATE messages to it, then you must enable this option. + ; Otherwise, we will have to wait until we can send a reinvite to + ; transmit the information. ;progressinband=never ; If we should generate in-band ringing always -; use 'never' to never use in-band signalling, even in cases -; where some buggy devices might not render it -; Valid values: yes, no, never Default: never + ; use 'never' to never use in-band signalling, even in cases + ; where some buggy devices might not render it + ; Valid values: yes, no, never Default: never ;useragent=Asterisk PBX ; Allows you to change the user agent string -; The default user agent string also contains the Asterisk -; version. If you don't want to expose this, change the -; useragent string. + ; The default user agent string also contains the Asterisk + ; version. If you don't want to expose this, change the + ; useragent string. ;sdpsession=Asterisk PBX ; Allows you to change the SDP session name string, (s=) -; Like the useragent parameter, the default user agent string -; also contains the Asterisk version. + ; Like the useragent parameter, the default user agent string + ; also contains the Asterisk version. ;sdpowner=root ; Allows you to change the username field in the SDP owner string, (o=) -; This field MUST NOT contain spaces + ; This field MUST NOT contain spaces ;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address -; Note that promiscredir when redirects are made to the -; local system will cause loops since Asterisk is incapable -; of performing a "hairpin" call. + ; Note that promiscredir when redirects are made to the + ; local system will cause loops since Asterisk is incapable + ; of performing a "hairpin" call. ;usereqphone = no ; If yes, ";user=phone" is added to uri that contains -; a valid phone number + ; a valid phone number ;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833 -; Other options: -; info : SIP INFO messages (application/dtmf-relay) -; shortinfo : SIP INFO messages (application/dtmf) -; inband : Inband audio (requires 64 kbit codec -alaw, ulaw) -; auto : Use rfc2833 if offered, inband otherwise + ; Other options: + ; info : SIP INFO messages (application/dtmf-relay) + ; shortinfo : SIP INFO messages (application/dtmf) + ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw) + ; auto : Use rfc2833 if offered, inband otherwise ;compactheaders = yes ; send compact sip headers. ; ;videosupport=yes ; Turn on support for SIP video. You need to turn this -; on in this section to get any video support at all. -; You can turn it off on a per peer basis if the general -; video support is enabled, but you can't enable it for -; one peer only without enabling in the general section. -; If you set videosupport to "always", then RTP ports will -; always be set up for video, even on clients that don't -; support it. This assists callfile-derived calls and -; certain transferred calls to use always use video when -; available. [yes|NO|always] + ; on in this section to get any video support at all. + ; You can turn it off on a per peer basis if the general + ; video support is enabled, but you can't enable it for + ; one peer only without enabling in the general section. + ; If you set videosupport to "always", then RTP ports will + ; always be set up for video, even on clients that don't + ; support it. This assists callfile-derived calls and + ; certain transferred calls to use always use video when + ; available. [yes|NO|always] ;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s) -; Videosupport and maxcallbitrate is settable -; for peers and users as well -;callevents=no ; generate manager events when sip ua -; performs events (e.g. hold) + ; Videosupport and maxcallbitrate is settable + ; for peers and users as well +;callevents=no ; generate manager events when sip ua + ; performs events (e.g. hold) ;authfailureevents=no ; generate manager "peerstatus" events when peer can't -; authenticate with Asterisk. Peerstatus will be "rejected". + ; authenticate with Asterisk. Peerstatus will be "rejected". ;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected, -; for any reason, always reject with an identical response -; equivalent to valid username and invalid password/hash -; instead of letting the requester know whether there was -; a matching user or peer for their request. This reduces -; the ability of an attacker to scan for valid SIP usernames. + ; for any reason, always reject with an identical response + ; equivalent to valid username and invalid password/hash + ; instead of letting the requester know whether there was + ; a matching user or peer for their request. This reduces + ; the ability of an attacker to scan for valid SIP usernames. ;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing -; order instead of RFC3551 packing order (this is required -; for Sipura and Grandstream ATAs, among others). This is -; contrary to the RFC3551 specification, the peer _should_ -; be negotiating AAL2-G726-32 instead :-( + ; order instead of RFC3551 packing order (this is required + ; for Sipura and Grandstream ATAs, among others). This is + ; contrary to the RFC3551 specification, the peer _should_ + ; be negotiating AAL2-G726-32 instead :-( ;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the devices ;outboundproxy=proxy.provider.domain:8080 ; send outbound signaling to this proxy, not directly to the devices ;outboundproxy=proxy.provider.domain,force ; Send ALL outbound signalling to proxy, ignoring route: headers -;outboundproxy=tls://proxy.provider.domain ; same as '=proxy.provider.domain' except we try to connect with tls +;outboundproxy=tls://proxy.provider.domain ; same as '=proxy.provider.domain' except we try to connect with tls ; ; (could also be tcp,udp) - defining transports on the proxy line only ; ; applies for the global proxy, otherwise use the transport= option ;matchexterniplocally = yes ; Only substitute the externip or externhost setting if it matches -; your localnet setting. Unless you have some sort of strange network -; setup you will not need to enable this. + ; your localnet setting. Unless you have some sort of strange network + ; setup you will not need to enable this. ;dynamic_exclude_static = yes ; Disallow all dynamic hosts from registering -; as any IP address used for staticly defined -; hosts. This helps avoid the configuration -; error of allowing your users to register at -; the same address as a SIP provider. + ; as any IP address used for staticly defined + ; hosts. This helps avoid the configuration + ; error of allowing your users to register at + ; the same address as a SIP provider. ;contactdeny=0.0.0.0/0.0.0.0 ; Use contactpermit and contactdeny to ;contactpermit=172.16.0.0/255.255.0.0 ; restrict at what IPs your users may -; register their phones. + ; register their phones. ;engine=asterisk ; RTP engine to use when communicating with the device ; ; If regcontext is specified, Asterisk will dynamically create and destroy a ; NoOp priority 1 extension for a given peer who registers or unregisters with -; us and have a "regexten=" configuration item. -; Multiple contexts may be specified by separating them with '&'. The +; us and have a "regexten=" configuration item. +; Multiple contexts may be specified by separating them with '&'. The ; actual extension is the 'regexten' parameter of the registering peer or its ; name if 'regexten' is not provided. If more than one context is provided, ; the context must be specified within regexten by appending the desired -; context after '@'. More than one regexten may be supplied if they are +; context after '@'. More than one regexten may be supplied if they are ; separated by '&'. Patterns may be used in regexten. ; ;regcontext=sipregistrations ;regextenonqualify=yes ; Default "no" -; If you have qualify on and the peer becomes unreachable -; this setting will enforce inactivation of the regexten -; extension for the peer + ; If you have qualify on and the peer becomes unreachable + ; this setting will enforce inactivation of the regexten + ; extension for the peer ; ;--------------------------- SIP timers ---------------------------------------------------- -; These timers are used primarily in INVITE transactions. +; These timers are used primarily in INVITE transactions. ; The default for Timer T1 is 500 ms or the measured run-trip time between ; Asterisk and the device if you have qualify=yes for the device. ; ;t1min=100 ; Minimum roundtrip time for messages to monitored hosts -; Defaults to 100 ms + ; Defaults to 100 ms ;timert1=500 ; Default T1 timer -; Defaults to 500 ms or the measured round-trip -; time to a peer (qualify=yes). + ; Defaults to 500 ms or the measured round-trip + ; time to a peer (qualify=yes). ;timerb=32000 ; Call setup timer. If a provisional response is not received -; in this amount of time, the call will autocongest -; Defaults to 64*timert1 + ; in this amount of time, the call will autocongest + ; Defaults to 64*timert1 ;--------------------------- RTP timers ---------------------------------------------------- ; These timers are currently used for both audio and video streams. The RTP timeouts @@ -356,15 +356,15 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; The settings are settable in the global section as well as per device ; ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity -; on the audio channel -; when we're not on hold. This is to be able to hangup -; a call in the case of a phone disappearing from the net, -; like a powerloss or grandma tripping over a cable. + ; on the audio channel + ; when we're not on hold. This is to be able to hangup + ; a call in the case of a phone disappearing from the net, + ; like a powerloss or grandma tripping over a cable. ;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity -; on the audio channel -; when we're on hold (must be > rtptimeout) + ; on the audio channel + ; when we're on hold (must be > rtptimeout) ;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open -; (default is off - zero) + ; (default is off - zero) ;--------------------------- SIP Session-Timers (RFC 4028)------------------------------------ ; SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions. @@ -403,22 +403,22 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;--------------------------- SIP DEBUGGING --------------------------------------------------- ;sipdebug = yes ; Turn on SIP debugging by default, from -; the moment the channel loads this configuration -;recordhistory=yes ; Record SIP history by default -; (see sip history / sip no history) + ; the moment the channel loads this configuration +;recordhistory=yes ; Record SIP history by default + ; (see sip history / sip no history) ;dumphistory=yes ; Dump SIP history at end of SIP dialogue -; SIP history is output to the DEBUG logging channel + ; SIP history is output to the DEBUG logging channel ;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ---------------------------- ; You can subscribe to the status of extensions with a "hint" priority ; (See extensions.conf.sample for examples) -; chan_sip support two major formats for notifications: dialog-info and SIMPLE +; chan_sip support two major formats for notifications: dialog-info and SIMPLE ; ; You will get more detailed reports (busy etc) if you have a call counter enabled -; for a device. +; for a device. ; -; If you set the busylevel, we will indicate busy when we have a number of calls that +; If you set the busylevel, we will indicate busy when we have a number of calls that ; matches the busylevel treshold. ; ; For queues, you will need this level of detail in status reporting, regardless @@ -430,38 +430,38 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; ;allowsubscribe=no ; Disable support for subscriptions. (Default is yes) ;subscribecontext = default ; Set a specific context for SUBSCRIBE requests -; Useful to limit subscriptions to local extensions -; Settable per peer/user also + ; Useful to limit subscriptions to local extensions + ; Settable per peer/user also ;notifyringing = no ; Control whether subscriptions already INUSE get sent -; RINGING when another call is sent (default: yes) + ; RINGING when another call is sent (default: yes) ;notifyhold = yes ; Notify subscriptions on HOLD state (default: no) -; Turning on notifyringing and notifyhold will add a lot -; more database transactions if you are using realtime. + ; Turning on notifyringing and notifyhold will add a lot + ; more database transactions if you are using realtime. ;notifycid = yes ; Control whether caller ID information is sent along with -; dialog-info+xml notifications (supported by snom phones). -; Note that this feature will only work properly when the -; incoming call is using the same extension and context that -; is being used as the hint for the called extension. This means -; that it won't work when using subscribecontext for your sip -; user or peer (if subscribecontext is different than context). -; This is also limited to a single caller, meaning that if an -; extension is ringing because multiple calls are incoming, -; only one will be used as the source of caller ID. Specify -; 'ignore-context' to ignore the called context when looking -; for the caller's channel. The default value is 'no.' Setting -; notifycid to 'ignore-context' also causes call-pickups attempted -; via SNOM's NOTIFY mechanism to set the context for the call pickup -; to PICKUPMARK. + ; dialog-info+xml notifications (supported by snom phones). + ; Note that this feature will only work properly when the + ; incoming call is using the same extension and context that + ; is being used as the hint for the called extension. This means + ; that it won't work when using subscribecontext for your sip + ; user or peer (if subscribecontext is different than context). + ; This is also limited to a single caller, meaning that if an + ; extension is ringing because multiple calls are incoming, + ; only one will be used as the source of caller ID. Specify + ; 'ignore-context' to ignore the called context when looking + ; for the caller's channel. The default value is 'no.' Setting + ; notifycid to 'ignore-context' also causes call-pickups attempted + ; via SNOM's NOTIFY mechanism to set the context for the call pickup + ; to PICKUPMARK. ;callcounter = yes ; Enable call counters on devices. This can be set per -; device too. + ; device too. ;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT ----------------------- ; ; This setting is available in the [general] section as well as in device configurations. ; Setting this to yes, enables T.38 fax (UDPTL) passthrough on SIP to SIP calls, provided -; both parties have T38 support enabled in their Asterisk configuration +; both parties have T38 support enabled in their Asterisk configuration ; This has to be enabled in the general section for all devices to work. You can then -; disable it on a per device basis. +; disable it on a per device basis. ; ; T.38 faxing only works in SIP to SIP calls, with no local or agent channel being used. ; @@ -469,21 +469,21 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; ; Fax Detect will cause the SIP channel to jump to the 'fax' extension (if it exists) ; after T.38 is successfully negotiated. -; -; faxdetect = yes ; Default false +; +; faxdetect = yes ; Default false ; ;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------ ; Asterisk can register as a SIP user agent to a SIP proxy (provider) ; Format for the register statement is: ; register => [transport://]user[:secret[:authuser]]@domain[:port][/extension][~expiry] ; -; ; -; domain is either +; +; domain is either ; - domain in DNS ; - host name in DNS ; - the name of a peer defined below or in realtime -; The domain is where you register your username, so your SIP uri you are registering to +; The domain is where you register your username, so your SIP uri you are registering to ; is username@domain ; ; If no extension is given, the 's' extension is used. The extension needs to @@ -514,7 +514,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; ; Examples: ; -;register => 1234:password@mysipprovider.com +;register => 1234:password@mysipprovider.com ; ; This will pass incoming calls to the 's' extension ; @@ -536,9 +536,9 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;registertimeout=20 ; retry registration calls every 20 seconds (default) ;registerattempts=10 ; Number of registration attempts before we give up -; 0 = continue forever, hammering the other server -; until it accepts the registration -; Default is 0 tries, continue forever + ; 0 = continue forever, hammering the other server + ; until it accepts the registration + ; Default is 0 tries, continue forever ;----------------------------------------- OUTBOUND MWI SUBSCRIPTIONS ------------------------- ; Asterisk can subscribe to receive the MWI from another SIP server and store it locally for retrieval ; by other phones. @@ -635,7 +635,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; nat = no ; default. Use NAT mode only according to RFC3581 (;rport) ; nat = yes ; Always ignore info and assume NAT ; nat = never ; Never attempt NAT mode or RFC3581 support -; nat = route ; route = Assume NAT, don't send rport +; nat = route ; route = Assume NAT, don't send rport ; ; (work around more UNIDEN bugs) ;----------------------------------- MEDIA HANDLING -------------------------------- @@ -645,43 +645,43 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat ; ;canreinvite=yes ; Asterisk by default tries to redirect the -; RTP media stream (audio) to go directly from -; the caller to the callee. Some devices do not -; support this (especially if one of them is behind a NAT). -; The default setting is YES. If you have all clients -; behind a NAT, or for some other reason wants Asterisk to -; stay in the audio path, you may want to turn this off. - -; This setting also affect direct RTP -; at call setup (a new feature in 1.4 - setting up the -; call directly between the endpoints instead of sending -; a re-INVITE). + ; RTP media stream (audio) to go directly from + ; the caller to the callee. Some devices do not + ; support this (especially if one of them is behind a NAT). + ; The default setting is YES. If you have all clients + ; behind a NAT, or for some other reason wants Asterisk to + ; stay in the audio path, you may want to turn this off. + + ; This setting also affect direct RTP + ; at call setup (a new feature in 1.4 - setting up the + ; call directly between the endpoints instead of sending + ; a re-INVITE). ;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up -; the call directly with media peer-2-peer without re-invites. -; Will not work for video and cases where the callee sends -; RTP payloads and fmtp headers in the 200 OK that does not match the -; callers INVITE. This will also fail if canreinvite is enabled when -; the device is actually behind NAT. + ; the call directly with media peer-2-peer without re-invites. + ; Will not work for video and cases where the callee sends + ; RTP payloads and fmtp headers in the 200 OK that does not match the + ; callers INVITE. This will also fail if canreinvite is enabled when + ; the device is actually behind NAT. ;canreinvite=nonat ; An additional option is to allow media path redirection -; (reinvite) but only when the peer where the media is being -; sent is known to not be behind a NAT (as the RTP core can -; determine it based on the apparent IP address the media -; arrives from). + ; (reinvite) but only when the peer where the media is being + ; sent is known to not be behind a NAT (as the RTP core can + ; determine it based on the apparent IP address the media + ; arrives from). ;canreinvite=update ; Yet a third option... use UPDATE for media path redirection, -; instead of INVITE. This can be combined with 'nonat', as -; 'canreinvite=update,nonat'. It implies 'yes'. + ; instead of INVITE. This can be combined with 'nonat', as + ; 'canreinvite=update,nonat'. It implies 'yes'. ;ignoresdpversion=yes ; By default, Asterisk will honor the session version -; number in SDP packets and will only modify the SDP -; session if the version number changes. This option will -; force asterisk to ignore the SDP session version number -; and treat all SDP data as new data. This is required -; for devices that send us non standard SDP packets -; (observed with Microsoft OCS). By default this option is -; off. + ; number in SDP packets and will only modify the SDP + ; session if the version number changes. This option will + ; force asterisk to ignore the SDP session version number + ; and treat all SDP data as new data. This is required + ; for devices that send us non standard SDP packets + ; (observed with Microsoft OCS). By default this option is + ; off. ;----------------------------------------- REALTIME SUPPORT ------------------------ ; For additional information on ARA, the Asterisk Realtime Architecture, @@ -689,38 +689,38 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; source code. ; ;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list -; just like friends added from the config file only on a -; as-needed basis? (yes|no) + ; just like friends added from the config file only on a + ; as-needed basis? (yes|no) ;rtsavesysname=yes ; Save systemname in realtime database at registration -; Default= no + ; Default= no ;rtupdate=yes ; Send registry updates to database using realtime? (yes|no) -; If set to yes, when a SIP UA registers successfully, the ip address, -; the origination port, the registration period, and the username of -; the UA will be set to database via realtime. -; If not present, defaults to 'yes'. Note: realtime peers will -; probably not function across reloads in the way that you expect, if -; you turn this option off. + ; If set to yes, when a SIP UA registers successfully, the ip address, + ; the origination port, the registration period, and the username of + ; the UA will be set to database via realtime. + ; If not present, defaults to 'yes'. Note: realtime peers will + ; probably not function across reloads in the way that you expect, if + ; you turn this option off. ;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule -; as if it had just registered? (yes|no|<seconds>) -; If set to yes, when the registration expires, the friend will -; vanish from the configuration until requested again. If set -; to an integer, friends expire within this number of seconds -; instead of the registration interval. + ; as if it had just registered? (yes|no|<seconds>) + ; If set to yes, when the registration expires, the friend will + ; vanish from the configuration until requested again. If set + ; to an integer, friends expire within this number of seconds + ; instead of the registration interval. ;ignoreregexpire=yes ; Enabling this setting has two functions: -; -; For non-realtime peers, when their registration expires, the -; information will _not_ be removed from memory or the Asterisk database -; if you attempt to place a call to the peer, the existing information -; will be used in spite of it having expired -; -; For realtime peers, when the peer is retrieved from realtime storage, -; the registration information will be used regardless of whether -; it has expired or not; if it expires while the realtime peer -; is still in memory (due to caching or other reasons), the -; information will not be removed from realtime storage + ; + ; For non-realtime peers, when their registration expires, the + ; information will _not_ be removed from memory or the Asterisk database + ; if you attempt to place a call to the peer, the existing information + ; will be used in spite of it having expired + ; + ; For realtime peers, when the peer is retrieved from realtime storage, + ; the registration information will be used regardless of whether + ; it has expired or not; if it expires while the realtime peer + ; is still in memory (due to caching or other reasons), the + ; information will not be removed from realtime storage ;----------------------------------------- SIP DOMAIN SUPPORT ------------------------ ; Incoming INVITE and REFER messages can be matched against a list of 'allowed' @@ -744,45 +744,45 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; allowexternaldomains=no ;domain=mydomain.tld,mydomain-incoming -; Add domain and configure incoming context -; for external calls to this domain + ; Add domain and configure incoming context + ; for external calls to this domain ;domain=1.2.3.4 ; Add IP address as local domain -; You can have several "domain" settings + ; You can have several "domain" settings ;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains -; Default is yes + ; Default is yes ;autodomain=yes ; Turn this on to have Asterisk add local host -; name and local IP to domain list. + ; name and local IP to domain list. ; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to -; non-peers, use your primary domain "identity" -; for From: headers instead of just your IP -; address. This is to be polite and -; it may be a mandatory requirement for some -; destinations which do not have a prior -; account relationship with your server. + ; non-peers, use your primary domain "identity" + ; for From: headers instead of just your IP + ; address. This is to be polite and + ; it may be a mandatory requirement for some + ; destinations which do not have a prior + ; account relationship with your server. ;------------------------------ JITTER BUFFER CONFIGURATION -------------------------- ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a -; SIP 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 SIP channel can accept jitter, -; thus a jitterbuffer on the receive SIP side will be used only -; if it is forced and enabled. + ; SIP 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 SIP channel can accept jitter, + ; thus a jitterbuffer on the receive SIP side will be used only + ; if it is forced and enabled. ; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP -; channel. Defaults to "no". + ; channel. Defaults to "no". ; 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. + ; 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 jbmaxsize) and "adaptive" (with -; variable size, actually the new jb of IAX2). Defaults to fixed. + ; channel. Two implementations are currently available - "fixed" + ; (with size always equals to jbmaxsize) and "adaptive" (with + ; variable size, actually the new jb of IAX2). Defaults to fixed. ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no". ;----------------------------------------------------------------------------------- @@ -793,20 +793,20 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; any credentials in peer/register definition if realm is matched. ; ; This way, Asterisk can authenticate for outbound calls to other -; realms. We match realm on the proxy challenge and pick an set of +; realms. We match realm on the proxy challenge and pick an set of ; credentials from this list ; Syntax: ; auth = <user>:<secret>@<realm> ; auth = <user>#<md5secret>@<realm> ; Example: ;auth=mark:topsecret@digium.com -; -; You may also add auth= statements to [peer] definitions +; +; You may also add auth= statements to [peer] definitions ; Peer auth= override all other authentication settings if we match on realm ;------------------------------------------------------------------------------ ; DEVICE CONFIGURATION -; +; ; The SIP channel has two types of devices, the friend and the peer. ; * The type=friend is a device type that accepts both incoming and outbound calls, ; where Asterisk match on the From: username on incoming calls. @@ -817,16 +817,16 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; trunks. ; ; For device names, we recommend using only a-z, numerics (0-9) and underscore -; +; ; For local phones, type=friend works most of the time ; -; If you have one-way audio, you probably have NAT problems. +; If you have one-way audio, you probably have NAT problems. ; If Asterisk is on a public IP, and the phone is inside of a NAT device ; you will need to configure nat option for those phones. ; Also, turn on qualify=yes to keep the nat session open -; -; Configuration options available -; -------------------- +; +; Configuration options available +; -------------------- ; context ; callingpres ; permit @@ -895,7 +895,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;[sip_proxy] ; For incoming calls only. Example: FWD (Free World Dialup) -; We match on IP address of the proxy for incoming calls +; We match on IP address of the proxy for incoming calls ; since we can not match on username (caller id) ;type=peer ;context=from-fwd @@ -906,7 +906,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;remotesecret=guessit ; Our password to their service ;defaultuser=yourusername ; Authentication user for outbound proxies ;fromuser=yourusername ; Many SIP providers require this! -;fromdomain=provider.sip.domain +;fromdomain=provider.sip.domain ;host=box.provider.com ;transport=udp,tcp ; This sets the default transport type to udp for outgoing, and will ; ; accept both tcp and udp. The default transport type is only used for @@ -919,7 +919,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;busylevel=2 ; Signal busy at 2 or more calls ;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer ;port=80 ; The port number we want to connect to on the remote side -; Also used as "defaultport" in combination with "defaultip" settings + ; Also used as "defaultport" in combination with "defaultip" settings ;--- sample definition for a provider ;[provider1] @@ -940,30 +940,30 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; the templates uncommented as they will not harm: [basic-options](!) ; a template -dtmfmode=rfc2833 -context=from-office -type=friend + dtmfmode=rfc2833 + context=from-office + type=friend [natted-phone](!,basic-options) ; another template inheriting basic-options -nat=yes -canreinvite=no -host=dynamic + nat=yes + canreinvite=no + host=dynamic [public-phone](!,basic-options) ; another template inheriting basic-options -nat=no -canreinvite=yes + nat=no + canreinvite=yes [my-codecs](!) ; a template for my preferred codecs -disallow=all -allow=ilbc -allow=g729 -allow=gsm -allow=g723 -allow=ulaw + disallow=all + allow=ilbc + allow=g729 + allow=gsm + allow=g723 + allow=ulaw [ulaw-phone](!) ; and another one for ulaw-only -disallow=all -allow=ulaw + disallow=all + allow=ulaw ; and finally instantiate a few phones ; @@ -979,34 +979,34 @@ allow=ulaw ; Standard configurations not using templates look like this: ; ;[grandstream1] -;type=friend +;type=friend ;context=from-sip ; Where to start in the dialplan when this phone calls ;callerid=John Doe <1234> ; Full caller ID, to override the phones config -; on incoming calls to Asterisk + ; on incoming calls to Asterisk ;host=192.168.0.23 ; we have a static but private IP address -; No registration allowed + ; No registration allowed ;nat=no ; there is not NAT between phone and Asterisk ;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk ;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone ;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time -; from the phone to asterisk (deprecated) -; 1 for the explicit peer, 1 for the explicit user, -; remember that a friend equals 1 peer and 1 user in -; memory -; There is no combined call counter for a "friend" -; so there's currently no way in sip.conf to limit -; to one inbound or outbound call per phone. Use -; the group counters in the dial plan for that. -; + ; from the phone to asterisk (deprecated) + ; 1 for the explicit peer, 1 for the explicit user, + ; remember that a friend equals 1 peer and 1 user in + ; memory + ; There is no combined call counter for a "friend" + ; so there's currently no way in sip.conf to limit + ; to one inbound or outbound call per phone. Use + ; the group counters in the dial plan for that. + ; ;mailbox=1234@default ; mailbox 1234 in voicemail context "default" ;disallow=all ; need to disallow=all before we can use allow= ;allow=ulaw ; Note: In user sections the order of codecs -; listed with allow= does NOT matter! + ; listed with allow= does NOT matter! ;allow=alaw ;allow=g723.1 ; Asterisk only supports g723.1 pass-thru! ;allow=g729 ; Pass-thru only unless g729 license obtained ;callingpres=allowed_passed_screen ; Set caller ID presentation -; See README.callingpres for more information + ; See README.callingpres for more information ;[xlite1] ; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)! @@ -1029,16 +1029,16 @@ allow=ulaw ;context=from-sip ; Context for incoming calls from this user ;secret=blah ;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions -;language=de ; Use German prompts for this user +;language=de ; Use German prompts for this user ;host=dynamic ; This peer register with us ;dtmfmode=inband ; Choices are inband, rfc2833, or info ;defaultip=192.168.0.59 ; IP used until peer registers ;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator -;subscribemwi=yes ; Only send notifications if this phone -; subscribes for mailbox notification -;vmexten=voicemail ; dialplan extension to reach mailbox -; sets the Message-Account in the MWI notify message -; defaults to global vmexten which defaults to "asterisk" +;subscribemwi=yes ; Only send notifications if this phone + ; subscribes for mailbox notification +;vmexten=voicemail ; dialplan extension to reach mailbox + ; sets the Message-Account in the MWI notify message + ; defaults to global vmexten which defaults to "asterisk" ;disallow=all ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw! @@ -1051,7 +1051,7 @@ allow=ulaw ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info ;defaultuser=polly ; Username to use in INVITE until peer registers ;defaultip=192.168.40.123 -; Normally you do NOT need to set this parameter + ; Normally you do NOT need to set this parameter ;disallow=all ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw! ;progressinband=no ; Polycom phones don't work properly with "never" @@ -1061,17 +1061,17 @@ allow=ulaw ;type=friend ;secret=blah ;host=dynamic -;insecure=port ; Allow matching of peer by IP address without -; matching port number +;insecure=port ; Allow matching of peer by IP address without + ; matching port number ;insecure=invite ; Do not require authentication of incoming INVITEs ;insecure=port,invite ; (both) ;qualify=1000 ; Consider it down if it's 1 second to reply -; Helps with NAT session -; qualify=yes uses default value -;qualifyfreq=60 ; Qualification: How often to check for the -; host to be up in seconds -; Set to low value if you use low timeout for -; NAT of UDP sessions + ; Helps with NAT session + ; qualify=yes uses default value +;qualifyfreq=60 ; Qualification: How often to check for the + ; host to be up in seconds + ; Set to low value if you use low timeout for + ; NAT of UDP sessions ; ; Call group and Pickup group should be in the range from 0 to 63 ; @@ -1086,30 +1086,30 @@ allow=ulaw ;secret=blah ;qualify=200 ; Qualify peer is no more than 200ms away ;nat=yes ; This phone may be natted -; Send SIP and RTP to the IP address that packet is -; received from instead of trusting SIP headers + ; Send SIP and RTP to the IP address that packet is + ; received from instead of trusting SIP headers ;host=dynamic ; This device registers with us ;canreinvite=no ; Asterisk by default tries to redirect the -; RTP media stream (audio) to go directly from -; the caller to the callee. Some devices do not -; support this (especially if one of them is -; behind a NAT). + ; RTP media stream (audio) to go directly from + ; the caller to the callee. Some devices do not + ; support this (especially if one of them is + ; behind a NAT). ;defaultip=192.168.0.4 ; IP address to use until registration ;defaultuser=goran ; Username to use when calling this device before registration -; Normally you do NOT need to set this parameter + ; Normally you do NOT need to set this parameter ;setvar=CUSTID=5678 ; Channel variable to be set for all calls from or to this device ;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will -; cause the given audio file to -; be played upon completion of -; an attended transfer. + ; cause the given audio file to + ; be played upon completion of + ; an attended transfer. ;[pre14-asterisk] ;type=friend ;secret=digium ;host=dynamic ;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine. -; You must have this turned on or DTMF reception will work improperly. + ; You must have this turned on or DTMF reception will work improperly. ;t38pt_usertpsource=yes ; Use the source IP address of RTP as the destination IP address for UDPTL packets -; if the nat option is enabled. If a single RTP packet is received Asterisk will know the -; external IP address of the remote device. If port forwarding is done at the client side -; then UDPTL will flow to the remote device. + ; if the nat option is enabled. If a single RTP packet is received Asterisk will know the + ; external IP address of the remote device. If port forwarding is done at the client side + ; then UDPTL will flow to the remote device. diff --git a/configs/skinny.conf.sample b/configs/skinny.conf.sample index f5613ac21..701723923 100644 --- a/configs/skinny.conf.sample +++ b/configs/skinny.conf.sample @@ -5,22 +5,22 @@ bindaddr=0.0.0.0 ; Address to bind to bindport=2000 ; Port to bind to, default tcp/2000 dateformat=M-D-Y ; M,D,Y in any order (6 chars max) -; "A" may also be used, but it must be at the end. -; Use M for month, D for day, Y for year, A for 12-hour time. + ; "A" may also be used, but it must be at the end. + ; Use M for month, D for day, Y for year, A for 12-hour time. keepalive=120 ;vmexten=8500 ; Systemwide voicemailmain pilot number -; It must be in the same context as the calling -; device/line + ; It must be in the same context as the calling + ; device/line ; If regcontext is specified, Asterisk will dynamically create and destroy a ; NoOp priority 1 extension for a given line which registers or unregisters with -; us and have a "regexten=" configuration item. -; Multiple contexts may be specified by separating them with '&'. The +; us and have a "regexten=" configuration item. +; Multiple contexts may be specified by separating them with '&'. The ; actual extension is the 'regexten' parameter of the registering line or its ; name if 'regexten' is not provided. If more than one context is provided, ; the context must be specified within regexten by appending the desired -; context after '@'. More than one regexten may be supplied if they are +; context after '@'. More than one regexten may be supplied if they are ; separated by '&'. Patterns may be used in regexten. ; ;regcontext=skinnyregistrations @@ -38,27 +38,27 @@ keepalive=120 ;------------------------------ JITTER BUFFER CONFIGURATION -------------------------- ;jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a -; skinny 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 skinny channel can accept -; jitter, thus a jitterbuffer on the receive skinny side will be -; used only if it is forced and enabled. + ; skinny 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 skinny channel can accept + ; jitter, thus a jitterbuffer on the receive skinny side will be + ; used only if it is forced and enabled. ;jbforce = no ; Forces the use of a jitterbuffer on the receive side of a skinny -; channel. Defaults to "no". + ; channel. Defaults to "no". ;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. + ; 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 -; skinny channel. Two implementations are currently available -; - "fixed" (with size always equals to jbmaxsize) -; - "adaptive" (with variable size, actually the new jb of IAX2). -; Defaults to fixed. + ; skinny channel. Two implementations are currently available + ; - "fixed" (with size always equals to jbmaxsize) + ; - "adaptive" (with variable size, actually the new jb of IAX2). + ; Defaults to fixed. ;jblog = no ; Enables jitterbuffer frame logging. Defaults to "no". ;----------------------------------------------------------------------------------- @@ -93,8 +93,8 @@ keepalive=120 ;vmexten=8500 ; Device level voicemailmain pilot number ;regexten=100 ;context=inbound -;linelabel="Support Line" ; Displays next to the line -; button on 7940's and 7960s +;linelabel="Support Line" ; Displays next to the line + ; button on 7940's and 7960s ;[110] ;callerid="John Chambers" <408-526-4000> ;context=did @@ -110,21 +110,21 @@ keepalive=120 ;callerid="George W. Bush" <202-456-1414> ;setvar=CUSTID=5678 ; Channel variable to be set for all calls from this device ;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will -; cause the given audio file to -; be played upon completion of -; an attended transfer. + ; cause the given audio file to + ; be played upon completion of + ; an attended transfer. ;mailbox=500 ;callwaiting=yes ;transfer=yes ;threewaycalling=yes ;context=default ;mohinterpret=default ; This option specifies a default music on hold class to -; use when put on hold if the channel's moh class was not -; explicitly set with Set(CHANNEL(musicclass)=whatever) and -; the peer channel did not suggest a class to use. + ; use when put on hold if the channel's moh class was not + ; explicitly set with Set(CHANNEL(musicclass)=whatever) and + ; the peer channel did not suggest a class to use. ;mohsuggest=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-user or per-peer basis. + ; when this channel places the peer on hold. It may be specified globally or on + ; a per-user or per-peer basis. [devices] diff --git a/configs/sla.conf.sample b/configs/sla.conf.sample index 9fdb3f336..c2015b622 100644 --- a/configs/sla.conf.sample +++ b/configs/sla.conf.sample @@ -8,10 +8,10 @@ [general] ;attemptcallerid=no ; Attempt CallerID handling. The default value for this -; is "no" because CallerID handling with an SLA setup is -; known to not work properly in some situations. However, -; feel free to enable it if you would like. If you do, and -; you find problems, please do not report them. + ; is "no" because CallerID handling with an SLA setup is + ; known to not work properly in some situations. However, + ; feel free to enable it if you would like. If you do, and + ; you find problems, please do not report them. ; ------------------------------------- @@ -22,30 +22,30 @@ ;type=trunk ; This line is what marks this entry as a trunk. ;device=DAHDI/3 ; Map this trunk declaration to a specific device. -; NOTE: You can not just put any type of channel here. -; DAHDI channels can be directly used. IP trunks -; require some indirect configuration which is -; described in doc/asterisk.pdf. + ; NOTE: You can not just put any type of channel here. + ; DAHDI channels can be directly used. IP trunks + ; require some indirect configuration which is + ; described in doc/asterisk.pdf. -;autocontext=line1 ; This supports automatic generation of the dialplan entries -; if the autocontext option is used. Each trunk should have -; a unique context name. Then, in chan_dahdi.conf, this device -; should be configured to have incoming calls go to this context. +;autocontext=line1 ; This supports automatic generation of the dialplan entries + ; if the autocontext option is used. Each trunk should have + ; a unique context name. Then, in chan_dahdi.conf, this device + ; should be configured to have incoming calls go to this context. -;ringtimeout=30 ; Set how long to allow this trunk to ring on an inbound call before hanging -; it up as an unanswered call. The value is in seconds. +;ringtimeout=30 ; Set how long to allow this trunk to ring on an inbound call before hanging + ; it up as an unanswered call. The value is in seconds. ;barge=no ; If this option is set to "no", then no station will be -; allowed to join a call that is in progress. The default -; value is "yes". + ; allowed to join a call that is in progress. The default + ; value is "yes". ;hold=private ; This option configure hold permissions for this trunk. -; "open" - This means that any station can put this trunk -; on hold, and any station can retrieve it from -; hold. This is the default. -; "private" - This means that once a station puts the -; trunk on hold, no other station will be -; allowed to retrieve the call from hold. + ; "open" - This means that any station can put this trunk + ; on hold, and any station can retrieve it from + ; hold. This is the default. + ; "private" - This means that once a station puts the + ; trunk on hold, no other station will be + ; allowed to retrieve the call from hold. ;[line2] ;type=trunk @@ -60,9 +60,9 @@ ;[line4] ;type=trunk ;device=Local/disa@line4_outbound ; A Local channel in combination with the Disa -; application can be used to support IP trunks. -; See doc/asterisk.pdf on more information on how -; IP trunks work. + ; application can be used to support IP trunks. + ; See doc/asterisk.pdf on more information on how + ; IP trunks work. ;autocontext=line4 ; -------------------------------------- @@ -75,55 +75,55 @@ ;device=SIP/station1 ; Each station must be mapped to a device. -;autocontext=sla_stations ; This supports automatic generation of the dialplan entries if -; the autocontext option is used. All stations can use the same -; context without conflict. The device for this station should -; have its context configured to the same one listed here. +;autocontext=sla_stations ; This supports automatic generation of the dialplan entries if + ; the autocontext option is used. All stations can use the same + ; context without conflict. The device for this station should + ; have its context configured to the same one listed here. -;ringtimeout=10 ; Set a timeout for how long to allow the station to ring for an -; incoming call, in seconds. +;ringtimeout=10 ; Set a timeout for how long to allow the station to ring for an + ; incoming call, in seconds. ;ringdelay=10 ; Set a time for how long to wait before beginning to ring this station -; once there is an incoming call, in seconds. + ; once there is an incoming call, in seconds. ;hold=private ; This option configure hold permissions for this station. Note -; that if private hold is set in the trunk entry, that will override -; anything here. However, if a trunk has open hold access, but this -; station is set to private hold, then the private hold will be in -; effect. -; "open" - This means that once this station puts a call -; on hold, any other station is allowed to retrieve -; it. This is the default. -; "private" - This means that once this station puts a -; call on hold, no other station will be -; allowed to retrieve the call from hold. + ; that if private hold is set in the trunk entry, that will override + ; anything here. However, if a trunk has open hold access, but this + ; station is set to private hold, then the private hold will be in + ; effect. + ; "open" - This means that once this station puts a call + ; on hold, any other station is allowed to retrieve + ; it. This is the default. + ; "private" - This means that once this station puts a + ; call on hold, no other station will be + ; allowed to retrieve the call from hold. ;trunk=line1 ; Individually list all of the trunks that will appear on this station. This -; order is significant. It should be the same order as they appear on the -; phone. The order here defines the order of preference that the trunks will -; be used. + ; order is significant. It should be the same order as they appear on the + ; phone. The order here defines the order of preference that the trunks will + ; be used. ;trunk=line2 ;trunk=line3,ringdelay=5 ; A ring delay for the station can also be specified for a specific trunk. -; If a ring delay is specified both for the whole station and for a specific -; trunk on a station, the setting for the specific trunk will take priority. -; This value is in seconds. + ; If a ring delay is specified both for the whole station and for a specific + ; trunk on a station, the setting for the specific trunk will take priority. + ; This value is in seconds. ;trunk=line4,ringtimeout=5 ; A ring timeout for the station can also be specified for a specific trunk. -; If a ring timeout is specified both for the whole station and for a specific -; trunk on a station, the setting for the specific trunk will take priority. -; This value is in seconds. + ; If a ring timeout is specified both for the whole station and for a specific + ; trunk on a station, the setting for the specific trunk will take priority. + ; This value is in seconds. ;[station](!) ; When there are a lot of stations that are configured the same way, -; it is convenient to use a configuration template like this so that -; the common settings stay in one place. + ; it is convenient to use a configuration template like this so that + ; the common settings stay in one place. ;type=station ;autocontext=sla_stations ;trunk=line1 -;trunk=line2 +;trunk=line2 ;trunk=line3 -;trunk=line4 +;trunk=line4 ;[station2](station) ; Define a station that uses the configuration from the template "station". ;device=SIP/station2 diff --git a/configs/telcordia-1.adsi b/configs/telcordia-1.adsi index 96eb1db21..1486aa95e 100644 --- a/configs/telcordia-1.adsi +++ b/configs/telcordia-1.adsi @@ -28,15 +28,15 @@ STATE "inactive" ; No active call ; Begin soft key definitions ; KEY "CB_OH" IS "Block" OR "Call Block" -OFFHOOK -VOICEMODE -WAITDIALTONE -SENDDTMF "*60" -SUBSCRIPT "offHook" + OFFHOOK + VOICEMODE + WAITDIALTONE + SENDDTMF "*60" + SUBSCRIPT "offHook" ENDKEY KEY "CB" IS "Block" OR "Call Block" -SENDDTMF "*60" + SENDDTMF "*60" ENDKEY ; @@ -44,38 +44,38 @@ ENDKEY ; SUB "main" IS -IFEVENT NEARANSWER THEN -CLEAR -SHOWDISPLAY "talkingto" AT 1 -GOTO "stableCall" -ENDIF -IFEVENT OFFHOOK THEN -CLEAR -SHOWDISPLAY "titles" AT 1 -SHOWKEYS "CB" -GOTO "offHook" -ENDIF -IFEVENT IDLE THEN -CLEAR -SHOWDISPLAY "titles" AT 1 -SHOWKEYS "CB_OH" -ENDIF -IFEVENT CALLERID THEN -CLEAR -SHOWDISPLAY "newcall" AT 1 -ENDIF + IFEVENT NEARANSWER THEN + CLEAR + SHOWDISPLAY "talkingto" AT 1 + GOTO "stableCall" + ENDIF + IFEVENT OFFHOOK THEN + CLEAR + SHOWDISPLAY "titles" AT 1 + SHOWKEYS "CB" + GOTO "offHook" + ENDIF + IFEVENT IDLE THEN + CLEAR + SHOWDISPLAY "titles" AT 1 + SHOWKEYS "CB_OH" + ENDIF + IFEVENT CALLERID THEN + CLEAR + SHOWDISPLAY "newcall" AT 1 + ENDIF ENDSUB SUB "offHook" IS -IFEVENT FARRING THEN -CLEAR -SHOWDISPLAY "ringing" AT 1 -ENDIF -IFEVENT FARANSWER THEN -CLEAR -SHOWDISPLAY "talkingto" AT 1 -GOTO "stableCall" -ENDIF + IFEVENT FARRING THEN + CLEAR + SHOWDISPLAY "ringing" AT 1 + ENDIF + IFEVENT FARANSWER THEN + CLEAR + SHOWDISPLAY "talkingto" AT 1 + GOTO "stableCall" + ENDIF ENDSUB SUB "stableCall" IS diff --git a/configs/unistim.conf.sample b/configs/unistim.conf.sample index 2b61a8646..39cb99875 100644 --- a/configs/unistim.conf.sample +++ b/configs/unistim.conf.sample @@ -14,29 +14,29 @@ port=5000 ; UDP port ;keepalive=120 ; in seconds, default = 120 ;public_ip= ; if asterisk is behind a nat, specify your public IP ;autoprovisioning=no ; Allow undeclared phones to register an extension. See README for important -; informations. no (default), yes, tn. + ; informations. no (default), yes, tn. ;------------------------------ JITTER BUFFER CONFIGURATION -------------------------- ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a -; SIP 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 SIP channel can accept jitter, -; thus a jitterbuffer on the receive SIP side will be used only -; if it is forced and enabled. + ; SIP 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 SIP channel can accept jitter, + ; thus a jitterbuffer on the receive SIP side will be used only + ; if it is forced and enabled. ; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP -; channel. Defaults to "no". + ; channel. Defaults to "no". ; 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. + ; 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 jbmaxsize) and "adaptive" (with -; variable size, actually the new jb of IAX2). Defaults to fixed. + ; channel. Two implementations are currently available - "fixed" + ; (with size always equals to jbmaxsize) and "adaptive" (with + ; variable size, actually the new jb of IAX2). Defaults to fixed. ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no". ;----------------------------------------------------------------------------------- @@ -63,13 +63,13 @@ port=5000 ; UDP port ;mailbox=1234 ; Specify the mailbox number. Used by Message Waiting Indication ;linelabel="Support" ; Softkey label for the next line=> entry, 9 char max. ;extension=none ; Add an extension into the dialplan. Only valid in context specified previously. -; none=don't add (default), ask=prompt user, line=use the line number + ; none=don't add (default), ask=prompt user, line=use the line number ;line => 100 ; Only one line by device is currently supported. -; Beware ! only bookmark and softkey entries are allowed after line=> + ; Beware ! only bookmark and softkey entries are allowed after line=> ;bookmark=Hans C.@123 ; Use a softkey to dial 123. Name : 9 char max ;bookmark=Mailbox@011@54 ; 54 shows a mailbox icon. See #define FAV_ICON_ for other values (32 to 63) ;bookmark=Test@*@USTM/violet ; Display an icon if violet is connected (dynamic), only for unistim device -;bookmark=4@Pager@54321@51 ; Display a pager icon and dial 54321 when softkey 4 is pressed +;bookmark=4@Pager@54321@51 ; Display a pager icon and dial 54321 when softkey 4 is pressed ;[violet] ;device=006038abcdef diff --git a/configs/usbradio.conf.sample b/configs/usbradio.conf.sample index 2b62ea809..5ba9815ca 100644 --- a/configs/usbradio.conf.sample +++ b/configs/usbradio.conf.sample @@ -30,23 +30,23 @@ ;------------------------------ JITTER BUFFER CONFIGURATION -------------------------- ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an -; USBRADIO 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 USBRADIO channel can't accept jitter, -; thus an enabled jitterbuffer on the receive USBRADIO side will always -; be used if the sending side can create jitter. + ; USBRADIO 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 USBRADIO channel can't accept jitter, + ; thus an enabled jitterbuffer on the receive USBRADIO 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, usualy sent from exotic devices -; and programs. Defaults to 1000. + ; resynchronized. Useful to improve the quality of the voice, with + ; big jumps in/broken timestamps, usualy sent from exotic devices + ; and programs. Defaults to 1000. ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of an USBRADIO -; channel. Two implementations are currenlty available - "fixed" -; (with size always equals to jbmax-size) and "adaptive" (with -; variable size, actually the new jb of IAX2). Defaults to fixed. + ; channel. Two implementations are currenlty 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/configs/users.conf.sample b/configs/users.conf.sample index 171b891c1..9258cd3d6 100644 --- a/configs/users.conf.sample +++ b/configs/users.conf.sample @@ -2,11 +2,11 @@ ; User configuration ; ; Creating entries in users.conf is a "shorthand" for creating individual -; entries in each configuration file. Using users.conf is not intended to +; entries in each configuration file. Using users.conf is not intended to ; provide you with as much flexibility as using the separate configuration ; files (e.g. sip.conf, iax.conf, etc) but is intended to accelerate the ; simple task of adding users. Note that creating individual items (e.g. -; custom SIP peers, IAX friends, etc.) will allow you to override specific +; custom SIP peers, IAX friends, etc.) will allow you to override specific ; parameters within this file. Parameter names here are the same as they ; appear in the other configuration files. There is no way to change the ; value of a parameter here for just one subsystem. diff --git a/configs/voicemail.conf.sample b/configs/voicemail.conf.sample index 5d6397608..25336425d 100644 --- a/configs/voicemail.conf.sample +++ b/configs/voicemail.conf.sample @@ -65,7 +65,7 @@ maxlogins=3 ;userscontext=default ; ; If you need to have an external program, i.e. /usr/bin/myapp -; called when a voicemail is left, delivered, or your voicemailbox +; called when a voicemail is left, delivered, or your voicemailbox ; is checked, uncomment this. ;externnotify=/usr/bin/myapp @@ -93,7 +93,7 @@ maxlogins=3 ;directoryintro=dir-intro ; The character set for voicemail messages can be specified here ;charset=ISO-8859-1 -; The ADSI feature descriptor number to download to +; The ADSI feature descriptor number to download to ;adsifdn=0000000F ; The ADSI security lock code ;adsisec=9BDBF7AC @@ -156,58 +156,58 @@ emaildateformat=%A, %B %d, %Y at %r ; ; enables polling mailboxes for changes. Normally, it will ; ; expect that changes are only made when someone called in ; ; to one of the voicemail applications. -; ; Examples of situations that would require this option are -; ; web interfaces to voicemail or an email client in the case +; ; Examples of situations that would require this option are +; ; web interfaces to voicemail or an email client in the case ; ; of using IMAP storage. ; ;pollfreq=30 ; If the "pollmailboxes" option is enabled, this option ; ; sets the polling frequency. The default is once every ; ; 30 seconds. -; If using IMAP storage, specify whether voicemail greetings should be stored +; If using IMAP storage, specify whether voicemail greetings should be stored ; via IMAP. If no, then greetings are stored as if IMAP storage were not enabled ;imapgreetings=no ; If imapgreetings=yes, then specify which folder to store your greetings in. If ; you do not specify a folder, then INBOX will be used ;greetingsfolder=INBOX -; Some IMAP server implementations store folders under INBOX instead of +; Some IMAP server implementations store folders under INBOX instead of ; using a top level folder (ex. INBOX/Friends). In this case, user ; imapparentfolder to set the parent folder. For example, Cyrus IMAP does ; NOT use INBOX as the parent. Default is to have no parent folder set. ;imapparentfolder=INBOX -; -; Users may be located in different timezones, or may have different -; message announcements for their introductory message when they enter -; the voicemail system. Set the message and the timezone each user -; hears here. Set the user into one of these zones with the tz= attribute -; in the options field of the mailbox. Of course, language substitution -; still applies here so you may have several directory trees that have -; alternate language choices. -; -; Look in /usr/share/zoneinfo/ for names of timezones. -; Look at the manual page for strftime for a quick tutorial on how the -; variable substitution is done on the values below. -; -; Supported values: +; +; Users may be located in different timezones, or may have different +; message announcements for their introductory message when they enter +; the voicemail system. Set the message and the timezone each user +; hears here. Set the user into one of these zones with the tz= attribute +; in the options field of the mailbox. Of course, language substitution +; still applies here so you may have several directory trees that have +; alternate language choices. +; +; Look in /usr/share/zoneinfo/ for names of timezones. +; Look at the manual page for strftime for a quick tutorial on how the +; variable substitution is done on the values below. +; +; Supported values: ; 'filename' filename of a soundfile (single ticks around the filename ; required) -; ${VAR} variable substitution -; A or a Day of week (Saturday, Sunday, ...) -; B or b or h Month name (January, February, ...) -; d or e numeric day of month (first, second, ..., thirty-first) -; Y Year -; I or l Hour, 12 hour clock -; H Hour, 24 hour clock (single digit hours preceded by "oh") -; k Hour, 24 hour clock (single digit hours NOT preceded by "oh") -; M Minute, with 00 pronounced as "o'clock" +; ${VAR} variable substitution +; A or a Day of week (Saturday, Sunday, ...) +; B or b or h Month name (January, February, ...) +; d or e numeric day of month (first, second, ..., thirty-first) +; Y Year +; I or l Hour, 12 hour clock +; H Hour, 24 hour clock (single digit hours preceded by "oh") +; k Hour, 24 hour clock (single digit hours NOT preceded by "oh") +; M Minute, with 00 pronounced as "o'clock" ; N Minute, with 00 pronounced as "hundred" (US military time) -; P or p AM or PM +; P or p AM or PM ; Q "today", "yesterday" or ABdY -; (*note: not standard strftime value) +; (*note: not standard strftime value) ; q "" (for today), "yesterday", weekday, or ABdY -; (*note: not standard strftime value) -; R 24 hour time, including minute -; -; +; (*note: not standard strftime value) +; R 24 hour time, including minute +; +; ; ; Each mailbox is listed in the form <mailbox>=<password>,<name>,<email>,<pager_email>,<options> ; if the e-mail is specified, a message will be sent when a message is @@ -218,88 +218,88 @@ emaildateformat=%A, %B %d, %Y at %r ; Advanced options example is extension 4069 ; NOTE: All options can be expressed globally in the general section, and ; overridden in the per-mailbox settings, unless listed otherwise. -; +; ; tz=central ; Timezone from zonemessages below. Irrelevant if envelope=no. ; attach=yes ; Attach the voicemail to the notification email *NOT* the pager email ; attachfmt=wav49 ; Which format to attach to the email. Normally this is the -; first format specified in the format parameter above, but this -; option lets you customize the format sent to particular mailboxes. -; Useful if Windows users want wav49, but Linux users want gsm. -; [per-mailbox only] -; saycid=yes ; Say the caller id information before the message. If not described, -; or set to no, it will be in the envelope -; cidinternalcontexts=intern ; Internal Context for Name Playback instead of -; extension digits when saying caller id. + ; first format specified in the format parameter above, but this + ; option lets you customize the format sent to particular mailboxes. + ; Useful if Windows users want wav49, but Linux users want gsm. + ; [per-mailbox only] +; saycid=yes ; Say the caller id information before the message. If not described, + ; or set to no, it will be in the envelope +; cidinternalcontexts=intern ; Internal Context for Name Playback instead of + ; extension digits when saying caller id. ; sayduration=no ; Turn on/off the duration information before the message. [ON by default] ; saydurationm=2 ; Specify the minimum duration to say. Default is 2 minutes -; dialout=fromvm ; Context to dial out from [option 4 from mailbox's advanced menu]. -; If not specified, option 4 will not be listed and dialing out -; from within VoiceMailMain() will not be permitted. -sendvoicemail=yes ; Allow the user to compose and send a voicemail while inside -; VoiceMailMain() [option 5 from mailbox's advanced menu]. -; If set to 'no', option 5 will not be listed. +; dialout=fromvm ; Context to dial out from [option 4 from mailbox's advanced menu]. + ; If not specified, option 4 will not be listed and dialing out + ; from within VoiceMailMain() will not be permitted. +sendvoicemail=yes ; Allow the user to compose and send a voicemail while inside + ; VoiceMailMain() [option 5 from mailbox's advanced menu]. + ; If set to 'no', option 5 will not be listed. ; searchcontexts=yes ; Current default behavior is to search only the default context -; if one is not specified. The older behavior was to search all contexts. -; This option restores the old behavior [DEFAULT=no] -; Note: If you have this option enabled, then you will be required to have -; unique mailbox names across all contexts. Otherwise, an ambiguity is created -; since it is impossible to know which mailbox to retrieve when one is requested. -; callback=fromvm ; Context to call back from -; if not listed, calling the sender back will not be permitted + ; if one is not specified. The older behavior was to search all contexts. + ; This option restores the old behavior [DEFAULT=no] + ; Note: If you have this option enabled, then you will be required to have + ; unique mailbox names across all contexts. Otherwise, an ambiguity is created + ; since it is impossible to know which mailbox to retrieve when one is requested. +; callback=fromvm ; Context to call back from + ; if not listed, calling the sender back will not be permitted ; exitcontext=fromvm ; Context to go to on user exit such as * or 0 -; The default is the current context. + ; The default is the current context. ; review=yes ; Allow sender to review/rerecord their message before saving it [OFF by default ; operator=yes ; Allow sender to hit 0 before/after/during leaving a voicemail to -; reach an operator. This option REQUIRES an 'o' extension in the -; same context (or in exitcontext, if set), as that is where the -; 0 key will send you. [OFF by default] -; envelope=no ; Turn on/off envelope playback before message playback. [ON by default] -; This does NOT affect option 3,3 from the advanced options menu + ; reach an operator. This option REQUIRES an 'o' extension in the + ; same context (or in exitcontext, if set), as that is where the + ; 0 key will send you. [OFF by default] +; envelope=no ; Turn on/off envelope playback before message playback. [ON by default] + ; This does NOT affect option 3,3 from the advanced options menu ; delete=yes ; After notification, the voicemail is deleted from the server. [per-mailbox only] -; This is intended for use with users who wish to receive their -; voicemail ONLY by email. Note: "deletevoicemail" is provided as an -; equivalent option for Realtime configuration. + ; This is intended for use with users who wish to receive their + ; voicemail ONLY by email. Note: "deletevoicemail" is provided as an + ; equivalent option for Realtime configuration. ; volgain=0.0 ; Emails bearing the voicemail may arrive in a volume too -; quiet to be heard. This parameter allows you to specify how -; much gain to add to the message when sending a voicemail. -; NOTE: sox must be installed for this option to work. + ; quiet to be heard. This parameter allows you to specify how + ; much gain to add to the message when sending a voicemail. + ; NOTE: sox must be installed for this option to work. ; nextaftercmd=yes ; Skips to the next message after hitting 7 or 9 to delete/save current message. -; [global option only at this time] + ; [global option only at this time] ; forcename=yes ; Forces a new user to record their name. A new user is -; determined by the password being the same as -; the mailbox number. The default is "no". + ; determined by the password being the same as + ; the mailbox number. The default is "no". ; forcegreetings=no ; This is the same as forcename, except for recording -; greetings. The default is "no". + ; greetings. The default is "no". ; hidefromdir=yes ; Hide this mailbox from the directory produced by app_directory -; The default is "no". + ; The default is "no". ; tempgreetwarn=yes ; Remind the user that their temporary greeting is set ;messagewrap=no ; Enable next/last message to wrap around to -; first (from last) and last (from first) message -; The default is "no". + ; first (from last) and last (from first) message + ; The default is "no". ; minpassword=0 ; Enforce minimum password length ; vm-password=custom_sound -; Customize which sound file is used instead of the default -; prompt that says: "password" + ; Customize which sound file is used instead of the default + ; prompt that says: "password" ; vm-newpassword=custom_sound -; Customize which sound file is used instead of the default -; prompt that says: "Please enter your new password followed by -; the pound key." + ; Customize which sound file is used instead of the default + ; prompt that says: "Please enter your new password followed by + ; the pound key." ; vm-passchanged=custom_sound -; Customize which sound file is used instead of the default -; prompt that says: "Your password has been changed." + ; Customize which sound file is used instead of the default + ; prompt that says: "Your password has been changed." ; vm-reenterpassword=custom_sound -; Customize which sound file is used instead of the default -; prompt that says: "Please re-enter your password followed by -; the pound key" + ; Customize which sound file is used instead of the default + ; prompt that says: "Please re-enter your password followed by + ; the pound key" ; vm-mismatch=custom_sound -; Customize which sound file is used instead of the default -; prompt that says: "The passwords you entered and re-entered -; did not match. Please try again." + ; Customize which sound file is used instead of the default + ; prompt that says: "The passwords you entered and re-entered + ; did not match. Please try again." ; vm-invalid-password=custom_sound -; Customize which sound file is used instead of the default -; prompt that says: ... + ; Customize which sound file is used instead of the default + ; prompt that says: ... ; listen-control-forward-key=# ; Customize the key that fast-forwards message playback ; listen-control-reverse-key=* ; Customize the key that rewinds message playback ; listen-control-pause-key=0 ; Customize the key that pauses/unpauses message playback |