committing changes in /etc after emerge run

Package changes:
+net-misc/asterisk-11.17.1

asterisk/acl.conf

@@ -0,0 +1,80 @@
+;
+; Named Access Control Lists (ACLs)
+;
+; A convenient way to share acl definitions
+;
+; This configuration file is read on startup
+;
+; CLI Commands
+; -----------------------------------------------------------
+;   acl show                         Show all named ACLs configured
+;   acl show <name>                  Show contents of a particular named ACL
+;   reload acl                       Reload configuration file
+;
+; Any configuration that uses ACLs which has been made to be able to use named
+; ACLs will specify a named ACL with the 'acl' option in its configuration in
+; a similar fashion to the usual 'permit' and 'deny' options. Example:
+; acl=my_named_acl
+;
+; Multiple named ACLs can be applied by either comma separating the arguments or
+; just by adding additional ACL lines. Example:
+; acl=my_named_acl
+; acl=my_named_acl2
+;
+; or
+;
+; acl=my_named_acl,my_named_acl2
+;
+; ACLs specified by name are evaluated independently from the ACL specified via
+; permit/deny. In order for an address to pass a given ACL, it must pass both
+; the ACL specified by permit/deny for a given item as well as any named ACLs
+; that were specified.
+;
+;[example_named_acl1]
+;deny=0.0.0.0/0.0.0.0
+;permit=209.16.236.0
+;permit=209.16.236.1
+;
+;[example_named_acl2]
+;permit=0.0.0.0/0.0.0.0
+;deny=10.24.20.171
+;deny=10.24.20.103
+;deny=209.16.236.1
+;
+; example_named_acl1 above shows an example of whitelisting. When whitelisting, the
+; named ACLs should follow a deny that blocks everything (like deny=0.0.0.0/0.0.0.0)
+; The following example explains how combining the ACLs works:
+; <in another configuration>
+; [example_item_with_acl]
+; acl=example_named_acl1
+; acl=example_named_acl2
+;
+; Suppose 209.16.236.0 tries to communicate and the ACL for that example is applied to it...
+; First, example_named_acl1 is evaluated. The address is allowed by that ACL.
+; Next, example_named_acl2 is evaluated. The address isn't blocked by example_named_acl2
+; either, so it passes.
+;
+; Suppose instead 209.16.236.1 tries to communicate and the same ACL is applied.
+; First, example_named_acl1 is evaluated and the address is allowed.
+; However, it is blocked by example_named_acl2, so the address is blocked from the combined
+; ACL.
+;
+; Similarly, the permits/denies in specific configurations that make up an ACL definition
+; are also treated as a separate ACL for evaluation. So if we change the example above to:
+; <in another configuration>
+; [example_item_with_acl]
+; acl=example_named_acl1
+; acl=example_named_acl2
+; deny=209.16.236.0
+;
+; Then 209.16.236.0 will be rejected by the non-named component of the combined ACL even
+; though it passes the two named components.
+;
+;
+; Named ACLs can use ipv6 addresses just like normal ACLs.
+;[ipv6_example_1]
+;deny = ::
+;permit = ::1/128
+;
+;[ipv6_example_2]
+;permit = fe80::21d:bad:fad:2323

asterisk/adsi.conf

@@ -0,0 +1,8 @@
+;
+; Sample ADSI Configuration file
+;
+[intro]
+alignment = center
+greeting => Welcome to the
+greeting => Asterisk
+greeting => Open Source PBX

asterisk/agents.conf

@@ -0,0 +1,97 @@
+;
+; Agent configuration
+;
+
+[general]
+
+[agents]
+;
+; Define maxlogintries to allow agent to try max logins before
+; failed.
+; default to 3
+;
+;maxlogintries=5
+;
+;
+; Define autologoff times if appropriate.  This is how long
+; the phone has to ring with no answer before the agent is
+; automatically logged off (in seconds)
+;
+;autologoff=15
+;
+; Define autologoffunavail to have agents automatically logged
+; out when the extension that they are at returns a CHANUNAVAIL
+; status when a call is attempted to be sent there.
+; Default is "no".
+;
+;autologoffunavail=yes
+;
+; Define ackcall to require a DTMF acknowledgement when
+; a logged-in agent receives a call.  Default is "no".
+; Use the acceptdtmf option to configure what DTMF key
+; press should be used to acknowledge the call. The
+; default is '#'.
+;
+;ackcall=no
+;acceptdtmf=#
+;
+; Define endcall to allow an agent to hangup a call with a
+; DTMF keypress. Default is "yes". Use the enddtmf option to
+; configure which DTMF key will end a call. The default is
+; '*'.
+;
+;endcall=yes
+;enddtmf=*
+;
+; Define wrapuptime.  This is the minimum amount of time when
+; after disconnecting before the caller can receive a new call
+; note this is in milliseconds.
+;
+;wrapuptime=5000
+;
+; Define the default musiconhold for agents
+; musiconhold => music_class
+;
+;musiconhold => default
+;
+; Define updatecdr. This is whether or not to change the source
+; channel in the CDR record for this call to agent/agent_id so
+; that we know which agent generates the call
+;
+;updatecdr=no
+;
+; Group memberships for agents (may change in mid-file)
+;
+;group=3
+;group=1,2
+;group=
+;
+; --------------------------------------------------
+; This section is devoted to recording agent's calls
+; The keywords are global to the chan_agent channel driver
+;
+; Enable recording calls addressed to agents. It's turned off by default.
+;recordagentcalls=yes
+;
+; The format to be used to record the calls: wav, gsm, wav49.
+; By default its "wav".
+;recordformat=gsm
+;
+; The text to be added to the name of the recording. Allows forming a url link.
+;urlprefix=http://localhost/calls/
+;
+; The optional directory to save the conversations in. The default is
+; /var/spool/asterisk/monitor
+;savecallsin=/var/calls
+;
+; An optional custom beep sound file to play to always-connected agents.
+;custom_beep=beep
+;
+; --------------------------------------------------
+;
+; This section contains the agent definitions, in the form:
+;
+; agent => agentid,agentpassword,name
+;
+;agent => 1001,4321,Mark Spencer
+;agent => 1002,4321,Will Meadows

asterisk/alarmreceiver.conf

@@ -0,0 +1,80 @@
+;
+; alarmreceiver.conf
+;
+; Sample configuration file for the Asterisk alarm receiver application.
+;
+
+
+[general]
+
+;
+; Specify a timestamp format for the metadata section of the event files
+; Default is %a %b %d, %Y @ %H:%M:%S %Z
+
+timestampformat = %a %b %d, %Y @ %H:%M:%S %Z
+
+;
+; Specify a command to execute when the caller hangs up
+;
+; Default is none
+;
+
+;eventcmd = yourprogram -yourargs ...
+
+;
+; Specify a spool directory for the event files. This setting is required
+; if you want the app to be useful. Event files written to the spool
+; directory will be of the template event-XXXXXX, where XXXXXX is a random
+; and unique alphanumeric string.
+;
+; Default is none, and the events will be dropped on the floor.
+;
+
+eventspooldir = /tmp
+
+;
+; The alarmreceiver app can either log the events one-at-a-time to individual
+; files in the spool directory, or it can store them until the caller
+; disconnects and write them all to one file.
+;
+; The default setting for logindividualevents is no.
+;
+
+logindividualevents = no
+
+;
+; The timeout for receiving the first DTMF digit is adjustable from 1000 msec.
+; to 10000 msec. The default is 2000 msec. Note: if you wish to test the
+; receiver by entering digits manually, set this to a reasonable time out
+; like 10000 milliseconds.
+
+fdtimeout = 2000
+
+;
+; The timeout for receiving subsequent DTMF digits is adjustable from
+; 110 msec. to 4000 msec. The default is 200 msec. Note: if you wish to test
+; the receiver by entering digits manually, set this to a reasonable time out
+; like 4000 milliseconds.
+;
+
+sdtimeout = 200
+
+;
+; The loudness of the ACK and Kissoff tones is adjustable from 100 to 8192.
+; The default is 8192. This shouldn't need to be messed with, but is included
+; just in case there are problems with signal levels.
+;
+
+loudness = 8192
+
+;
+; The db-family setting allows the user to capture statistics on the number of
+; calls, and the errors the alarm receiver sees. The default is for no
+; db-family name to be defined and the database logging to be turned off.
+;
+
+;db-family = yourfamily:
+
+;
+; End of alarmreceiver.conf
+;

asterisk/alsa.conf

@@ -0,0 +1,78 @@
+;
+; Open Sound System Console Driver Configuration File
+;
+[general]
+;
+; Automatically answer incoming calls on the console?  Choose yes if
+; for example you want to use this as an intercom.
+;
+autoanswer=yes
+;
+; Default context (is overridden with @context syntax)
+;
+context=local
+;
+; Default extension to call
+;
+extension=s
+;
+; Default language
+;
+;language=en
+;
+; Default Music on Hold class to use when this channel is placed on hold in
+; the case that the music class is not set on the channel with
+; Set(CHANNEL(musicclass)=whatever) in the dialplan and the peer channel
+; putting this one on hold did not suggest a class to use.
+;
+;mohinterpret=default
+;
+; Silence suppression can be enabled when sound is over a certain threshold.
+; The value for the threshold should probably be between 500 and 2000 or so,
+; but your mileage may vary.  Use the echo test to evaluate the best setting.
+;silencesuppression = yes
+;silencethreshold = 1000
+;
+; To set which ALSA device to use, change this parameter
+;input_device=hw:0,0
+;output_device=hw:0,0
+
+;
+; Default mute state (can also be toggled via CLI)
+;mute=true
+
+;
+; If enabled, no audio capture device will be opened.  This is useful on
+; systems where there will be no return audio path, such as overhead pagers.
+;noaudiocapture=true
+
+;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
+; jbenable = yes              ; Enables the use of a jitterbuffer on the receiving side of an
+                              ; ALSA channel. Defaults to "no". An enabled jitterbuffer will
+                              ; be used only if the sending side can create and the receiving
+                              ; side can not accept jitter. The ALSA channel can't accept jitter,
+                              ; thus an enabled jitterbuffer on the receive ALSA side will always
+                              ; be used if the sending side can create jitter.
+
+; jbmaxsize = 200             ; Max length of the jitterbuffer in milliseconds.
+
+; jbresyncthreshold = 1000    ; Jump in the frame timestamps over which the jitterbuffer is
+                              ; resynchronized. Useful to improve the quality of the voice, with
+                              ; big jumps in/broken timestamps, usually sent from exotic devices
+                              ; and programs. Defaults to 1000.
+
+; jbimpl = fixed              ; Jitterbuffer implementation, used on the receiving side of a SIP
+                              ; channel. Two implementations are currently available - "fixed"
+                              ; (with size always equals to jbmax-size) and "adaptive" (with
+                              ; variable size, actually the new jb of IAX2). Defaults to fixed.
+
+; jbtargetextra = 40          ; This option only affects the jb when 'jbimpl = adaptive' is set.
+                              ; The option represents the number of milliseconds by which the new
+                              ; jitter buffer will pad its size. the default is 40, so without
+                              ; modification, the new jitter buffer will set its size to the jitter
+                              ; value plus 40 milliseconds. increasing this value may help if your
+                              ; network normally has low jitter, but occasionally has spikes.
+
+; jblog = no                  ; Enables jitterbuffer frame logging. Defaults to "no".
+;-----------------------------------------------------------------------------------
+

asterisk/amd.conf

@@ -0,0 +1,19 @@
+;
+; Answering Machine Detection Configuration
+;
+
+[general]
+initial_silence = 2500		; Maximum silence duration before the greeting.
+				; If exceeded then MACHINE.
+greeting = 1500			; Maximum length of a greeting. If exceeded then MACHINE.
+after_greeting_silence = 800	; Silence after detecting a greeting.
+				; If exceeded then HUMAN
+total_analysis_time = 5000	; Maximum time allowed for the algorithm to decide
+				; on a HUMAN or MACHINE
+min_word_length = 100		; Minimum duration of Voice to considered as a word
+between_words_silence = 50	; Minimum duration of silence after a word to consider
+				; the audio what follows as a new word
+maximum_number_of_words = 3	; Maximum number of words in the greeting.
+				; If exceeded then MACHINE
+maximum_word_length = 5000  ; Maximum duration of a single Voice utterance allowed.
+silence_threshold = 256

asterisk/app_mysql.conf

@@ -0,0 +1,24 @@
+; Configuration file for the MYSQL app addon
+
+[general]
+;
+; Nullvalue governs how NULL values are returned from the database.  In
+; previous versions, the special NULL value was returned as the "NULL"
+; string.  We now provide an option for the behavior, configured globally.
+; nullstring  - the string "NULL"
+; emptystring - the string ""
+; null        - unset the variable
+;
+; WARNING: setting nullvalue=null may have undesireable consequences, in
+; particular if you use subroutines in AEL or the LOCAL() variable construct.
+; You have been warned.  Don't complain if you use that setting in combination
+; with Gosub or AEL and get buggy behavior.
+;
+nullvalue = nullstring
+
+; If set, autoclear will destroy allocated statement and connection resources
+; when the channel ends.  For most usage of the MYSQL app, this is what you
+; want, but it's conceivable that somebody is sharing MYSQL connections across
+; multiple channels, in which case, this should be set to 'no'.  Defaults to
+; 'no', as this was the original behavior.
+autoclear=yes

asterisk/app_skel.conf

@@ -0,0 +1,27 @@
+[general]
+games=3
+cheat=no
+
+[sounds]
+prompt=please-enter-your,number,queue-less-than
+wrong_guess=vm-pls-try-again
+right_guess=auth-thankyou
+too_high=high
+too_low=low
+lose=vm-goodbye
+
+[easy]
+max_number=10
+max_guesses=4
+
+[medium]
+max_number=100
+max_guesses=6
+
+[hard]
+max_number=1000
+max_guesses=7
+
+[nightmare]
+max_number=1000
+max_guesses=1

asterisk/asterisk.adsi

@@ -0,0 +1,159 @@
+;
+; Asterisk default ADSI script
+;
+;
+; Begin with the preamble requirements
+;
+DESCRIPTION "Asterisk PBX"		; Name of vendor
+VERSION 0x00						; Version of stuff
+;SECURITY "_AST"					; Security code
+SECURITY 0X9BDBF7AC				; Security code
+FDN 0x0000000F						; Descriptor number
+
+;
+; Flags
+;
+FLAG "nocallwaiting"
+
+;
+; Predefined strings
+;
+DISPLAY "titles" IS "** Asterisk PBX **"
+DISPLAY "talkingto" IS "Call active." JUSTIFY LEFT
+DISPLAY "callname" IS "$Call1p" JUSTIFY LEFT
+DISPLAY "callnum" IS "$Call1s" JUSTIFY LEFT
+DISPLAY "incoming" IS "Incoming call!" JUSTIFY LEFT
+DISPLAY "ringing" IS "Calling... " JUSTIFY LEFT
+DISPLAY "callended" IS "Call ended." JUSTIFY LEFT
+DISPLAY "missedcall" IS "Missed call." JUSTIFY LEFT
+DISPLAY "busy" IS "Busy." JUSTIFY LEFT
+DISPLAY "reorder" IS "Reorder." JUSTIFY LEFT
+DISPLAY "cwdisabled" IS "Callwait disabled"
+DISPLAY "empty" IS "asdf"
+
+;
+; Begin soft key definitions
+;
+KEY "callfwd" IS "CallFwd" OR "Call Forward"
+	OFFHOOK
+	VOICEMODE
+	WAITDIALTONE
+	SENDDTMF "*60"
+	GOTO "offHook"
+ENDKEY
+
+KEY "vmail_OH" IS "VMail" OR "Voicemail"
+	OFFHOOK
+	VOICEMODE
+	WAITDIALTONE
+	SENDDTMF "8500"
+ENDKEY
+
+KEY "vmail" IS "VMail" OR "Voicemail"
+	SENDDTMF "8500"
+ENDKEY
+
+KEY "backspace" IS "BackSpc" OR "Backspace"
+	BACKSPACE
+ENDKEY
+
+KEY "cwdisable" IS "CWDsble" OR "Disable Call Wait"
+	SENDDTMF "*70"
+	SETFLAG "nocallwaiting"
+	SHOWDISPLAY "cwdisabled" AT 4
+	TIMERCLEAR
+	TIMERSTART 1
+ENDKEY
+
+KEY "cidblock" IS "CIDBlk" OR "Block Callerid"
+	SENDDTMF "*67"
+	SETFLAG "nocallwaiting"
+ENDKEY
+
+;
+; Begin main subroutine
+;
+
+SUB "main" IS
+	IFEVENT NEARANSWER THEN
+		CLEAR
+		SHOWDISPLAY "titles" AT 1 NOUPDATE
+		SHOWDISPLAY "talkingto" AT 2 NOUPDATE
+		SHOWDISPLAY "callname" AT 3
+		SHOWDISPLAY "callnum" AT 4
+		GOTO "stableCall"
+	ENDIF
+	IFEVENT OFFHOOK THEN
+		CLEAR
+		CLEARFLAG "nocallwaiting"
+		CLEARDISPLAY
+		SHOWDISPLAY "titles" AT 1
+		SHOWKEYS "vmail"
+		SHOWKEYS "cidblock"
+		SHOWKEYS "cwdisable" UNLESS "nocallwaiting"
+		GOTO "offHook"
+	ENDIF
+	IFEVENT IDLE THEN
+		CLEAR
+		SHOWDISPLAY "titles" AT 1
+		SHOWKEYS "vmail_OH"
+	ENDIF
+	IFEVENT CALLERID THEN
+		CLEAR
+;		SHOWDISPLAY "titles" AT 1 NOUPDATE
+;		SHOWDISPLAY "incoming" AT 2 NOUPDATE
+		SHOWDISPLAY "callname" AT 3 NOUPDATE
+		SHOWDISPLAY "callnum" AT 4
+	ENDIF
+	IFEVENT RING THEN
+		CLEAR
+		SHOWDISPLAY "titles" AT 1 NOUPDATE
+		SHOWDISPLAY "incoming" AT 2
+	ENDIF
+	IFEVENT ENDOFRING THEN
+		SHOWDISPLAY "missedcall" AT 2
+		CLEAR
+		SHOWDISPLAY "titles" AT 1
+		SHOWKEYS "vmail_OH"
+	ENDIF
+	IFEVENT TIMER THEN
+		CLEAR
+		SHOWDISPLAY "empty" AT 4
+	ENDIF
+ENDSUB
+
+SUB "offHook" IS
+	IFEVENT FARRING THEN
+		CLEAR
+		SHOWDISPLAY "titles" AT 1 NOUPDATE
+		SHOWDISPLAY "ringing" AT 2 NOUPDATE
+		SHOWDISPLAY "callname" at 3 NOUPDATE
+		SHOWDISPLAY "callnum" at 4
+	ENDIF
+	IFEVENT FARANSWER THEN
+		CLEAR
+		SHOWDISPLAY "talkingto" AT 2
+		GOTO "stableCall"
+	ENDIF
+	IFEVENT BUSY THEN
+		CLEAR
+		SHOWDISPLAY "titles" AT 1 NOUPDATE
+		SHOWDISPLAY "busy" AT 2 NOUPDATE
+		SHOWDISPLAY "callname" at 3 NOUPDATE
+		SHOWDISPLAY "callnum" at 4
+	ENDIF
+	IFEVENT REORDER THEN
+		CLEAR
+		SHOWDISPLAY "titles" AT 1 NOUPDATE
+		SHOWDISPLAY "reorder" AT 2 NOUPDATE
+		SHOWDISPLAY "callname" at 3 NOUPDATE
+		SHOWDISPLAY "callnum" at 4
+	ENDIF
+ENDSUB
+
+SUB "stableCall" IS
+	IFEVENT REORDER THEN
+		SHOWDISPLAY "callended" AT 2
+	ENDIF
+ENDSUB
+

asterisk/asterisk.conf

@@ -0,0 +1,102 @@
+[directories](!)
+astetcdir => /etc/asterisk
+astmoddir => /usr/lib64/asterisk/modules
+astvarlibdir => /var/lib/asterisk
+astdbdir => /var/lib/asterisk
+astkeydir => /var/lib/asterisk
+astdatadir => /var/lib/asterisk
+astagidir => /var/lib/asterisk/agi-bin
+astspooldir => /var/spool/asterisk
+astrundir => /var/run/asterisk
+astlogdir => /var/log/asterisk
+astsbindir => /usr/sbin
+
+[options]
+;verbose = 3
+;debug = 3
+;alwaysfork = yes		; Same as -F at startup.
+;nofork = yes			; Same as -f at startup.
+;quiet = yes			; Same as -q at startup.
+;timestamp = yes		; Same as -T at startup.
+;execincludes = yes		; Support #exec in config files.
+;console = yes			; Run as console (same as -c at startup).
+;highpriority = yes		; Run realtime priority (same as -p at
+				; startup).
+;initcrypto = yes		; Initialize crypto keys (same as -i at
+				; startup).
+;nocolor = yes			; Disable console colors.
+;dontwarn = yes			; Disable some warnings.
+;dumpcore = yes			; Dump core on crash (same as -g at startup).
+;languageprefix = yes		; Use the new sound prefix path syntax.
+;systemname = my_system_name	; Prefix uniqueid with a system name for
+				; Global uniqueness issues.
+;autosystemname = yes		; Automatically set systemname to hostname,
+				; uses 'localhost' on failure, or systemname if
+				; set.
+;mindtmfduration = 80		; Set minimum DTMF duration in ms (default 80 ms)
+				; If we get shorter DTMF messages, these will be
+				; changed to the minimum duration
+;maxcalls = 10			; Maximum amount of calls allowed.
+;maxload = 0.9			; Asterisk stops accepting new calls if the
+				; load average exceed this limit.
+;maxfiles = 1000		; Maximum amount of openfiles.
+;minmemfree = 1			; In MBs, Asterisk stops accepting new calls if
+				; the amount of free memory falls below this
+				; watermark.
+;cache_record_files = yes	; Cache recorded sound files to another
+				; directory during recording.
+;record_cache_dir = /tmp	; Specify cache directory (used in conjunction
+				; with cache_record_files).
+;transmit_silence = yes		; Transmit silence while a channel is in a
+				; waiting state, a recording only state, or
+				; when DTMF is being generated.  Note that the
+				; silence internally is generated in raw signed
+				; linear format. This means that it must be
+				; transcoded into the native format of the
+				; channel before it can be sent to the device.
+				; It is for this reason that this is optional,
+				; as it may result in requiring a temporary
+				; codec translation path for a channel that may
+				; not otherwise require one.
+;transcode_via_sln = yes	; Build transcode paths via SLINEAR, instead of
+				; directly.
+;runuser = asterisk		; The user to run as.
+;rungroup = asterisk		; The group to run as.
+;lightbackground = yes		; If your terminal is set for a light-colored
+				; background.
+;forceblackbackground = yes     ; Force the background of the terminal to be 
+                                ; black, in order for terminal colors to show
+                                ; up properly.
+;defaultlanguage = en           ; Default language
+documentation_language = en_US	; Set the language you want documentation
+				; displayed in. Value is in the same format as
+				; locale names.
+;hideconnect = yes		; Hide messages displayed when a remote console
+				; connects and disconnects.
+;lockconfdir = no		; Protect the directory containing the
+				; configuration files (/etc/asterisk) with a
+				; lock.
+;stdexten = gosub		; How to invoke the extensions.conf stdexten.
+				; macro - Invoke the stdexten using a macro as
+				;         done by legacy Asterisk versions.
+				; gosub - Invoke the stdexten using a gosub as
+				;         documented in extensions.conf.sample.
+				; Default gosub.
+;live_dangerously = no		; Enable the execution of 'dangerous' dialplan
+				; functions from external sources (AMI,
+				; etc.) These functions (such as SHELL) are
+				; considered dangerous because they can allow
+				; privilege escalation.
+				; Default yes, for backward compatability.
+
+; Changing the following lines may compromise your security.
+;[files]
+;astctlpermissions = 0660
+;astctlowner = root
+;astctlgroup = apache
+;astctl = asterisk.ctl
+
+[compat]
+pbx_realtime=1.6
+res_agi=1.6
+app_set=1.6

asterisk/calendar.conf

@@ -0,0 +1,109 @@
+;[calendar1]
+;type = ical              ;  type of calendar--currently supported: ical, caldav, exchange, or ews
+;url = https://example.com/home/jdoe/Calendar/   ; URL to shared calendar (Zimbra example)
+;user = jdoe              ; web username
+;secret = supersecret     ; web password
+;refresh = 15             ; refresh calendar every n minutes
+;timeframe = 60           ; number of minutes of calendar data to pull for each refresh period
+;                         ; should always be >= refresh
+;
+; You can set up res_calendar to execute a call upon an upcoming busy status
+; The following fields are available from the ${CALENDAR_EVENT(<field>)} dialplan function:
+;
+; summary     : The VEVENT Summary property or Exchange subject
+; description : The text description of the vent
+; organizer   : The organizer of the event
+; location    : The location field of the event
+; calendar    : The name of the calendar tied to the event
+; uid         : The unique ID for this event
+; start       : Start time of the event
+; end         : The end time of the event
+; busystate   : 0=FREE, 1=TENTATIVE, 2=BUSY
+;
+;autoreminder = 10        ; Override event-defined reminder before each busy status (in mins)
+;
+;channel = SIP/60001      ; Channel to dial
+;context = default        ; Context to connect to on answer
+;extension = 123          ; Extension to connect to on answer
+;
+; or
+;
+;app = Playback          ; Application to execute on answer (instead of context/extension)
+;appdata = tt-weasels    ; Data part of application to execute on answer
+;
+;waittime = 30            ; How long to wait for an answer, defaults to 30 seconds
+;
+; Channel variables can be set on the notification channel. The format is
+; setvar=name=value. Variable subsitution is done on the value to allow the use of dialplan
+; functions like CALENDAR_EVENT. The variables are set in order, so one can use the value
+; of earlier variables in the definition of later ones.
+;
+;setvar = CALLERID(name)=${CALENDAR_EVENT(summary)}
+
+;[calendar2]
+; Note: Support for Exchange Server 2003 
+;
+;type = exchange          ;  type of calendar--currently supported: ical, caldav, exchange, or ews
+;url = https://example.com/exchange/jdoe   ; URL to MS Exchange OWA for user (usually includes exchange/user)
+;user = jdoe              ; Exchange username
+;secret = mysecret        ; Exchange password
+;refresh = 15             ; refresh calendar every n minutes
+;timeframe = 60           ; number of minutes of calendar data to pull for each refresh period
+;                         ; should always be >= refresh
+;
+; You can set up res_calendar to execute a call upon an upcoming busy status
+;autoreminder = 10        ; Override event-defined reminder before each busy status (in mins)
+;
+;channel = SIP/1234       ; Channel to dial
+;context = default        ; Context to connect to on answer
+;extension = 1234         ; Extension to connect to on answer
+;
+; or
+;
+;[calendar3]
+; Note: Support for Exchange Server 2007+
+;
+;type = ews               ; type of calendar--currently supported: ical, caldav, exchange, or ews
+;url = https://example.com/ews/Exchange.asmx ; URL to MS Exchange EWS
+;user = jdoe              ; Exchange username
+;secret = mysecret        ; Exchange password
+;refresh = 15             ; refresh calendar every n minutes
+;timeframe = 60           ; number of minutes of calendar data to pull for each refresh period
+;                         ; should always be >= refresh
+;
+; You can set up res_calendar to execute a call upon an upcoming busy status
+;autoreminder = 10        ; Override event-defined reminder before each busy status (in mins)
+;
+;channel = SIP/1234       ; Channel to dial
+;context = default        ; Context to connect to on answer
+;extension = 1234         ; Extension to connect to on answer
+;
+; or
+;
+;app = Playback          ; Application to execute on answer (instead of context/extension)
+;appdata = tt-weasels    ; Data part of application to execute on answer
+;
+;waittime = 30            ; How long to wait for an answer, defaults to 30 seconds
+
+;[calendar4]
+;type = caldav            ;  type of calendar--currently supported: ical, caldav, exchange, or ews
+;url = https://www.google.com/calendar/dav/username@gmail.com/events/  ; Main GMail calendar (the trailing slash is significant!)
+;user = jdoe@gmail.com    ; username
+;secret = mysecret        ; password
+;refresh = 15             ; refresh calendar every n minutes
+;timeframe = 60           ; number of minutes of calendar data to pull for each refresh period
+;                         ; should always be >= refresh
+;
+; You can set up res_calendar to execute a call upon an upcoming busy status
+;autoreminder = 10        ; Override event-defined reminder before each busy status (in mins)
+;
+;channel = SIP/1234       ; Channel to dial
+;context = default        ; Context to connect to on answer
+;extension = 1234         ; Extension to connect to on answer
+;
+; or
+;
+;app = Playback          ; Application to execute on answer (instead of context/extension)
+;appdata = tt-weasels    ; Data part of application to execute on answer
+;
+;waittime = 30            ; How long to wait for an answer, defaults to 30 seconds

asterisk/ccss.conf

@@ -0,0 +1,205 @@
+;
+; --- Call Completion Supplementary Services ---
+;
+; For more information about CCSS, see the CCSS user documentation
+; https://wiki.asterisk.org/wiki/display/AST/Call+Completion+Supplementary+Services+(CCSS)
+;
+
+[general]
+; The cc_max_requests option is a global limit on the number of
+; CC requests that may be in the Asterisk system at any time.
+;
+;cc_max_requests = 20
+;
+; The cc_STATE_devstate variables listed below can be used to change the
+; default mapping of the internal state machine tracking the state of
+; call completion to an Asterisk Device State value. The acceptable values
+; that can be provided are as follows, with a description of what the
+; equivalent device BLF that this maps to:
+;
+;	UNKNOWN      ; Device is valid but channel didn't know state
+;	NOT_INUSE    ; Device is not used
+;	INUSE        ; Device is in use
+;	BUSY         ; Device is busy
+;	INVALID      ; Device is invalid
+;	UNAVAILABLE  ; Device is unavailable
+;	RINGING      ; Device is ringing
+;	RINGINUSE    ; Device is ringing *and* in use
+;	ONHOLD       ; Device is on hold
+;
+; These states are used to generate DEVICE_STATE information that can be
+; included with Asterisk hints for phones to subscribe to the state information
+; or dialplan to check the state using the EXTENSION_STATE() function or
+; the DEVICE_STATE() function.
+;
+; The states are in the format of: "ccss:TECH/ID" so an example of device
+; SIP/3000 making a CallCompletionRequest() could be checked  by looking at
+; DEVICE_STATE(ccss:SIP/3000) or an Asterisk Hint could be generated such as
+;
+; [hint-context]
+; exten => *843000,hint,ccss:SIP/3000
+;
+; and then accessed with EXTENSION_STATE(*843000@hint-context)
+; or subscribed to with a BLF button on a phone.
+;
+; The available state mapping and default values are:
+;
+; cc_available_devstate = NOT_INUSE
+; cc_offered_devstate = NOT_INUSE
+; cc_caller_requested_devstate = NOT_INUSE
+; cc_active_devstate = INUSE
+; cc_callee_ready_devstate = INUSE
+; cc_caller_busy_devstate = ONHOLD
+; cc_recalling_devstate = RINGING
+; cc_complete_devstate = NOT_INUSE
+; cc_failed_devstate = NOT_INUSE
+
+;
+;============================================
+;           PLEASE READ THIS!!!
+; The options described below should NOT be
+; set in this file. Rather, they should be
+; set per-device in a channel driver
+; configuration file.
+;           PLEASE READ THIS!!!
+;===========================================
+;
+;---------------------------------------------------------------------
+;                                Timers
+;---------------------------------------------------------------------
+;There are three configurable timers for all types of CC: the
+;cc_offer_timer, the ccbs_available_timer, and the ccnr_available_timer.
+;In addition, when using a generic agent, there is a fourth timer,
+;the cc_recall_timer. All timers are configured in seconds, and the
+;values shown below are the defaults.
+;
+;When a caller is offered CCBS or CCNR, the cc_offer_timer will
+;be started. If the caller does not request CC before the
+;cc_offer_timer expires, then the caller will be unable to request
+;CC for this call.
+;
+;cc_offer_timer = 20
+;
+;Once a caller has requested CC, then either the ccbs_available_timer
+;or the ccnr_available_timer will run, depending on the service
+;requested. The reason why there are two separate timers for CCBS
+;and CCNR is that it is reasonable to want to have a shorter timeout
+;configured for CCBS than for CCNR. If the available timer expires
+;before the called party becomes available, then the CC attempt
+;will have failed and monitoring of the called party will stop.
+;
+;ccbs_available_timer = 4800
+;ccnr_available_timer = 7200
+;
+; When using a generic agent, the original caller is called back
+; when one of the original called parties becomes available. The
+; cc_recall_timer tells Asterisk how long it should let the original
+; caller's phone ring before giving up. Please note that this parameter
+; only affects operation when using a generic agent.
+;
+;cc_recall_timer = 20
+;---------------------------------------------------------------------
+;                                Policies
+;---------------------------------------------------------------------
+; Policy settings tell Asterisk how to behave and what sort of
+; resources to allocate in order to facilitate CC. There are two
+; settings to control the actions Asterisk will take.
+;
+; The cc_agent_policy describes the behavior that Asterisk will
+; take when communicating with the caller during CC. There are
+; three possible options.
+;
+;never:   Never offer CC to the caller. Setting the cc_agent_policy
+;         to this value is the way to disable CC for a call.
+;
+;generic: A generic CC agent is one which uses no protocol-specific
+;         mechanisms to offer CC to the caller. Instead, the caller
+;         requests CC using a dialplan function. Due to internal
+;         restrictions, you should only use a generic CC agent on
+;         phones (i.e. not "trunks"). If you are using phones which
+;         do not support a protocol-specific method of using CC, then
+;         generic CC agents are what you should use.
+;
+;native:  A native CC agent is one which uses protocol-specific
+;         signaling to offer CC to the caller and accept CC requests
+;         from the caller. The supported protocols for native CC
+;         agents are SIP, ISDN ETSI PTP, ISDN ETSI PTMP, and Q.SIG
+;cc_agent_policy=never
+;
+; The cc_monitor_policy describes the behavior that Asterisk will
+; take when communicating with the called party during CC. There
+; are four possible options.
+;
+;never:   Analogous to the cc_agent_policy setting. We will never
+;         attempt to request CC services on this interface.
+;
+;generic: Analogous to the cc_agent_policy setting. We will monitor
+;         the called party's progress using protocol-agnostic
+;         capabilities. Like with generic CC agents, generic CC
+;         monitors should only be used for phones.
+;
+;native:  Analogous to the cc_agent_policy setting. We will use
+;         protocol-specific methods to request CC from this interface
+;         and to monitor the interface for availability.
+;
+;always:  If an interface is set to "always," then we will accept
+;         protocol-specific CC offers from the caller and use
+;         a native CC monitor for the remainder of the CC transaction.
+;         However, if the interface does not offer protocol-specific
+;         CC, then we will fall back to using a generic CC monitor
+;         instead. This is a good setting to use for phones for which
+;         you do not know if they support protocol-specific CC
+;         methodologies.
+;cc_monitor_policy=never
+;
+;
+;---------------------------------------------------------------------
+;                              Limits
+;---------------------------------------------------------------------
+;
+; The use of CC requires Asterisk to potentially use more memory than
+; some administrators would like. As such, it is a good idea to limit
+; the number of CC requests that can be in the system at a given time.
+; The values shown below are the defaults.
+;
+; The cc_max_agents setting limits the number of outstanding CC
+; requests a caller may have at any given time. Please note that due
+; to implementation restrictions, this setting is ignored when using
+; generic CC agents. Generic CC agents may only have one outstanding
+; CC request.
+;
+;cc_max_agents = 5
+;
+; The cc_max_monitors setting limits the number of outstanding CC
+; requests can be made to a specific interface at a given time.
+;
+;cc_max_monitors = 5
+;
+;---------------------------------------------------------------------
+;                            Other
+;---------------------------------------------------------------------
+;
+; When using a generic CC agent, the caller who requested CC will be
+; called back when a called party becomes available. When the caller
+; answers his phone, the administrator may opt to have a macro run.
+; What this macro does is up to the administrator. By default there
+; is no callback macro configured.
+;
+;cc_callback_macro=
+;
+; Alternatively, the administrator may run a subroutine. By default
+; there is no callback subroutine configured.  The subroutine should
+; be specified in the format: [[context,]exten,]priority
+;
+;cc_callback_sub=
+;
+; When using an ISDN phone and a generic CC agent, Asterisk is unable
+; to determine the dialstring that should be used when calling back
+; the original caller. Furthermore, if you desire to use any dialstring-
+; specific options, such as distinctive ring, you must set this
+; configuration option. For non-ISDN phones, it is not necessary to
+; set this, since Asterisk can determine the dialstring to use since
+; it is identical to the name of the calling device. By default, there
+; is no cc_agent_dialstring set.
+;
+;cc_agent_dialstring=

asterisk/cdr.conf

@@ -0,0 +1,171 @@
+;
+; Asterisk Call Detail Record engine configuration
+;
+; CDR is Call Detail Record, which provides logging services via a variety of
+; pluggable backend modules.  Detailed call information can be recorded to
+; databases, files, etc.  Useful for billing, fraud prevention, compliance with
+; Sarbanes-Oxley aka The Enron Act, QOS evaluations, and more.
+;
+
+[general]
+
+; Define whether or not to use CDR logging.  Setting this to "no" will override
+; any loading of backend CDR modules.  Default is "yes".
+;enable=yes
+
+; Define whether or not to log unanswered calls. Setting this to "yes" will
+; report every attempt to ring a phone in dialing attempts, when it was not
+; answered. For example, if you try to dial 3 extensions, and this option is "yes",
+; you will get 3 CDR's, one for each phone that was rung. Default is "no". Some
+; find this information horribly useless. Others find it very valuable. Note, in "yes"
+; mode, you will see one CDR, with one of the call targets on one side, and the originating
+; channel on the other, and then one CDR for each channel attempted. This may seem
+; redundant, but cannot be helped.
+;
+; In brief, this option controls the reporting of unanswered calls which only have an A 
+; party. Calls which get offered to an outgoing line, but are unanswered, are still 
+; logged, and that is the intended behaviour. (It also results in some B side CDRs being
+; output, as they have the B side channel as their source channel, and no destination 
+; channel.)
+;unanswered = no
+
+; Define whether or not to log congested calls. Setting this to "yes" will
+; report each call that fails to complete due to congestion conditions. Default
+; is "no".
+;congestion = no
+
+; Normally, CDR's are not closed out until after all extensions are finished
+; executing.  By enabling this option, the CDR will be ended before executing
+; the "h" extension and hangup handlers so that CDR values such as "end" and
+; "billsec" may be retrieved inside of of this extension.
+; The default value is "no".
+;endbeforehexten=no
+
+; 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
+; 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
+; part of the answer time, then the billsec time is incremented one second.
+; The default value is "no".
+;initiatedseconds=no
+
+; Define the CDR batch mode, where instead of posting the CDR at the end of
+; every call, the data will be stored in a buffer to help alleviate load on the
+; asterisk server.  Default is "no".
+;
+; WARNING WARNING WARNING
+; Use of batch mode may result in data loss after unsafe asterisk termination
+; ie. software crash, power failure, kill -9, etc.
+; WARNING WARNING WARNING
+;
+;batch=no
+
+; Define the maximum number of CDRs to accumulate in the buffer before posting
+; them to the backend engines.  'batch' must be set to 'yes'.  Default is 100.
+;size=100
+
+; Define the maximum time to accumulate CDRs in the buffer before posting them
+; to the backend engines.  If this time limit is reached, then it will post the
+; records, regardless of the value defined for 'size'.  'batch' must be set to
+; 'yes'.  Note that time is in seconds.  Default is 300 (5 minutes).
+;time=300
+
+; The CDR engine uses the internal asterisk scheduler to determine when to post
+; records.  Posting can either occur inside the scheduler thread, or a new
+; thread can be spawned for the submission of every batch.  For small batches,
+; it might be acceptable to just use the scheduler thread, so set this to "yes".
+; For large batches, say anything over size=10, a new thread is recommended, so
+; set this to "no".  Default is "no".
+;scheduleronly=no
+
+; When shutting down asterisk, you can block until the CDRs are submitted.  If
+; you don't, then data will likely be lost.  You can always check the size of
+; the CDR batch buffer with the CLI "cdr status" command.  To enable blocking on
+; submission of CDR data during asterisk shutdown, set this to "yes".  Default
+; is "yes".
+;safeshutdown=yes
+
+;
+;
+; CHOOSING A CDR "BACKEND"  (what kind of output to generate)
+;
+; To choose a backend, you have to make sure either the right category is
+; defined in this file, or that the appropriate config file exists, and has the
+; proper definitions in it. If there are any problems, usually, the entry will
+; silently ignored, and you get no output.
+;
+; Also, please note that you can generate CDR records in as many formats as you
+; wish. If you configure 5 different CDR formats, then each event will be logged
+; in 5 different places! In the example config files, all formats are commented
+; out except for the cdr-csv format.
+;
+; Here are all the possible back ends:
+;
+;   csv, custom, manager, odbc, pgsql, radius, sqlite, tds
+;    (also, mysql is available via the asterisk-addons, due to licensing
+;     requirements)
+;   (please note, also, that other backends can be created, by creating
+;    a new backend module in the source cdr/ directory!)
+;
+; Some of the modules required to provide these backends will not build or install
+; unless some dependency requirements are met. Examples of this are pgsql, odbc,
+; etc. If you are not getting output as you would expect, the first thing to do
+; is to run the command "make menuselect", and check what modules are available,
+; by looking in the "2. Call Detail Recording" option in the main menu. If your
+; backend is marked with XXX, you know that the "configure" command could not find
+; the required libraries for that option.
+;
+; To get CDRs to be logged to the plain-jane /var/log/asterisk/cdr-csv/Master.csv
+; file, define the [csv] category in this file. No database necessary. The example
+; config files are set up to provide this kind of output by default.
+;
+; To get custom csv CDR records, make sure the cdr_custom.conf file
+; is present, and contains the proper [mappings] section. The advantage to
+; using this backend, is that you can define which fields to output, and in
+; what order. By default, the example configs are set up to mimic the cdr-csv
+; output. If you don't make any changes to the mappings, you are basically generating
+; the same thing as cdr-csv, but expending more CPU cycles to do so!
+;
+; To get manager events generated, make sure the cdr_manager.conf file exists,
+; and the [general] section is defined, with the single variable 'enabled = yes'.
+;
+; For odbc, make sure all the proper libs are installed, that "make menuselect"
+; shows that the modules are available, and the cdr_odbc.conf file exists, and
+; has a [global] section with the proper variables defined.
+;
+; For pgsql, make sure all the proper libs are installed, that "make menuselect"
+; shows that the modules are available, and the cdr_pgsql.conf file exists, and
+; has a [global] section with the proper variables defined.
+;
+; For logging to radius databases, make sure all the proper libs are installed, that
+; "make menuselect" shows that the modules are available, and the [radius]
+; category is defined in this file, and in that section, make sure the 'radiuscfg'
+; variable is properly pointing to an existing radiusclient.conf file.
+;
+; For logging to sqlite databases, make sure the 'cdr.db' file exists in the log directory,
+; which is usually /var/log/asterisk. Of course, the proper libraries should be available
+; during the 'configure' operation.
+;
+; For tds logging, make sure the proper libraries are available during the 'configure'
+; phase, and that cdr_tds.conf exists and is properly set up with a [global] category.
+;
+; Also, remember, that if you wish to log CDR info to a database, you will have to define
+; a specific table in that databse to make things work! See the doc directory for more details
+; on how to create this table in each database.
+;
+
+[csv]
+usegmtime=yes    ; log date/time in GMT.  Default is "no"
+loguniqueid=yes  ; log uniqueid.  Default is "no"
+loguserfield=yes ; log user field.  Default is "no"
+accountlogs=yes  ; create separate log file for each account code. Default is "yes"
+
+;[radius]
+;usegmtime=yes    ; log date/time in GMT
+;loguniqueid=yes  ; log uniqueid
+;loguserfield=yes ; log user field
+; Set this to the location of the radiusclient-ng configuration file
+; The default is /etc/radiusclient-ng/radiusclient.conf
+;radiuscfg => /usr/local/etc/radiusclient-ng/radiusclient.conf

asterisk/cdr_adaptive_odbc.conf

@@ -0,0 +1,60 @@
+; The point of this module is to allow you log whatever you like in terms of
+; the CDR variables.  Do you want to log uniqueid?  Then simply ensure that
+; your table has that column.  If you don't want the column, ensure that it
+; does not exist in the table structure.  If you'd like to call uniqueid
+; something else in your table, simply provide an alias in the configuration
+; file that maps the standard CDR field name (uniqueid) to whatever column
+; name you like.  Perhaps you'd like some extra CDR values logged that aren't
+; in the standard repertoire of CDR variables (some that come to mind are
+; certain values used for LCR:  route, per_minute_cost, and per_minute_price).
+; Simply set those CDR variables in your dialplan, i.e. Set(CDR(route)=27),
+; ensure that a corresponding column exists in your table, and cdr_adaptive_odbc
+; will do the rest.
+;
+; This configuration defines the connections and tables for which CDRs may
+; be populated.  Each context specifies a different CDR table to be used.
+;
+; The columns in the tables should match up word-for-word (case-insensitive)
+; to the CDR variables set in the dialplan.  The natural advantage to this
+; system is that beyond setting up the configuration file to tell you what
+; tables to look at, there isn't anything more to do beyond creating the
+; columns for the fields that you want, and populating the corresponding
+; CDR variables in the dialplan.  For the builtin variables only, you may
+; create aliases for the real column name.
+;
+; Please note that after adding columns to the database, it is necessary to
+; reload this module to get the new column names and types read.
+;
+; Warning: if you specify two contexts with exactly the same connection and
+; table names, you will get duplicate records in that table.  So be careful.
+;
+
+;[first]
+;connection=mysql1
+;table=cdr
+
+;[second]
+;connection=mysql1
+;table=extracdr
+
+;[third]
+;connection=sqlserver
+;table=AsteriskCDR
+;schema=public ; for databases which support schemas
+;usegmtime=yes ; defaults to no
+;alias src => source
+;alias channel => source_channel
+;alias dst => dest
+;alias dstchannel => dest_channel
+;
+; Any filter specified MUST match exactly or the CDR will be discarded
+;filter accountcode => somename
+;filter src => 123
+; Negative filters are also now available
+;filter src != 456
+;
+; Additionally, we now support setting static values per column.  The reason
+; for this is to allow different sections to specify different values for
+; a certain named column, presumably separated by filters.
+;static "Some Special Value" => identifier_code
+

asterisk/cdr_custom.conf

@@ -0,0 +1,14 @@
+;
+; Mappings for custom config file
+;
+; To get your CSV output in a format tailored to your liking, uncomment the
+; following lines and look for the output in the cdr-custom directory (usually
+; in /var/log/asterisk).  Depending on which mapping you uncomment, you may see
+; Master.csv, Simple.csv, or both.
+;
+;[mappings]
+;Master.csv => ${CSV_QUOTE(${CDR(clid)})},${CSV_QUOTE(${CDR(src)})},${CSV_QUOTE(${CDR(dst)})},${CSV_QUOTE(${CDR(dcontext)})},${CSV_QUOTE(${CDR(channel)})},${CSV_QUOTE(${CDR(dstchannel)})},${CSV_QUOTE(${CDR(lastapp)})},${CSV_QUOTE(${CDR(lastdata)})},${CSV_QUOTE(${CDR(start)})},${CSV_QUOTE(${CDR(answer)})},${CSV_QUOTE(${CDR(end)})},${CSV_QUOTE(${CDR(duration)})},${CSV_QUOTE(${CDR(billsec)})},${CSV_QUOTE(${CDR(disposition)})},${CSV_QUOTE(${CDR(amaflags)})},${CSV_QUOTE(${CDR(accountcode)})},${CSV_QUOTE(${CDR(uniqueid)})},${CSV_QUOTE(${CDR(userfield)})},${CDR(sequence)}
+;
+; High Resolution Time for billsec and duration fields
+;Master.csv => ${CSV_QUOTE(${CDR(clid)})},${CSV_QUOTE(${CDR(src)})},${CSV_QUOTE(${CDR(dst)})},${CSV_QUOTE(${CDR(dcontext)})},${CSV_QUOTE(${CDR(channel)})},${CSV_QUOTE(${CDR(dstchannel)})},${CSV_QUOTE(${CDR(lastapp)})},${CSV_QUOTE(${CDR(lastdata)})},${CSV_QUOTE(${CDR(start)})},${CSV_QUOTE(${CDR(answer)})},${CSV_QUOTE(${CDR(end)})},${CSV_QUOTE(${CDR(duration,f)})},${CSV_QUOTE(${CDR(billsec,f)})},${CSV_QUOTE(${CDR(disposition)})},${CSV_QUOTE(${CDR(amaflags)})},${CSV_QUOTE(${CDR(accountcode)})},${CSV_QUOTE(${CDR(uniqueid)})},${CSV_QUOTE(${CDR(userfield)})},${CDR(sequence)}
+;Simple.csv => ${CSV_QUOTE(${EPOCH})},${CSV_QUOTE(${CDR(src)})},${CSV_QUOTE(${CDR(dst)})}

asterisk/cdr_manager.conf

@@ -0,0 +1,15 @@
+;
+; Asterisk Call Management CDR
+;
+[general]
+enabled = no
+
+; The "mappings" category can be used to define additional "key: value" pairs
+; that will be included in the manager event. (after AccountCode, Source, etc).
+;
+; Each line like "varname => label" will include a "label: ${CDR(varname)}"
+; in the generated event where ${CDR(varname)} its replaced with its value
+;
+;[mappings]
+;rate => Rate
+;carrier => Carrier

asterisk/cdr_mysql.conf

@@ -0,0 +1,60 @@
+;
+; Note - if the database server is hosted on the same machine as the
+; asterisk server, you can achieve a local Unix socket connection by
+; setting hostname=localhost
+;
+; port and sock are both optional parameters.  If hostname is specified
+; and is not "localhost" (you can use address 127.0.0.1 instead), then 
+; cdr_mysql will attempt to connect to the port specified or use the 
+; default port.  If hostname is not specified or if hostname is 
+; "localhost", then cdr_mysql will attempt to connect to the socket file 
+; specified by sock or otherwise use the default socket file.
+;
+;[global]
+;hostname=database.host.name
+;dbname=asteriskcdrdb
+;table=cdr
+;password=password 
+;user=asteriskcdruser
+;port=3306
+;sock=/tmp/mysql.sock
+;timezone=UTC ; Previously called usegmtime
+;
+; If your system's locale differs from mysql database character set,
+; cdr_mysql can damage non-latin characters in CDR variables. Use this
+; option to protect your data.
+;charset=koi8r
+;
+; Older versions of cdr_mysql set the calldate field to whenever the
+; record was posted, rather than the start date of the call.  This flag
+; reverts to the old (incorrect) behavior.  Note that you'll also need
+; to comment out the "start=calldate" alias, below, to use this.
+;compat=no
+;
+; ssl connections (optional)
+;ssl_ca=<path to CA cert>
+;ssl_cert=<path to cert>
+;ssl_key=<path to keyfile>
+;
+; You may also configure the field names used in the CDR table.
+;
+[columns]
+;static "<value>" => <column>
+;alias <cdrvar> => <column>
+alias start => calldate
+;alias clid => <a_field_not_named_clid>
+;alias src => <a_field_not_named_src>
+;alias dst => <a_field_not_named_dst>
+;alias dcontext => <a_field_not_named_dcontext>
+;alias channel => <a_field_not_named_channel>
+;alias dstchannel => <a_field_not_named_dstchannel>
+;alias lastapp => <a_field_not_named_lastapp>
+;alias lastdata => <a_field_not_named_lastdata>
+;alias duration => <a_field_not_named_duration>
+;alias billsec => <a_field_not_named_billsec>
+;alias disposition => <a_field_not_named_disposition>
+;alias amaflags => <a_field_not_named_amaflags>
+;alias accountcode => <a_field_not_named_accountcode>
+;alias userfield => <a_field_not_named_userfield>
+;alias uniqueid => <a_field_not_named_uniqueid>
+

asterisk/cdr_odbc.conf

@@ -0,0 +1,11 @@
+;
+; cdr_odbc.conf
+;
+
+;[global]
+;dsn=MySQL-test
+;loguniqueid=yes
+;dispositionstring=yes
+;table=cdr		;"cdr" is default table name
+;usegmtime=no             ; set to "yes" to log in GMT
+;hrtime=yes		;Enables microsecond accuracy with the billsec and duration fields

asterisk/cdr_pgsql.conf

@@ -0,0 +1,15 @@
+; Sample Asterisk config file for CDR logging to PostgresSQL
+;
+; Note that you can use TLS connections to your database server.
+; This is configured for your PostgreSQL client installation
+; on this system (check for pgsessions.conf)
+
+[global]
+;hostname=localhost
+;port=5432
+;dbname=asterisk
+;password=password
+;user=postgres
+;table=cdr		;SQL table where CDRs will be inserted
+;encoding=LATIN9	; Encoding of logged characters in Asterisk
+;timezone=UTC		; Uncomment if you want datetime fields in UTC/GMT

asterisk/cdr_sqlite3_custom.conf

@@ -0,0 +1,10 @@
+;
+; Mappings for custom config file
+;
+[master] ; currently, only file "master.db" is supported, with only one table at a time.
+;table => cdr
+;columns => calldate, clid, dcontext, channel, dstchannel, lastapp, lastdata, duration, billsec, disposition, amaflags, accountcode, uniqueid, userfield, test
+;values => '${CDR(start)}','${CDR(clid)}','${CDR(dcontext)}','${CDR(channel)}','${CDR(dstchannel)}','${CDR(lastapp)}','${CDR(lastdata)}','${CDR(duration)}','${CDR(billsec)}','${CDR(disposition)}','${CDR(amaflags)}','${CDR(accountcode)}','${CDR(uniqueid)}','${CDR(userfield)}','${CDR(test)}'
+
+;Enable High Resolution Times for billsec and duration fields
+;values => '${CDR(start)}','${CDR(clid)}','${CDR(dcontext)}','${CDR(channel)}','${CDR(dstchannel)}','${CDR(lastapp)}','${CDR(lastdata)}','${CDR(duration,f)}','${CDR(billsec,f)}','${CDR(disposition)}','${CDR(amaflags)}','${CDR(accountcode)}','${CDR(uniqueid)}','${CDR(userfield)}','${CDR(test)}'

asterisk/cdr_syslog.conf

@@ -0,0 +1,83 @@
+;
+; Asterisk Call Detail Records (CDR) - Syslog Backend
+;
+
+; The cdr_syslog module writes CDRs using the facilities provided by syslog.
+;
+; Not only must you configure cdr_syslog from this file (cdr_syslog.conf) but
+; you will also need to make changes to your /etc/syslog.conf before CDRs will
+; be written to syslog.
+;
+; As an example, you can add the following to /etc/syslog.conf:
+;
+;    local4.info        /var/log/asterisk-cdr.log
+;
+; And then instruct syslogd to re-read the configuration file by sending it a
+; HUP signal.  On Linux this can be done like this:
+;
+;    kill -HUP `cat /var/run/syslogd.pid`
+;
+; Finally, you will need to uncomment the [cdr-simple] section below, and restart
+; Asterisk.  When calls are placed, you should start seeing records appear in
+; /var/log/asterisk-cdr.log.
+
+[general]
+; Facility
+;
+; The 'facility' keyword specifies the syslog facility to use when writing out
+; CDRs.
+;
+; Accepted values: One of the following:
+;                  user, local0, local1, local2, local3, local4, local5, local6
+;                  and local7.
+;
+;                  Note: Depending on your platform, the following may also be
+;                        available:
+;                        auth, authpriv, cron, daemon, ftp, kern, lpr, mail,
+;                        news, syslog, and uucp.
+;
+; Default value:   local4
+
+;facility=local0
+
+; Priority
+;
+; Use the 'priority' keyword to select which of the syslog priority levels to
+; use when logging CDRs.
+;
+; Accepted values: One of the following:
+;                  alert, crit, debug, emerg, err, info, notice, warning
+; Default value:   info
+
+;priority=warn
+
+; Note: The settings for 'facility' and 'priority' in the [general] section
+;       define the default values for all of the logging locations created
+;       below in separate sections.
+
+;[cdr-master]
+;facility = local5
+;priority = debug
+
+; Template
+;
+; The 'template' value allows you to specify a custom format for messages
+; written to syslog.  This is similar to how cdr_custom is configured.
+;
+; Allowed values: A diaplan style string.
+; Default value:  None, this is required field.
+;
+; Note: Because of the way substitution is done, the only meaningful values
+;       available when the record is logged are those available via the CDR()
+;       dialplan function.  All other channel variables will be unavailable.
+
+;template = "${CDR(clid)}","${CDR(src)}","${CDR(dst)}","${CDR(dcontext)}","${CDR(channel)}","${CDR(dstchannel)}","${CDR(lastapp)}","${CDR(lastdata)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}"
+
+; High Resolution Time for billsec and duration fields
+;template = "${CDR(clid)}","${CDR(src)}","${CDR(dst)}","${CDR(dcontext)}","${CDR(channel)}","${CDR(dstchannel)}","${CDR(lastapp)}","${CDR(lastdata)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration,f)}","${CDR(billsec,f)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}"
+;[cdr-simple]
+
+; Since we don't specify a facility or priority for this logging location, the
+; records will use the defaults specified in the [general] section.
+
+;template = "We received a call from ${CDR(src)}"

asterisk/cdr_tds.conf

@@ -0,0 +1,78 @@
+;
+; Asterisk Call Detail Records (CDR) - FreeTDS Backend
+;
+
+;[global]
+
+; Connection
+;
+; Use the 'connection' keyword to specify one of the instance names from your
+; 'freetds.conf' file.  Note that 'freetds.conf' is not an Asterisk
+; configuration file, but one specific to the FreeTDS library.  See the FreeTDS
+; documentation on 'freetds.conf' for more information:
+;
+;         http://www.freetds.org/userguide/freetdsconf.htm
+;
+; Accepted values: One of the connections specified in freetds.conf
+
+;connection=ConnectionFromFreeTDSConf
+
+; Database Name
+;
+; The 'dbname' keyword specifies the database name to use when logging CDRs.
+;
+; Accepted values: Any valid database name
+
+;dbname=AsteriskCDRs
+
+; Database Table Name
+;
+; The 'table' keyword identifies which database table is used to log CDRs.
+;
+; Accepted value: Any valid table name
+; Default value:  If not specified, a table named 'cdr' is assumed
+
+;table=cdr
+
+; Credentials
+;
+; The 'username' and 'password' keywords specify the user credentials that
+; Asterisk should use when connecting to the database.
+;
+; Accepted value: Any valid username and password
+
+;username=mangUsr
+;password=
+
+; Language
+;
+; The 'language' keyword changes the language which are used for error and
+; information messages returned by SQL Server.  Each database and user has their
+; own default value, and this default can be overriden here.
+;
+; Accepted value: Any language installed on the target SQL Server.
+; Default value:  us_english
+
+;language=us_english
+
+; Character Set
+;
+; The 'charset' setting is used to change the character set used when connecting
+; to the database server.  Each database and database user has their own
+; character set setting, and this default can be overriden here.
+;
+; Accepted value: Any valid character set available on the target SQL server.
+; Default value:  iso_1
+
+;charset=BIG5
+
+; High Resolution Times
+;
+; The 'hrtime' setting is used to store high resolution (sub second) times for
+; billsec and duration fields.
+;
+; Accepted value: true or false
+; Default value: false
+
+;hrtime=false
+

asterisk/cel.conf

@@ -0,0 +1,126 @@
+;
+; Asterisk Channel Event Logging (CEL)
+;
+
+; Channel Event Logging is a mechanism to provide fine-grained event information
+; that can be used to generate billing information. Such event information can
+; be recorded to various backend modules.
+;
+
+[general]
+
+; CEL Activation
+;
+; Use the 'enable' keyword to turn CEL on or off.
+;
+; Accepted values: yes and no
+; Default value:   no
+
+;enable=yes
+
+; Application Tracking
+;
+; Use the 'apps' keyword to specify the list of applications for which you want
+; to receive CEL events.  This is a comma separated list of Asterisk dialplan
+; applications, such as Dial, Queue, and Park.
+;
+; Accepted values: A comma separated list of Asterisk dialplan applications
+; Default value:   none
+;
+; Note: You may also use 'all' which will result in CEL events being reported
+;       for all Asterisk applications.  This may affect Asterisk's performance
+;       significantly.
+
+apps=dial,park
+
+; Event Tracking
+;
+; Use the 'events' keyword to specify the list of events which you want to be
+; raised when they occur.  This is a comma separated list of the values in the
+; table below.
+;
+; Accepted values: A comma separated list of one or more of the following:
+;  ALL              -- Generate entries on all events
+;  CHAN_START       -- The time a channel was created
+;  CHAN_END         -- The time a channel was terminated
+;  ANSWER           -- The time a channel was answered (ie, phone taken off-hook)
+;  HANGUP           -- The time at which a hangup occurred
+;  CONF_ENTER       -- The time a channel was connected into a conference room
+;  CONF_EXIT        -- The time a channel was removed from a conference room
+;  CONF_START       -- The time the first person enters a conference room
+;  CONF_END         -- The time the last person left a conference room (and
+;                      turned out the lights?)
+;  APP_START        -- The time a tracked application was started
+;  APP_END          -- the time a tracked application ended
+;  PARK_START       -- The time a call was parked
+;  PARK_END         -- Unpark event
+;  BRIDGE_START     -- The time a bridge is started
+;  BRIDGE_END       -- The time a bridge is ended
+;  BRIDGE_UPDATE    -- This is a replacement channel (Masquerade)
+;  3WAY_START       -- When a 3-way conference starts (usually via attended transfer)
+;  3WAY_END         -- When one or all exit a 3-way conference
+;  BLINDTRANSFER    -- When a blind transfer is initiated
+;  ATTENDEDTRANSFER -- When an attended transfer is initiated
+;  TRANSFER         -- Generic transfer initiated; not used yet...?
+;  PICKUP           -- This channel picked up the peer channel
+;  FORWARD          -- This channel is being forwarded somewhere else
+;  HOOKFLASH        -- So far, when a hookflash event occurs on a DAHDI
+;                      interface
+;  LINKEDID_END     -- The last channel with the given linkedid is retired
+;  USER_DEFINED     -- Triggered from the dialplan, and has a name given by the
+;                      user
+;
+; Default value: none
+;                (Track no events)
+
+events=APP_START,CHAN_START,CHAN_END,ANSWER,HANGUP,BRIDGE_START,BRIDGE_END
+
+; Date Format
+;
+; Use the 'dateformat' keyword to specify the date format used when CEL events
+; are raised.
+;
+; Accepted values: A strftime format string (see man strftime)
+;
+; Example: "%F %T"
+;  -> This gives the date and time in the format "2009-06-23 17:02:35"
+;
+; If this option is not specified, the default format is "<seconds>.<microseconds>"
+; since epoch.  The microseconds field will always be 6 digits in length, meaning it
+; may have leading zeros.
+;
+;dateformat = %F %T
+
+;
+; Asterisk Manager Interface (AMI) CEL Backend
+;
+[manager]
+
+; AMI Backend Activation
+;
+; Use the 'enable' keyword to turn CEL logging to the Asterisk Manager Interface
+; on or off.
+;
+; Accepted values: yes and no
+; Default value:   no
+;enabled=yes
+
+; Use 'show_user_defined' to put "USER_DEFINED" in the EventName header,
+; instead of (by default) just putting the user defined event name there.
+; When enabled the UserDefType header is added for user defined events to
+; provide the user defined event name.
+;
+;show_user_defined=yes
+
+;
+; RADIUS CEL Backend 
+;
+[radius]
+;
+; Log date/time in GMT
+;usegmtime=yes
+;
+; Set this to the location of the radiusclient-ng configuration file
+; The default is /etc/radiusclient-ng/radiusclient.conf
+;radiuscfg => /usr/local/etc/radiusclient-ng/radiusclient.conf
+;

asterisk/cel_custom.conf

@@ -0,0 +1,35 @@
+;
+; Asterisk Channel Event Logging (CEL) - Custom CSV Backend
+;
+
+; This is the configuration file for the customizable CSV backend for CEL
+; logging.
+;
+; In order to create custom CSV logs for CEL, uncomment the template below
+; (Master.csv) and start Asterisk.  Once CEL events are generated, a file will
+; appear in the following location:
+;
+;                 /var/log/asterisk/cel-custom/Master.csv
+;
+; (Note that /var/log/asterisk is the default and may differ on your system)
+;
+; You can also create more than one template if desired.  All logs will appear
+; in the cel-custom directory under your Asterisk logs directory.
+;
+
+;
+; Within a mapping, use the CALLERID() and CHANNEL() functions to retrieve
+; details from the CEL event.  There are also a few variables created by this
+; module that can be used in a mapping:
+;
+;    eventtype   - The name of the CEL event.
+;    eventtime   - The timestamp of the CEL event.
+;    eventenum   - Like eventtype but is "USER_DEFINED" for a user defined event.
+;    userdeftype - User defined event type name from CELGenUserEvent().
+;    eventextra  - Extra data included with this CEL event, typically along with
+;                  an event of type USER_DEFINED from CELGenUserEvent().
+;    BRIDGEPEER  - Bridged peer channel name at the time of the CEL event.
+;                  CHANNEL(peer) could also be used.
+;
+[mappings]
+;Master.csv => ${CSV_QUOTE(${eventtype})},${CSV_QUOTE(${eventtime})},${CSV_QUOTE(${CALLERID(name)})},${CSV_QUOTE(${CALLERID(num)})},${CSV_QUOTE(${CALLERID(ANI)})},${CSV_QUOTE(${CALLERID(RDNIS)})},${CSV_QUOTE(${CALLERID(DNID)})},${CSV_QUOTE(${CHANNEL(exten)})},${CSV_QUOTE(${CHANNEL(context)})},${CSV_QUOTE(${CHANNEL(channame)})},${CSV_QUOTE(${CHANNEL(appname)})},${CSV_QUOTE(${CHANNEL(appdata)})},${CSV_QUOTE(${CHANNEL(amaflags)})},${CSV_QUOTE(${CHANNEL(accountcode)})},${CSV_QUOTE(${CHANNEL(uniqueid)})},${CSV_QUOTE(${CHANNEL(linkedid)})},${CSV_QUOTE(${BRIDGEPEER})},${CSV_QUOTE(${CHANNEL(userfield)})},${CSV_QUOTE(${userdeftype})},${CSV_QUOTE(${eventextra})}

asterisk/cel_odbc.conf

@@ -0,0 +1,108 @@
+;
+; Asterisk Channel Event Logging (CEL) - Adaptive ODBC Backend
+;
+
+; General module options category.
+[general]
+; Use 'show_user_defined' to put "USER_DEFINED" in the eventtype field,
+; instead of (by default) just putting the user defined event name there.
+;
+;show_user_defined=yes
+
+; This configuration defines the connections and tables for which CEL records
+; may be populated.  Each context specifies a different CEL table to be used.
+;
+; The columns in the tables should match up word-for-word (case-insensitive) to
+; the CEL variables set in the dialplan.  The natural advantage to this system
+; is that beyond setting up the configuration file to tell you what tables to
+; look at, there isn't anything more to do beyond creating the columns for the
+; fields that you want, and populating the corresponding CEL variables in the
+; dialplan.
+;
+; Please note that after adding columns to the database, it is necessary to
+; reload this module to get the new column names and types read.
+;
+; Warning: if you specify two contexts with exactly the same connection and
+; table names, you will get duplicate records in that table.  So be careful.
+;
+; CEL FIELDS:
+;	eventtype
+;	  CHANNEL_START = 1
+;	  CHANNEL_END = 2
+;	  HANGUP = 3
+;	  ANSWER = 4
+;	  APP_START = 5
+;	  APP_END = 6
+;	  BRIDGE_START = 7
+;	  BRIDGE_END = 8
+;	  CONF_START = 9
+;	  CONF_END = 10
+;	  PARK_START = 11
+;	  PARK_END = 12
+;	  BLINDTRANSFER = 13
+;	  ATTENDEDTRANSFER = 14
+;	  TRANSFER = 15
+;	  HOOKFLASH = 16
+;	  3WAY_START = 17
+;	  3WAY_END = 18
+;	  CONF_ENTER = 19
+;	  CONF_EXIT = 20
+;	  USER_DEFINED = 21
+;	  LINKEDID_END = 22
+;	  BRIDGE_UPDATE = 23
+;	  PICKUP = 24
+;	  FORWARD = 25
+;	eventtime  (timeval, includes microseconds)
+;	userdeftype (set only if eventtype == USER_DEFINED)
+;	cid_name
+;	cid_num
+;	cid_ani
+;	cid_rdnis
+;	cid_dnid
+;	exten
+;	context
+;	channame
+;	appname
+;	appdata
+;	accountcode
+;	peeraccount
+;	uniqueid
+;	linkedid
+;	amaflags  (an int)
+;	userfield
+;	peer
+;	extra
+
+; The point of this module is to allow you log whatever you like in terms of the
+; CEL variables.  Do you want to log uniqueid?  Then simply ensure that your
+; table has that column.  If you don't want the column, ensure that it does not
+; exist in the table structure.  If you'd like to call uniqueid something else
+; in your table, simply provide an alias in this file that maps the standard CEL
+; field name (uniqueid) to whatever column name you like.
+
+;[first]
+;connection=mysql1
+;table=cel
+
+;[second]
+;connection=mysql1
+;table=extracel
+
+;[third]
+;connection=sqlserver
+;table=AsteriskCEL
+;usegmtime=yes ; defaults to no
+;allowleapsecond=no ; allow leap second in SQL column for eventtime, default yes.
+;alias src => source
+;alias channel => source_channel
+;alias dst => dest
+;alias dstchannel => dest_channel
+
+; Any filter specified MUST match exactly or the event will be discarded
+;filter accountcode => somename
+;filter src => 123
+
+; Additionally, we now support setting static values per column.  Reason
+; for this is to allow different sections to specify different values for
+; a certain named column, presumably separated by filters.
+;static "Some Special Value" => identifier_code

asterisk/cel_pgsql.conf

@@ -0,0 +1,67 @@
+;
+; Asterisk Channel Event Logging (CEL) - PostgreSQL Backend
+;
+
+; Sample Asterisk config file for CEL logging to PostgreSQL
+;
+; CEL field names:
+;
+;	eventtype
+;	  CHANNEL_START = 1
+;	  CHANNEL_END = 2
+;	  HANGUP = 3
+;	  ANSWER = 4
+;	  APP_START = 5
+;	  APP_END = 6
+;	  BRIDGE_START = 7
+;	  BRIDGE_END = 8
+;	  CONF_START = 9
+;	  CONF_END = 10
+;	  PARK_START = 11
+;	  PARK_END = 12
+;	  BLINDTRANSFER = 13
+;	  ATTENDEDTRANSFER = 14
+;	  TRANSFER = 15
+;	  HOOKFLASH = 16
+;	  3WAY_START = 17
+;	  3WAY_END = 18
+;	  CONF_ENTER = 19
+;	  CONF_EXIT = 20
+;	  USER_DEFINED = 21
+;	  LINKEDID_END = 22
+;	  BRIDGE_UPDATE = 23
+;	  PICKUP = 24
+;	  FORWARD = 25
+;	eventtime  (timeval, includes microseconds)
+;	userdeftype (set only if eventtype == USER_DEFINED)
+;	cid_name
+;	cid_num
+;	cid_ani
+;	cid_rdnis
+;	cid_dnid
+;	exten
+;	context
+;	channame
+;	appname
+;	appdata
+;	accountcode
+;	peeraccount
+;	uniqueid
+;	linkedid
+;	amaflags  (an int)
+;	userfield
+;	peer
+;	extra
+
+[global]
+; Use 'show_user_defined' to put "USER_DEFINED" in the eventtype field,
+; instead of (by default) just putting the user defined event name there.
+;
+;show_user_defined=yes
+
+;hostname=localhost
+;port=5432
+;dbname=asterisk
+;password=password
+;user=postgres
+;table=cel		;SQL table where CEL's will be inserted

asterisk/cel_sqlite3_custom.conf

@@ -0,0 +1,24 @@
+;
+; Asterisk Channel Event Logging (CEL) - SQLite 3 Backend
+;
+
+;
+; Mappings for sqlite3 config file
+;
+; Within a mapping, use the CALLERID() and CHANNEL() functions to retrieve
+; details from the CEL event.  There are also a few variables created by this
+; module that can be used in a mapping:
+;
+;    eventtype   - The name of the CEL event.
+;    eventtime   - The timestamp of the CEL event.
+;    eventenum   - Like eventtype but is "USER_DEFINED" for a user defined event.
+;    userdeftype - User defined event type name from CELGenUserEvent().
+;    eventextra  - Extra data included with this CEL event, typically along with
+;                  an event of type USER_DEFINED from CELGenUserEvent().
+;    BRIDGEPEER  - Bridged peer channel name at the time of the CEL event.
+;                  CHANNEL(peer) could also be used.
+;
+;[master] ; currently, only file "master.db" is supported, with only one table at a time.
+;table	=> cel
+;columns	=> eventtype, eventtime, cidname, cidnum, cidani, cidrdnis, ciddnid, context, exten, channame, appname, appdata, amaflags, accountcode, uniqueid, userfield, peer, userdeftype, eventextra
+;values	=> '${eventtype}','${eventtime}','${CALLERID(name)}','${CALLERID(num)}','${CALLERID(ANI)}','${CALLERID(RDNIS)}','${CALLERID(DNID)}','${CHANNEL(context)}','${CHANNEL(exten)}','${CHANNEL(channame)}','${CHANNEL(appname)}','${CHANNEL(appdata)}','${CHANNEL(amaflags)}','${CHANNEL(accountcode)}','${CHANNEL(uniqueid)}','${CHANNEL(userfield)}','${BRIDGEPEER}','${userdeftype}','${eventextra}'

asterisk/cel_tds.conf

@@ -0,0 +1,69 @@
+;
+; Asterisk Channel Event Logging (CEL) - FreeTDS Backend
+;
+
+;[global]
+
+; Connection
+;
+; Use the 'connection' keyword to specify one of the instance names from your
+; 'freetds.conf' file.  Note that 'freetds.conf' is not an Asterisk
+; configuration file, but one specific to the FreeTDS library.  See the FreeTDS
+; documentation on 'freetds.conf' for more information:
+;
+;         http://www.freetds.org/userguide/freetdsconf.htm
+;
+; Accepted values: One of the connections specified in freetds.conf
+
+;connection=ConnectionFromFreeTDSConf
+
+; Database Name
+;
+; The 'dbname' keyword specifies the database name to use when logging CEL
+; records.
+;
+; Accepted values: Any valid database name
+
+;dbname=MalicoHN
+
+; Database Table Name
+;
+; The 'table' keyword identifies which database table is used to log CEL
+; records.
+;
+; Accepted value: Any valid table name
+; Default value:  If not specified, a table named 'cel' is assumed
+
+;table=cel
+
+; Credentials
+;
+; The 'username' and 'password' keywords specify the user credentials that
+; Asterisk should use when connecting to the database.
+;
+; Accepted value: Any valid username and password
+
+;username=mangUsr
+;password=
+
+; Language
+;
+; The 'language' keyword changes the language which are used for error and
+; information messages returned by SQL Server.  Each database and user has their
+; own default value, and this default can be overriden here.
+;
+; Accepted value: Any language installed on the target SQL Server.
+; Default value:  Server default
+
+;language=us_english
+
+; Character Set
+;
+; The 'charset' setting is used to change the character set used when connecting
+; to the database server.  Each database and database user has their own
+; character set setting, and this default can be overriden here.
+;
+; Accepted value: Any valid character set available on the target server.
+; Default value:  Server setting
+
+;charset=BIG5

asterisk/chan_mobile.conf

@@ -0,0 +1,69 @@
+;
+; chan_mobile.conf
+; configuration file for chan_mobile
+;
+
+[general]
+interval=30		; Number of seconds between trying to connect to devices. 
+
+; The following is a list of adapters we use.
+; id must be unique and address is the bdaddr of the adapter from hciconfig.
+; Each adapter may only have one device (headset or phone) connected at a time.
+; Add an [adapter] entry for each adapter you have.
+
+[adapter]
+id=blue
+address=00:09:DD:60:01:A3
+;forcemaster=yes	; attempt to force adapter into master mode. default is no.
+;alignmentdetection=yes ; enable this if you sometimes get 'white noise' on asterisk side of the call
+			; its a bug in the bluetooth adapter firmware, enabling this will compensate for it.
+			; default is no.
+
+[adapter]
+id=dlink
+address=00:80:C8:35:52:78
+
+; The following is a list of the devices we deal with.
+; Every device listed below will be available for calls in and out of Asterisk. 
+; Each device needs an adapter=xxxx entry which determines which bluetooth adapter is used.
+; Use the CLI command 'mobile search' to discover devices.
+; Use the CLI command 'mobile show devices' to see device status.
+;
+; To place a call out through a mobile phone use Dial(Mobile/[device]/NNN.....) or Dial(Mobile/gn/NNN......) in your dialplan.
+; To call a headset use Dial(Mobile/[device]).
+
+[LGTU550]
+address=00:E0:91:7F:46:44	; the address of the phone
+port=4				; the rfcomm port number (from mobile search)
+context=incoming-mobile		; dialplan context for incoming calls
+adapter=dlink			; adapter to use
+group=1				; this phone is in channel group 1
+;sms=no				; support SMS, defaults to yes
+;nocallsetup=yes		; set this only if your phone reports that it supports call progress notification, but does not do it. Motorola L6 for example.
+
+[blackberry]
+address=00:60:57:32:7E:B2
+port=2
+context=incoming-mobile
+adapter=dlink
+group=1
+;blackberry=yes			; set this if you are using a blackberry device
+
+[6310i]
+address=00:60:57:32:7E:B1
+port=13
+context=incoming-mobile
+adapter=dlink
+group=1				; this phone is in channel group 1 also.
+
+[headset]
+address=00:0B:9E:11:AE:C6
+port=1
+type=headset			; This is a headset, not a Phone !
+adapter=blue
+
+[headset1]
+address=00:0B:9E:11:74:A5
+port=1
+type=headset
+adapter=dlink

asterisk/cli.conf

@@ -0,0 +1,12 @@
+;
+; Asterisk CLI configuration
+;
+
+[startup_commands]
+;
+; Any commands listed in this section will get automatically executed
+; when Asterisk starts as a daemon or foreground process (-c).
+;
+;sip set debug on = yes
+;core set verbose 3 = yes
+;core set debug 1 = yes

asterisk/cli_aliases.conf

@@ -0,0 +1,201 @@
+;
+; CLI Aliases configuration
+;
+; This module also registers a "cli show aliases" CLI command to list
+; configured CLI aliases.
+
+[general]
+; Here you define what alias templates you want to use. You can also define
+; multiple templates to use as well. If you do, and there is a conflict, then
+; the first alias defined will win.
+;
+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
+
+
+; Because the Asterisk CLI syntax follows a "module verb argument" syntax,
+; sometimes we run into an issue between being consistant with this format
+; in the core system, and maintaining system friendliness. In order to get
+; around this we're providing some useful aliases by default.
+;
+[friendly]
+hangup request=channel request hangup
+originate=channel originate
+help=core show help
+pri intense debug span=pri set debug intense span
+reload=module reload
+
+; CLI Alias Templates
+; -------------------
+;
+; You can define several alias templates.
+; It works with context templates like all other configuration files
+;
+;[asterisk](!)
+; To create an alias you simply set the variable name as the alias and variable
+; value as the real CLI command you want executed
+;
+;die die die=stop now
+
+;[asterisk16](asterisk)
+; Alias for making voicemail reload actually do module reload app_voicemail.so
+;voicemail reload=module reload app_voicemail.so
+; This will make the CLI command "mr" behave as though it is "module reload".
+;mr=module reload
+;
+;
+; In addition, you could also include a flat file of aliases which is loaded by
+; the [individual_custom] template in the [general] section.
+;
+;[individual_custom]
+;#include "/etc/asterisk/aliases"
+
+; Implemented CLI Alias Templates
+; -------------------------------
+;
+; Below here we have provided you with some templates, easily allowing you to
+; utilize previous Asterisk CLI commands with any version of Asterisk. In this
+; way you will be able to use Asterisk 1.2 and 1.4 style CLI syntax with any
+; version Asterisk going forward into the future.
+;
+; We have also separated out the vanilla syntax into a context template which
+; allows you to keep your custom changes separate of the standard templates
+; we have provided you. In this way you can clearly see your custom changes,
+; and also allowing you to combine various templates as you see fit.
+;
+; The naming scheme we have used is recommended, but certainly is not enforced
+; 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
+
+[asterisk12_tpl](!)
+show channeltypes=core show channeltypes
+show channeltype=core show channeltype
+show manager command=manager show command
+show manager commands=manager show commands
+show manager connected=manager show connected
+show manager eventq=manager show eventq
+rtp no debug=rtp set debug off
+rtp rtcp debug ip=rtcp debug ip
+rtp rtcp debug=rtcp debug
+rtp rtcp no debug=rtcp debug off
+rtp rtcp stats=rtcp stats
+rtp rtcp no stats=rtcp stats off
+stun no debug=stun debug off
+udptl no debug=udptl debug off
+show image formats=core show image formats
+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 globals=core show globals
+show function=core show function
+show application=core show application
+set global=core set global
+show dialplan=dialplan show
+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
+moh classes show=moh show classes
+moh files show=moh show files
+agi no debug=agi debug off
+show agi=agi show
+dump agihtml=agi dumphtml
+show features=feature show
+show indications=indication show
+answer=console answer
+hangup=console hangup
+flash=console flash
+dial=console dial
+mute=console mute
+unmute=console unmute
+transfer=console transfer
+send text=console send text
+autoanswer=console autoanswer
+oss boost=console boost
+console=console active
+save dialplan=dialplan save
+add extension=dialplan add extension
+remove extension=dialplan remove extension
+add ignorepat=dialplan add ignorepat
+remove ignorepat=dialplan remove ignorepat
+include context=dialplan add include
+dont include=dialplan remove include
+extensions reload=dialplan reload
+show translation=core show translation
+convert=file convert
+show queue=queue show
+add queue member=queue add member
+remove queue member=queue remove member
+ael no debug=ael nodebug
+sip debug=sip set debug
+sip no debug=sip set debug off
+show voicemail users=voicemail show users
+show voicemail zones=voicemail show zones
+iax2 trunk debug=iax2 set debug trunk
+iax2 jb debug=iax2 set debug jb
+iax2 no debug=iax2 set debug off
+iax2 no trunk debug=iax2 set debug trunk off
+iax2 no jb debug=iax2 set debug jb off
+show agents=agent show
+show agents online=agent show online
+show memory allocations=memory show allocations
+show memory summary=memory show summary
+show version=core show version
+show version files=core show file version
+show profile=core show profile
+clear profile=core clear profile
+soft hangup=channel request hangup
+
+[asterisk12](asterisk12_tpl)
+; add any additional custom commands you want below here, for example:
+;die quickly=stop now
+
+[asterisk14_tpl](!)
+cdr status=cdr show status
+rtp debug=rtp set debug on
+rtcp debug=rtcp set debug on
+rtcp stats=rtcp set stats on
+stun debug=stun set debug on
+udptl debug=udptl set debug on
+core show globals=dialplan show globals
+core set global=dialplan set global
+core set chanvar=dialplan set chanvar
+agi dumphtml=agi dump html
+ael debug=ael set debug
+funcdevstate list=devstate list
+sip history=sip set history on
+skinny debug=skinny set debug on
+mgcp set debug=mgcp set debug on
+abort shutdown=core abort shutdown
+stop now=core stop now
+stop gracefully=core stop gracefully
+stop when convenient=core stop when convenient
+restart now=core restart now
+restart gracefully=core restart gracefully
+restart when convenient=core restart when convenient
+soft hangup=channel request hangup
+
+[asterisk14](asterisk14_tpl)
+; add any additional custom commands you want below here.
+
+[asterisk11_tpl](!)
+jabber list nodes=xmpp list nodes
+jabber purge nodes=xmpp purge nodes
+jabber delete node=xmpp delete node
+jabber create collection=xmpp create collection
+jabber create leaf=xmpp create leaf
+jabber set debug=xmpp set debug
+jabber show connections=xmpp show connections
+jabber show buddies=xmpp show buddies
+
+[asterisk11](asterisk11_tpl)
+; add any additional custom commands you want below here.

asterisk/cli_permissions.conf

@@ -0,0 +1,82 @@
+;
+; CLI permissions configuration example for Asterisk
+;
+; All the users that you want to connect with asterisk using
+; rasterisk, should have write/read access to the
+; asterisk socket (asterisk.ctl). You could change the permissions
+; of this file in 'asterisk.conf' config parameter: 'astctlpermissions' (0666)
+; found on the [files] section.
+;
+; general options:
+;
+; default_perm = permit | deny
+;                This is the default permissions to apply for a user that
+;                does not has a permissions definided.
+;
+; user options:
+; permit = <command name> | all		; allow the user to run 'command' |
+;					; allow the user to run 'all' the commands
+; deny = <command name> | all		; disallow the user to run 'command' |
+;					; disallow the user to run 'all' commands.
+;
+
+[general]
+
+default_perm=permit	; To leave asterisk working as normal
+			; we should set this parameter to 'permit'
+;
+; Follows the per-users permissions configs.
+;
+; This list is read in the sequence that is being written, so
+; In this example the user 'eliel' is allow to run only the following
+; commands:
+;          sip show peer
+;          core set debug
+;          core set verbose
+; If the user is not specified, the default_perm option will be apply to
+; every command.
+;
+; Notice that you can also use regular expressions to allow or deny access to a
+; certain command like: 'core show application D*'. In this example the user will be
+; allowed to view the documentation for all the applications starting with 'D'.
+; Another regular expression could be: 'channel originate SIP/[0-9]* extension *'
+; allowing the user to use 'channel originate' on a sip channel and with the 'extension'
+; parameter and avoiding the use of the 'application' parameter.
+;
+; We can also use the templates syntax:
+; [supportTemplate](!)
+; deny=all
+; permit=sip show       ; all commands starting with 'sip show' will be allowed
+; permit=core show
+;
+; You can specify permissions for a local group instead of a user,
+; just put a '@' and we will know that is a group.
+; IMPORTANT NOTE: Users permissions overwrite group permissions.
+;
+;[@adm]
+;deny=all
+;permit=sip
+;permit=core
+;
+;
+;[eliel]
+;deny=all
+;permit=sip show peer
+;deny=sip show peers
+;permit=core set
+;
+;
+;User 'tommy' inherits from template 'supportTemplate':
+;	deny=all
+;	permit=sip show
+;	permit=core show
+;[tommy](supportTemplate)
+;permit=core set debug
+;permit=dialplan show
+;
+;
+;[mark]
+;deny=all
+;permit=all
+;
+;

asterisk/codecs.conf

@@ -0,0 +1,153 @@
+[speex]
+; CBR encoding quality [0..10]
+; used only when vbr = false
+quality => 3
+
+; codec complexity [0..10]
+; tradeoff between cpu/quality
+complexity => 2
+
+; perceptual enhancement [true / false]
+; improves clarity of decoded speech
+enhancement => true
+
+; voice activity detection [true / false]
+; reduces bitrate when no voice detected, used only for CBR
+; (implicit in VBR/ABR)
+vad => true
+
+; variable bit rate [true / false]
+; uses bit rate proportionate to voice complexity
+vbr => true
+
+; available bit rate [bps, 0 = off]
+; encoding quality modulated to match this target bit rate
+; not recommended with dtx or pp_vad - may cause bandwidth spikes
+abr => 0
+
+; VBR encoding quality [0-10]
+; floating-point values allowed
+vbr_quality => 4
+
+; discontinuous transmission [true / false]
+; stops transmitting completely when silence is detected
+; pp_vad is far more effective but more CPU intensive
+dtx => false
+
+; preprocessor configuration
+; these options only affect Speex v1.1.8 or newer
+
+; enable preprocessor [true / false]
+; allows dsp functionality below but incurs CPU overhead
+preprocess => false
+
+; preproc voice activity detection [true / false]
+; more advanced equivalent of DTX, based on voice frequencies
+pp_vad => false
+
+; preproc automatic gain control [true / false]
+pp_agc => false
+pp_agc_level => 8000
+
+; preproc denoiser [true / false]
+pp_denoise => false
+
+; preproc dereverb [true / false]
+pp_dereverb => false
+pp_dereverb_decay => 0.4
+pp_dereverb_level => 0.3
+
+
+[plc]
+; for all codecs which do not support native PLC
+; this determines whether to perform generic PLC
+; there is a minor performance penalty for this
+genericplc => true
+
+; Generate custom formats for formats requiring attributes.
+; After defining the custom format, the name used in defining
+; the format can be used throughout Asterisk in the format 'allow'
+; and 'disallow' options.
+;
+; Example: silk8 is a predefined custom format in this config file.
+; Once this config file is loaded, silk8 can be used anywhere a
+; peer's codec capabilities are defined.
+;
+; In sip.conf 'silk8' can be defined as a capability for a peer.
+; [peer1]
+; type=peer
+; host=dynamic
+; disallow=all
+; allow=silk8 ;custom codec defined in codecs.conf
+;
+; LIMITATIONS
+; Custom formats can only be defined at startup.  Any changes to this
+; file made after startup will not take into effect until after Asterisk
+; is restarted.
+;
+
+; Default Custom SILK format definitions, only one custom SILK format per
+; sample rate is allowed.
+[silk8]
+type=silk
+samprate=8000
+fec=true           ; turn on or off encoding with forward error correction.
+                   ; On recommended, off by default.
+packetloss_percentage=10   ; Estimated packet loss percentage in uplink direction.  This
+                           ; affects how much redundancy is built in when using fec.
+                           ; The higher the percentage, the larger amount of bandwidth is
+                           ; used. Default is 0%, 10% is recommended when fec is in use.
+
+maxbitrate=10000    ; Use the table below to make sure a useful bitrate is choosen
+                    ; for maxbitrate.  If not set or value is not within the bounds
+                    ; of the encoder, a default value is chosen.
+                    ;
+                    ;  sample rate | bitrate range
+                    ;  8khz        | 5000 - 20000 bps
+                    ;  12khz       | 7000 - 25000 bps
+                    ;  16khz       | 8000 - 30000 bps
+                    ;  24khz       | 20000- 40000 bps
+                    ;
+;dtx=true       ; Encode using discontinuous transmission mode or not. Turning this
+                ; on will save bandwidth during periods of silence at the cost of
+                ; increased computational complexity.  Off by default.
+
+[silk12]
+type=silk
+samprate=12000
+maxbitrate=12000
+fec=true
+packetloss_percentage=10;
+
+[silk16]
+type=silk
+samprate=16000
+maxbitrate=20000
+fec=true
+packetloss_percentage=10;
+
+[silk24]
+type=silk
+samprate=24000
+maxbitrate=30000
+fec=true
+packetloss_percentage=10;
+
+
+; Default custom CELT codec definitions. Only one custom CELT definition is allowed
+; per a sample rate.
+;[celt44]
+;type=celt
+;samprate=44100  ; The samplerate in hz. This option is required.
+;framesize=480   ; The framesize option represents the duration of each frame in samples.
+                 ; This must be a factor of 2.  This option is only advertised in an SDP
+                 ; when it is set.  Otherwise a default of framesize of 480 is assumed
+                 ; internally
+
+;[celt48]
+;type=celt
+;samprate=48000
+
+;[celt32]
+;type=celt
+;samprate=32000

asterisk/confbridge.conf

@@ -0,0 +1,368 @@
+[general]
+; The general section of this config
+; is not currently used, but reserved
+; for future use.
+
+;
+; --- Default Information ---
+; The default_user and default_bridge sections are applied
+; automatically to all ConfBridge instances invoked without
+; a user, or bridge argument.  No menu is applied by default.
+;
+; Note that while properties of the default_user or default_bridge
+; profile can be overridden, if removed, they will be automatically
+; added and made available to the dialplan upon module load.
+;
+
+; --- ConfBridge User Profile Options ---
+[default_user]
+type=user
+;admin=yes     ; Sets if the user is an admin or not. Off by default.
+;marked=yes    ; Sets if this is a marked user or not. Off by default.
+;startmuted=yes; Sets if all users should start out muted. Off by default
+;music_on_hold_when_empty=yes  ; Sets whether MOH should be played when only
+                               ; one person is in the conference or when the
+                               ; the user is waiting on a marked user to enter
+                               ; the conference. Off by default.
+;music_on_hold_class=default   ; The MOH class to use for this user.
+;quiet=yes     ; When enabled enter/leave prompts and user intros are not played.
+               ; There are some prompts, such as the prompt to enter a PIN number,
+               ; that must be played regardless of what this option is set to.
+               ; Off by default
+;announce_user_count=yes  ; Sets if the number of users should be announced to the
+                          ; caller.  Off by default.
+;announce_user_count_all=yes ; Sets if the number of users should be announced to
+                             ; all the other users in the conference when someone joins.
+                             ; This option can be either set to 'yes' or a number.
+                             ; When set to a number, the announcement will only occur
+                             ; once the user count is above the specified number.
+;announce_only_user=yes   ; Sets if the only user announcement should be played
+                          ; when a channel enters a empty conference.  On by default.
+;wait_marked=yes   ; Sets if the user must wait for a marked user to enter before
+                   ; joining the conference. Off by default.
+;end_marked=yes ; This option will kick every user with this option set in their
+                ; user profile after the last Marked user exists the conference.
+
+;dsp_drop_silence=yes  ; This option drops what Asterisk detects as silence from
+                       ; entering into the bridge.  Enabling this option will drastically
+                       ; improve performance and help remove the buildup of background
+                       ; noise from the conference. Highly recommended for large conferences
+                       ; due to its performance enhancements.
+
+;dsp_talking_threshold=128  ; The time in milliseconds of sound above what the dsp has
+                            ; established as base line silence for a user before a user
+                            ; is considered to be talking.  This value affects several
+                            ; operations and should not be changed unless the impact on
+                            ; call quality is fully understood.
+                            ;
+                            ; What this value affects internally:
+                            ;
+                            ; 1. Audio is only mixed out of a user's incoming audio stream
+                            ;    if talking is detected.  If this value is set too
+                            ;    loose the user will hear themselves briefly each
+                            ;    time they begin talking until the dsp has time to
+                            ;    establish that they are in fact talking.
+                            ; 2. When talk detection AMI events are enabled, this value
+                            ;    determines when talking has begun which results in
+                            ;    an AMI event to fire.  If this value is set too tight
+                            ;    AMI events may be falsely triggered by variants in
+                            ;    room noise.
+                            ; 3. The drop_silence option depends on this value to determine
+                            ;    when the user's audio should be mixed into the bridge
+                            ;    after periods of silence.  If this value is too loose
+                            ;    the beginning of a user's speech will get cut off as they
+                            ;    transition from silence to talking.
+                            ;
+                            ; By default this value is 160 ms. Valid values are 1 through 2^31
+
+;dsp_silence_threshold=2000 ; The time in milliseconds of sound falling within the what
+                            ; the dsp has established as baseline silence before a user
+                            ; is considered be silent.  This value affects several
+                            ; operations and should not be changed unless the impact
+                            ; on call quality is fully understood.
+                            ;
+                            ; What this value affects internally:
+                            ;
+                            ; 1. When talk detection AMI events are enabled, this value
+                            ;    determines when the user has stopped talking after a
+                            ;    period of talking.  If this value is set too low
+                            ;    AMI events indicating the user has stopped talking
+                            ;    may get falsely sent out when the user briefly pauses
+                            ;    during mid sentence.
+                            ; 2. The drop_silence option depends on this value to
+                            ;    determine when the user's audio should begin to be
+                            ;    dropped from the conference bridge after the user
+                            ;    stops talking.  If this value is set too low the user's
+                            ;    audio stream may sound choppy to the other participants.
+                            ;    This is caused by the user transitioning constantly from
+                            ;    silence to talking during mid sentence.
+                            ;
+                            ; The best way to approach this option is to set it slightly above
+                            ; the maximum amount of ms of silence a user may generate during
+                            ; natural speech.
+                            ;
+                            ; By default this value is 2500ms. Valid values are 1 through 2^31
+
+;talk_detection_events=yes ; This option sets whether or not notifications of when a user
+                           ; begins and ends talking should be sent out as events over AMI.
+                           ; By default this option is off.
+
+;denoise=yes ; Sets whether or not a denoise filter should be applied
+             ; to the audio before mixing or not.  Off by default. Requires
+             ; func_speex to be built and installed.  Do not confuse this option
+             ; with drop_silence.  Denoise is useful if there is a lot of background
+             ; noise for a user as it attempts to remove the noise while preserving
+             ; the speech.  This option does NOT remove silence from being mixed into
+             ; the conference and does come at the cost of a slight performance hit.
+
+;jitterbuffer=yes  ; Enabling this option places a jitterbuffer on the user's audio stream
+                   ; before audio mixing is performed.  This is highly recommended but will
+                   ; add a slight delay to the audio.  This option is using the JITTERBUFFER
+                   ; dialplan function's default adaptive jitterbuffer.  For a more fine tuned
+                   ; jitterbuffer, disable this option and use the JITTERBUFFER dialplan function
+                   ; on the user before entering the ConfBridge application.
+
+;pin=1234  ; Sets if this user must enter a PIN number before entering
+           ; the conference.  The PIN will be prompted for.
+;announce_join_leave=yes ; When enabled, this option will prompt the user for a
+                         ; name when entering the conference.  After the name is
+                         ; recorded, it will be played as the user enters and exists
+                         ; the conference. This option is off by default.
+;dtmf_passthrough=yes  ; Sets whether or not DTMF should pass through the conference.
+                       ; This option is off by default.
+;announcement=</path/to/file> ; Play a sound file to the user when they join the conference.
+
+; --- ConfBridge Bridge Profile Options ---
+[default_bridge]
+type=bridge
+;max_members=50                ; This option limits the number of participants for a single
+                               ; conference to a specific number.  By default conferences
+                               ; have no participant limit. After the limit is reached, the
+                               ; conference will be locked until someone leaves.  Note however
+                               ; that an Admin user will always be alowed to join the conference
+                               ; regardless if this limit is reached or not.
+
+;record_conference=yes         ; Records the conference call starting when the first user
+                               ; enters the room, and ending when the last user exits the room.
+                               ; The default recorded filename is
+                               ; 'confbridge-<name of conference bridge>-<start time>.wav
+                               ; and the default format is 8khz slinear.  This file will be
+                               ; located in the configured monitoring directory in asterisk.conf.
+
+;record_file=</path/to/file>   ; When record_conference is set to yes, the specific name of the
+                               ; record file can be set using this option.  Note that since multiple
+                               ; conferences may use the same bridge profile, this may cause issues
+                               ; depending on the configuration.  It is recommended to only use this
+                               ; option dynamically with the CONFBRIDGE() dialplan function. This
+                               ; allows the record name to be specified and a unique name to be chosen.
+                               ; By default, the record_file is stored in Asterisk's spool/monitor directory
+                               ; with a unique filename starting with the 'confbridge' prefix.
+
+;internal_sample_rate=auto     ; Sets the internal native sample rate the
+                               ; conference is mixed at.  This is set to automatically
+                               ; adjust the sample rate to the best quality by default.
+                               ; Other values can be anything from 8000-192000.  If a
+                               ; sample rate is set that Asterisk does not support, the
+                               ; closest sample rate Asterisk does support to the one requested
+                               ; will be used.
+
+;mixing_interval=40     ; Sets the internal mixing interval in milliseconds for the bridge.  This
+                        ; number reflects how tight or loose the mixing will be for the conference.
+                        ; In order to improve performance a larger mixing interval such as 40ms may
+                        ; be chosen.  Using a larger mixing interval comes at the cost of introducing
+                        ; larger amounts of delay into the bridge.  Valid values here are 10, 20, 40,
+                        ; or 80.  By default 20ms is used.
+
+;video_mode = follow_talker; Sets how confbridge handles video distribution to the conference participants.
+                           ; Note that participants wanting to view and be the source of a video feed
+                           ; _MUST_ be sharing the same video codec.  Also, using video in conjunction with
+                           ; with the jitterbuffer currently results in the audio being slightly out of sync
+                           ; with the video.  This is a result of the jitterbuffer only working on the audio
+                           ; stream.  It is recommended to disable the jitterbuffer when video is used.
+                           ;
+                           ; --- MODES ---
+                           ; none: No video sources are set by default in the conference. It is still
+                           ;       possible for a user to be set as a video source via AMI or DTMF action
+                           ;       at any time.
+                           ;
+                           ; follow_talker: The video feed will follow whoever is talking and providing video.
+                           ;
+                           ; last_marked: The last marked user to join the conference with video capabilities
+                           ;              will be the single source of video distributed to all participants.
+                           ;              If multiple marked users are capable of video, the last one to join
+                           ;              is always the source, when that user leaves it goes to the one who
+                           ;              joined before them.
+                           ;
+                           ; first_marked: The first marked user to join the conference with video capabilities
+                           ;               is the single source of video distribution among all participants. If
+                           ;               that user leaves, the marked user to join after them becomes the source.
+
+;language=en               ; Set the language used for announcements to the conference.
+                           ; Default is en (English).
+
+; All sounds in the conference are customizable using the bridge profile options below.
+; Simply state the option followed by the filename or full path of the filename after
+; the option.  Example: sound_had_joined=conf-hasjoin  This will play the conf-hasjoin
+; sound file found in the sounds directory when announcing someone's name is joining the
+; conference.
+
+;sound_join  ; The sound played to everyone when someone enters the conference.
+;sound_leave ; The sound played to everyone when someone leaves the conference.
+;sound_has_joined ; The sound played before announcing someone's name has
+                  ; joined the conference. This is used for user intros.
+                  ; Example "_____ has joined the conference"
+;sound_has_left ; The sound played when announcing someone's name has
+                ; left the conference. This is used for user intros.
+                ; Example "_____ has left the conference"
+;sound_kicked ; The sound played to a user who has been kicked from the conference.
+;sound_muted  ; The sound played when the mute option it toggled on.
+;sound_unmuted  ; The sound played when the mute option it toggled off.
+;sound_only_person ; The sound played when the user is the only person in the conference.
+;sound_only_one ; The sound played to a user when there is only one other
+                ; person is in the conference.
+;sound_there_are  ; The sound played when announcing how many users there
+                  ; are in a conference.
+;sound_other_in_party; ; This file is used in conjunction with 'sound_there_are"
+                       ; when announcing how many users there are in the conference.
+                       ; The sounds are stringed together like this.
+                       ; "sound_there_are" <number of participants> "sound_other_in_party"
+;sound_place_into_conference ; The sound played when someone is placed into the conference
+                             ; after waiting for a marked user. This sound is now deprecated
+                             ; since it was only ever used improperly and correcting that bug
+                             ; made it completely unused.
+;sound_wait_for_leader  ; The sound played when a user is placed into a conference that
+                        ; can not start until a marked user enters.
+;sound_leader_has_left  ; The sound played when the last marked user leaves the conference.
+;sound_get_pin ; The sound played when prompting for a conference pin number.
+;sound_invalid_pin ; The sound played when an invalid pin is entered too many times.
+;sound_locked ; The sound played to a user trying to join a locked conference.
+;sound_locked_now ; The sound played to an admin after toggling the conference to locked mode.
+;sound_unlocked_now; The sound played to an admin after toggling the conference to unlocked mode.
+;sound_error_menu ; The sound played when an invalid menu option is entered.
+;sound_begin ; The sound played to the conference when the first marked user enters the conference.
+
+; --- ConfBridge Menu Options ---
+; The ConfBridge application also has the ability to
+; apply custom DTMF menus to each channel using the
+; application.  Like the User and Bridge profiles
+; a menu is passed in to ConfBridge as an argument in
+; the dialplan.
+;
+; Below is a list of menu actions that can be assigned
+; to a DTMF sequence.
+;
+; To have the first DTMF digit in a sequence be the '#' character, you need to
+; escape it.  If it is not escaped then normal config file processing will
+; think it is a directive like #include.  For example:
+; \#1=toggle_mute                      ; Pressing #1 will toggle the mute setting.
+;
+; A single DTMF sequence can have multiple actions associated with it. This is
+; accomplished by stringing the actions together and using a ',' as the delimiter.
+; Example:  Both listening and talking volume is reset when '5' is pressed.
+; 5=reset_talking_volume, reset_listening_volume
+;
+; playback(<name of audio file>&<name of audio file>)
+                                       ; Playback will play back an audio file to a channel
+                                       ; and then immediately return to the conference.
+                                       ; This file can not be interupted by DTMF.
+                                       ; Mutliple files can be chained together using the
+                                       ; '&' character.
+; playback_and_continue(<name of playback prompt>&<name of playback prompt>)
+                                       ; playback_and_continue will
+                                       ; play back a prompt while continuing to
+                                       ; collect the dtmf sequence.  This is useful
+                                       ; when using a menu prompt that describes all
+                                       ; the menu options.  Note however that any DTMF
+                                       ; during this action will terminate the prompts
+                                       ; playback.  Prompt files can be chained together
+                                       ; using the '&' character as a delimiter.
+; toggle_mute      ; Toggle turning on and off mute.  Mute will make the user silent
+                   ; to everyone else, but the user will still be able to listen in.
+
+; no_op ; This action does nothing (No Operation). Its only real purpose exists for
+        ; being able to reserve a sequence in the config as a menu exit sequence.
+; decrease_listening_volume ; Decreases the channel's listening volume.
+; increase_listening_volume ; Increases the channel's listening volume.
+; reset_listening_volume    ; Reset channel's listening volume to default level.
+
+; decrease_talking_volume ; Decreases the channel's talking volume.
+; increase_talking_volume ; Icreases the channel's talking volume.
+; reset_talking_volume    ; Reset channel's talking volume to default level.
+;
+; dialplan_exec(context,exten,priority)  ; The dialplan_exec action allows a user
+                                         ; to escape from the conference and execute
+                                         ; commands in the dialplan.  Once the dialplan
+                                         ; exits the user will be put back into the
+                                         ; conference.  The possibilities are endless!
+; leave_conference ; This action allows a user to exit the conference and continue
+                   ; execution in the dialplan.
+;
+; admin_kick_last  ; This action allows an Admin to kick the last participant from the
+                   ; conference. This action will only work for admins which allows
+                   ; a single menu to be used for both users and admins.
+;
+; admin_toggle_conference_lock ; This action allows an Admin to toggle locking and
+                               ; unlocking the conference.  Non admins can not use
+                               ; this action even if it is in their menu.
+
+; set_as_single_video_src   ; This action allows any user to set themselves as the
+                            ; single video source distributed to all participants.
+                            ; This will make the video feed stick to them regardless
+                            ; of what the video_mode is set to.
+
+; release_as_single_video_src ; This action allows a user to release themselves as
+                              ; the video source.  If video_mode is not set to "none"
+                              ; this action will result in the conference returning to
+                              ; whatever video mode the bridge profile is using.
+                              ;
+                              ; Note that this action will have no effect if the user
+                              ; is not currently the video source.  Also, the user is
+                              ; not guaranteed by using this action that they will not
+                              ; become the video source again.  The bridge will return
+                              ; to whatever operation the video_mode option is set to
+                              ; upon release of the video src.
+
+; admin_toggle_mute_participants ; This action allows an administrator to toggle the mute
+                                 ; state for all non-admins within a conference.  All
+                                 ; admin users are unaffected by this option.  Note that all
+                                 ; users, regardless of their admin status, are notified
+                                 ; that the conference is muted.
+
+; participant_count        ; This action plays back the number of participants currently
+                           ; in a conference
+
+[sample_user_menu]
+type=menu
+*=playback_and_continue(conf-usermenu)
+*1=toggle_mute
+1=toggle_mute
+*4=decrease_listening_volume
+4=decrease_listening_volume
+*6=increase_listening_volume
+6=increase_listening_volume
+*7=decrease_talking_volume
+7=decrease_talking_volume
+*8=leave_conference
+8=leave_conference
+*9=increase_talking_volume
+9=increase_talking_volume
+
+[sample_admin_menu]
+type=menu
+*=playback_and_continue(conf-adminmenu)
+*1=toggle_mute
+1=toggle_mute
+*2=admin_toggle_conference_lock ; only applied to admin users
+2=admin_toggle_conference_lock  ; only applied to admin users
+*3=admin_kick_last       ; only applied to admin users
+3=admin_kick_last        ; only applied to admin users
+*4=decrease_listening_volume
+4=decrease_listening_volume
+*6=increase_listening_volume
+6=increase_listening_volume
+*7=decrease_talking_volume
+7=decrease_talking_volume
+*8=no_op
+8=no_op
+*9=increase_talking_volume
+9=increase_talking_volume

asterisk/config_test.conf

@@ -0,0 +1,38 @@
+; Config to test config parsing
+; global and item have values that differ from defaults
+; global_defaults and item_defualts are to show all defaults are set
+; there should be an option for every default type, and a custom type
+
+[global]
+intopt=-1
+uintopt=1
+doubleopt=0.1
+sockaddropt=1.2.3.4:1234
+boolopt=true
+boolflag1=true
+boolflag2=false
+boolflag3=true
+deny=0.0.0.0/0
+permit=1.2.3.4/32
+codecopt=!all,ulaw,g729
+stropt=test
+customopt=yes
+
+[global_defaults]
+
+[item]
+intopt=-1
+uintopt=1
+doubleopt=0.1
+sockaddropt=1.2.3.4:1234
+boolopt=true
+boolflag1=true
+boolflag2=false
+boolflag3=true
+acldenyopt=0.0.0.0/0
+aclpermitopt=1.2.3.4/32
+codecopt=!all,ulaw,g729
+stropt=test
+customopt=yes
+
+[item_defaults]

asterisk/console.conf

@@ -0,0 +1,97 @@
+;
+; Configuration for chan_console, a cross-platform console channel driver.
+;
+
+[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.
+; The default value is "no".
+;
+;autoanswer = no
+
+; Set the default context to use for outgoing calls.  This can be overridden by
+; dialing some extension@context, unless the overridecontext option is enabled.
+; The default is "default".
+;
+;context = default
+
+; Set the default extension to use for outgoing calls.  The default is "s".
+;
+;extension = s
+
+; Set the default CallerID for created channels.
+;
+;callerid = MyName Here <(256) 428-6000>
+
+; Set the default language for created channels.
+;
+;language = en
+
+; 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".
+;
+;overridecontext = no    ; if 'no', the last @ will start the context
+                        ; if 'yes' the whole string is an extension.
+
+
+; Default Music on Hold class to use when this channel is placed on hold in
+; the case that the music class is not set on the channel with
+; Set(CHANNEL(musicclass)=whatever) in the dialplan and the peer channel
+; putting this one on hold did not suggest a class to use.
+;
+;mohinterpret=default
+
+;------------------------------ 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.
+
+; jbmaxsize = 200             ; Max length of the jitterbuffer in milliseconds.
+
+; jbresyncthreshold = 1000    ; Jump in the frame timestamps over which the jitterbuffer is
+                              ; resynchronized. Useful to improve the quality of the voice, with
+                              ; big jumps in/broken timestamps, usually sent from exotic devices
+                              ; and programs. Defaults to 1000.
+
+; jbimpl = fixed              ; Jitterbuffer implementation, used on the receiving side of a 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.
+
+; jbtargetextra = 40          ; This option only affects the jb when 'jbimpl = adaptive' is set.
+                              ; The option represents the number of milliseconds by which the new
+                              ; jitter buffer will pad its size. the default is 40, so without
+                              ; modification, the new jitter buffer will set its size to the jitter
+                              ; value plus 40 milliseconds. increasing this value may help if your
+                              ; network normally has low jitter, but occasionally has spikes.
+
+; jblog = no                  ; Enables jitterbuffer frame logging. Defaults to "no".
+;-----------------------------------------------------------------------------------
+
+
+;
+; Any configuration context defined beyond the [general] section configures
+; specific devices for use.
+;
+
+[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.
+autoanswer = no
+context = default
+extension = s
+callerid = MyName Here <(256) 428-6000>
+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.

asterisk/dbsep.conf

@@ -0,0 +1,35 @@
+#
+# Configuration file for dbsep.cgi
+#
+# The purpose of this file is to provide realtime access to a database,
+# possibly through ODBC, without needing to load the ODBC drivers into
+# Asterisk, since there are several backend drivers which are rather
+# buggy.
+#
+# We accomplish this separation by using the res_config_curl realtime
+# driver to connect to a server running dbsep.cgi (or another, which
+# implements the same protocol).
+#
+# This file contains the information necessary to configure dbsep.cgi.
+#
+#
+# Once installed to a web server, you'll need to preload func_curl.so
+# and res_config_curl.so in modules.conf and configure extconfig.conf:
+#
+# voicemail => curl,http://server/path/to/dbsep.cgi/voicemail
+# sippeers => curl,http://server/path/to/dbsep.cgi/sippeers
+#
+
+# The Data Source Name, as specified by the Perl DBI module.
+# Typically, this will be along the lines of 'DBI:mysql:astdbname[:dbhostname]' or 'DBI:Pg:dbname=astdbname;hostname=dbhostname'
+dsn=somedsn
+
+# Connected database user
+dbuser=someuser
+
+# And its password
+dbpass=password
+
+# For most databases, this is fine.  Set to 'no' for Sybase or MS SQL Server.
+backslash_is_escape=yes
+

asterisk/dnsmgr.conf

@@ -0,0 +1,5 @@
+[general]
+;enable=yes		; enable creation of managed DNS lookups
+			;   default is 'no'
+;refreshinterval=1200	; refresh managed DNS lookups every <n> seconds
+			;   default is 300 (5 minutes)

asterisk/dsp.conf

@@ -0,0 +1,43 @@
+[default]
+;
+; Length of sound (in milliseconds) before a period of silence is considered
+; to be a change from talking to silence or a period of noise converts silence
+; to talking.  [default=256]
+;
+;silencethreshold=256
+
+; DTMF Reverse Twist and Normal Twist is the difference in power between the row and column energies.
+;
+; Normal Twist is where the Column energy is greater than the Row energy
+; Reverse Twist is where the Row energy is greater.
+;
+; Power level difference between frequencies for different Administrations/RPOAs
+;		Power Gain		equiv
+;		normal	reverse		dB's
+; AT&T(default) 6.31	2.51		8dB(normal), 4dB(reverse)
+; NTT		3.16	3.16		Max. 5dB
+; Danish	3.98	3.98		Max. 6dB
+; Australian	10.0	10.0		Max. 10dB
+; Brazilian	7.94	7.94		Max. 9dB
+; ETSI		3.98	3.98		Max. 6dB
+
+;previous version compatible AT&T values
+; RADIO_RELAX disabled, and relaxdtmf=no
+;		6.30	2.50		7.99dB(normal), 3.98dB(reverse)
+; RADIO_RELAX disabled, and relaxdtmf=yes
+;		6.30	4.00		7.99dB(normal), 6.02dB(reverse)
+; RADIO_RELAX enabled, and relaxdtmf=no
+;		6.30	2.50		7.99dB(normal), 3.984dB(reverse)
+; RADIO_RELAX enabled, and relaxdtmf=yes
+;		6.30	6.50		7.99dB(normal), 8.13dB(reverse)
+
+;If you don't know what these mean, don't change them.
+;dtmf_normal_twist=6.31
+;dtmf_reverse_twist=2.51
+;relax_dtmf_normal_twist=6.31
+;relax_dtmf_reverse_twist=3.98
+
+;successive number hits/misses of 12.75ms before a digit/nodigit is considered valid
+;dtmf_hits_to_begin=2
+;dtmf_misses_to_end=3
+

asterisk/dundi.conf

@@ -0,0 +1,268 @@
+;
+; DUNDi configuration file
+;
+; For more information about DUNDi, see http://www.dundi.com
+;
+;
+[general]
+;
+; The "general" section contains general parameters relating
+; to the operation of the dundi client and server.
+;
+; The first part should be your complete contact information
+; should someone else in your peer group need to contact you.
+;
+;department=Your Department
+;organization=Your Company, Inc.
+;locality=Your City
+;stateprov=ST
+;country=US
+;email=your@email.com
+;phone=+12565551212
+;
+;
+; Specify bind address and port number.  Default is
+; 4520
+;
+;bindaddr=0.0.0.0
+;port=4520
+;
+; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of the tos parameter.
+;tos=ef
+;
+; Our entity identifier (Should generally be the MAC address of the
+; machine it's running on.  Defaults to the first eth address, but you
+; can override it here, as long as you set it to the MAC of *something*
+; you own!)  The EID can be overridden by a setting in asterisk.conf,
+; or by setting this option.
+;
+;entityid=00:07:E9:3B:76:60
+;
+; Peers shall cache our query responses for the specified time,
+; given in seconds. Default is 3600.
+;
+;cachetime=3600
+;
+; This defines the max depth in which to search the DUNDi system.
+; Note that the maximum time that we will wait for a response is
+; (2000 + 200 * ttl) ms.
+;
+ttl=32
+;
+; If we don't get ACK to our DPDISCOVER within 2000ms, and autokill is set
+; to yes, then we cancel the whole thing (that's enough time for one
+; retransmission only).  This is used to keep things from stalling for a long
+; time for a host that is not available, but would be ill advised for bad
+; connections.  In addition to 'yes' or 'no' you can also specify a number
+; of milliseconds.  See 'qualify' for individual peers to turn on for just
+; a specific peer.
+;
+autokill=yes
+;
+; pbx_dundi creates a rotating key called "secret", under the family
+; 'secretpath'.  The default family is dundi (resulting in
+; the key being held at dundi/secret).
+;
+;secretpath=dundi
+;
+; The 'storehistory' option (also changeable at runtime with
+; 'dundi store history' and 'dundi no store history') will
+; cause the DUNDi engine to keep track of the last several
+; queries and the amount of time each query took to execute
+; for the purpose of tracking slow nodes.  This option is
+; off by default due to performance impacts.
+;
+;storehistory=yes
+
+[mappings]
+;
+; The "mappings" section maps DUNDi contexts
+; to contexts on the local asterisk system.  Remember
+; that numbers that are made available under the e164
+; DUNDi context are regulated by the DUNDi General Peering
+; Agreement (GPA) if you are a member of the DUNDi E.164
+; Peering System.
+;
+; dundi_context => local_context,weight,tech,dest[,options]]
+;
+; 'dundi_context' is the name of the context being requested
+; within the DUNDi request
+;
+; 'local_context' is the name of the context on the local system
+; in which numbers can be looked up for which responses shall be given.
+;
+; 'weight' is the weight to use for the responses provided from this
+; mapping.  The number must be >= 0 and < 60000.  Since it is totally
+; valid to receive multiple responses to a query, responses received
+; with a lower weight are tried first.  Note that the weight has a
+; special meaning in the e164 context - see the GPA for more details.
+;
+; 'tech' is the technology to use (IAX, SIP, H323)
+;
+; 'dest' is the destination to supply for reaching that number.  The
+; following variables can be used in the destination string and will
+; be automatically substituted:
+; ${NUMBER}: The number being requested
+; ${IPADDR}: The IP address to connect to
+; ${SECRET}: The current rotating secret key to be used
+;
+; Further options may include:
+;
+; nounsolicited:  No unsolicited calls of any type permitted via this
+;                 route
+; nocomunsolicit: No commercial unsolicited calls permitted via
+;                 this route
+; residential:    This number is known to be a residence
+; commercial:     This number is known to be a business
+; mobile:         This number is known to be a mobile phone
+; nocomunsolicit: No commercial unsolicited calls permitted via
+;                 this route
+; nopartial:      Do not search for partial matches
+;
+; There *must* exist an entry in mappings for DUNDi to respond
+; to any request, although it may be empty.
+;
+;e164 => dundi-e164-canonical,0,IAX2,dundi:${SECRET}@${IPADDR}/${NUMBER},nounsolicited,nocomunsolicit,nopartial
+;e164 => dundi-e164-customers,100,IAX2,dundi:${SECRET}@${IPADDR}/${NUMBER},nounsolicited,nocomunsolicit,nopartial
+;e164 => dundi-e164-via-pstn,400,IAX2,dundi:${SECRET}@${IPADDR}/${NUMBER},nounsolicited,nocomunsolicit,nopartial
+
+;digexten => default,0,IAX2,guest@lappy/${NUMBER}
+;asdf =>
+
+;
+; Weights for mappings can be set a few different ways:
+;
+; 1) It can be set as a static number.
+;testmap1 => context1,222,IAX2,guest@peer1/${NUMBER}
+;
+; 2) It can be an Asterisk global variable.
+;testmap2 => context2,${DUNDITESTVAR},IAX2,guest@peer2${NUMBER}
+;
+; 3) It can be retrieved using a dialplan function.  This can be extremely
+;    useful if you want to let an external script decide what the weight
+;    in a response shouuld be.
+;testmap3 => context3,${SHELL(echo 123)},IAX2,guest@peer3/${NUMBER}
+;
+; The built in variables ${SECRET}, ${IPADDR} and ${NUMBER} can also be
+; passed to the weight. For example, you could pass the ${NUMBER} value
+; to your SHELL() script and use that to dynamically return a weight.
+;
+; Note than when using a global variable or dialplan function to set the
+; weight for a mapping, that response caching should be disabled if you
+; plan for these values to change frequently at all.  If the results are
+; cached, then any change in value will not take effect until the cache
+; has expired.
+;
+
+;
+; The remaining sections represent the peers
+; that we fundamentally trust.  The section name
+; represents the name and optionally at a specific
+; DUNDi context if you want the trust to be established
+; for only a specific DUNDi context.
+;
+; inkey - What key they will be authenticating to us with
+;
+; outkey - What key we use to authenticate to them
+;
+; host - What their host is
+;
+; port - The port where their host is listening (default: 4520)
+;
+; order - What search order to use.  May be 'primary', 'secondary',
+;         'tertiary' or 'quartiary'.  In large systems, it is beneficial
+;         to only query one up-stream host in order to maximize caching
+;         value.  Adding one with primary and one with secondary gives you
+;         redundancy without sacrificing performance.
+;
+; include - Includes this peer when searching a particular context
+;           for lookup (set "all" to perform all lookups with that
+;           host.  This is also the context in which peers are permitted
+;           to precache.
+;
+; noinclude - Disincludes this peer when searching a particular context
+;             for lookup (set "all" to perform no lookups with that
+;             host.
+;
+; permit - Permits this peer to search a given DUNDi context on
+;          the local system.  Set "all" to permit this host to
+;          lookup all contexts.  This is also a context for which
+;          we will create/forward PRECACHE commands.
+;
+; deny -   Denies this peer to search a given DUNDi context on
+;          the local system.  Set "all" to deny this host to
+;          lookup all contexts.
+;
+; model - inbound, outbound, or symmetric for whether we receive
+;         requests only, transmit requests only, or do both.
+;
+; precache - Utilize/Permit precaching with this peer (to pre
+;            cache means to provide an answer when no request
+;            was made and is used so that machines with few
+;            routes can push those routes up a to a higher level).
+;            outgoing means we send precache routes to this peer,
+;            incoming means we permit this peer to send us
+;            precache routes.  symmetric means we do both.
+;
+; Note: You cannot mix symmetric/outbound model with symmetric/inbound
+; precache, nor can you mix symmetric/inbound model with symmetric/outbound
+; precache.
+;
+;
+; The '*' peer is special and matches an unspecified entity
+;
+
+;
+; Sample Primary e164 DUNDi peer
+;
+;[00:50:8B:F3:75:BB]
+;model = symmetric
+;host = 64.215.96.114
+;inkey = digium
+;outkey = misery
+;include = e164
+;permit = e164
+;qualify = yes
+
+;
+; Sample Secondary e164 DUNDi peer
+;
+;[00:A0:C9:96:92:84]
+;model = symmetric
+;host = misery.digium.com
+;inkey = misery
+;outkey = ourkey
+;include = e164
+;permit = e164
+;qualify = yes
+;order = secondary
+
+;
+; Sample "push mode" downstream host
+;
+;[00:0C:76:96:75:28]
+;model = inbound
+;host = dynamic
+;precache = inbound
+;inkey = littleguy
+;outkey = ourkey
+;include = e164	; In this case used only for precaching
+;permit = e164
+;qualify = yes
+
+;
+; Sample "push mode" upstream host
+;
+;[00:07:E9:3B:76:60]
+;model = outbound
+;precache = outbound
+;host = 216.207.245.34
+;register = yes
+;inkey = dhcp34
+;permit = all ; In this case used only for precaching
+;include = all
+;qualify = yes
+;outkey=foo
+
+;[*]
+;

asterisk/enum.conf

@@ -0,0 +1,22 @@
+;
+; ENUM Configuration for resolving phone numbers over DNS
+;
+; Sample config for Asterisk
+; This file is reloaded at "module reload enum" in the CLI
+;
+[general]
+;
+; The search list for domains may be customized.  Domains are searched
+; in the order they are listed here.
+;
+search => e164.arpa
+;
+; If you'd like to use the E.164.org public ENUM registry in addition
+; to the official e164.arpa one, uncomment the following line
+;
+;search => e164.org
+;
+; As there are more H323 drivers available you have to select to which
+; drive a H323 URI will map. Default is "H323".
+;
+h323driver => H323

asterisk/extconfig.conf

@@ -0,0 +1,102 @@
+;
+; Static and realtime external configuration
+; engine configuration
+;
+; See https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
+; for basic table formatting information.
+;
+[settings]
+;
+; Static configuration files:
+;
+; file.conf => driver,database[,table[,priority]]
+;
+; maps a particular configuration file to the given
+; database driver, database and table (or uses the
+; name of the file as the table if not specified)
+;
+;uncomment to load queues.conf via the odbc engine.
+;
+;queues.conf => odbc,asterisk,ast_config
+;extensions.conf => sqlite,asterisk,ast_config
+;
+; The following files CANNOT be loaded from Realtime storage:
+;	asterisk.conf
+;	extconfig.conf (this file)
+;	logger.conf
+;
+; Additionally, the following files cannot be loaded from
+; Realtime storage unless the storage driver is loaded
+; early using 'preload' statements in modules.conf:
+;	manager.conf
+;	cdr.conf
+;	rtp.conf
+;
+; Named ACLs specified in realtime also can not be used
+; from manager.conf unless the storage driver is preloaded.
+; Attempting to use a realtime stored named ACL before the
+; driver is loaded will result in an invalid ACL which
+; rejects all addresses.
+;
+; Realtime configuration engine
+;
+; maps a particular family of realtime
+; configuration to a given database driver,
+; database and table (or uses the name of
+; the family if the table is not specified
+;
+;example => odbc,asterisk,alttable,1
+;example => mysql,asterisk,alttable,2
+;example2 => ldap,"dc=oxymium,dc=net",example2
+;
+; Additionally, priorities are now supported for use as failover methods
+; for retrieving realtime data.  If one connection fails to retrieve any
+; information, the next sequential priority will be tried next.  This
+; especially works well with ODBC connections, since res_odbc now caches
+; when connection failures occur and prevents immediately retrying those
+; connections until after a specified timeout.  Note:  priorities must
+; start at 1 and be sequential (i.e. if you have only priorities 1, 2,
+; and 4, then 4 will be ignored, because there is no 3).
+;
+;
+; Possible driver backends:
+;
+; "odbc" is shown in the examples below, but is not the only valid realtime
+; engine.  Here are several of the possible options:
+;    odbc ... res_config_odbc
+;    sqlite ... res_config_sqlite
+;    pgsql ... res_config_pgsql
+;    curl ... res_config_curl
+;    ldap ... res_config_ldap
+;    mysql ... res_config_mysql (available via add-ons in menuselect)
+;
+; Note: The res_config_pgsql and res_config_sqlite backends configure the
+; database used in their respective configuration files and ignore the
+; database name configured in this file.
+;
+;iaxusers => odbc,asterisk
+;iaxpeers => odbc,asterisk
+;sippeers => odbc,asterisk
+;sipregs => odbc,asterisk ; (avoid sipregs if possible, e.g. by using a view)
+;voicemail => odbc,asterisk
+;extensions => odbc,asterisk
+;meetme => mysql,general
+;queues => odbc,asterisk
+;queue_members => odbc,asterisk
+;acls => odbc,asterisk
+;musiconhold => mysql,general
+;queue_log => mysql,general
+;
+;
+; While most dynamic realtime engines are automatically used when defined in
+; this file, 'extensions', distinctively, is not.  To activate dynamic realtime
+; extensions, you must turn them on in each respective context within
+; extensions.conf with a switch statement.  The syntax is:
+;      switch => Realtime/[[db_context@]tablename]/<opts>
+; The only option available currently is the 'p' option, which disallows
+; extension pattern queries to the database.  If you have no patterns defined
+; in a particular context, this will save quite a bit of CPU time.  However,
+; note that using dynamic realtime extensions is not recommended anymore as a
+; best practice; instead, you should consider writing a static dialplan with
+; proper data abstraction via a tool like func_odbc.
+

asterisk/extensions.ael

@@ -0,0 +1,456 @@
+//
+// Example AEL config file
+//
+//
+// Static extension configuration file, used by
+// the pbx_ael module. This is where you configure all your
+// inbound and outbound calls in Asterisk.
+//
+// This configuration file is reloaded
+// - With the "ael reload" command in the CLI
+// - With the "reload" command (that reloads everything) in the CLI
+
+// The "Globals" category contains global variables that can be referenced
+// in the dialplan by using the GLOBAL dialplan function:
+//  ${GLOBAL(VARIABLE)}
+// ${${GLOBAL(VARIABLE)}} or ${text${GLOBAL(VARIABLE)}} or any hybrid
+// Unix/Linux environmental variables are reached with the ENV dialplan
+// function: ${ENV(VARIABLE)}
+//
+
+// NOTE! NOTE! NOTE!
+// Asterisk by default will load both extensions.conf and extensions.ael files.
+// Upon loading these files the dialplans generated from both with be merged,
+// so you must make sure that you don't have any overlapping contexts or global
+// variables. If you do, then unexpected behavior may result when the data is
+// merged.
+// NOTE! NOTE! NOTE!
+
+globals {
+	CONSOLE-AEL="Console/dsp"; 		// Console interface for demo
+	//CONSOLE-AEL=Zap/1;
+	//CONSOLE-AEL=Phone/phone0;
+	IAXINFO-AEL=guest;				// IAXtel username/password
+	//IAXINFO-AEL="myuser:mypass";
+	OUTBOUND-TRUNK="Zap/g2";		// Trunk interface
+	//
+	// Note the 'g2' in the OUTBOUND-TRUNK variable above. It specifies which group (defined
+	// in chan_dahdi.conf) to dial, i.e. group 2, and how to choose a channel to use in
+	// the specified group. The four possible options are:
+	//
+	// g: select the lowest-numbered non-busy 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).
+	//
+	OUTBOUND-TRUNKMSD=1;					// MSD digits to strip (usually 1 or 0)
+	//OUTBOUND-TRUNK2=IAX2/user:pass@provider;
+};
+
+//
+// Any category other than "General" and "Globals" represent
+// extension contexts, which are collections of extensions.
+//
+// Extension names may be numbers, letters, or combinations
+// thereof. If an extension name is prefixed by a '_'
+// character, it is interpreted as a pattern rather than a
+// literal.  In patterns, some characters have special meanings:
+//
+//   X - any digit from 0-9
+//   Z - any digit from 1-9
+//   N - any digit from 2-9
+//   [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
+//   . - wildcard, matches anything remaining (e.g. _9011. matches
+//	anything starting with 9011 excluding 9011 itself)
+//   ! - wildcard, causes the matching process to complete as soon as
+//       it can unambiguously determine that no other matches are possible
+//
+// For example the extension _NXXXXXX would match normal 7 digit dialings,
+// while _1NXXNXXXXXX would represent an area code plus phone number
+// preceded by a one.
+//
+// Each step of an extension is ordered by priority, which must
+// always start with 1 to be considered a valid extension.  The priority
+// "next" or "n" means the previous priority plus one, regardless of whether
+// the previous priority was associated with the current extension or not.
+// The priority "same" or "s" means the same as the previously specified
+// priority, again regardless of whether the previous entry was for the
+// same extension.  Priorities may be immediately followed by a plus sign
+// and another integer to add that amount (most useful with 's' or 'n').
+// Priorities may then also have an alias, or label, in
+// parenthesis after their name which can be used in goto situations
+//
+// Contexts contain several lines, one for each step of each
+// extension, which can take one of two forms as listed below,
+// with the first form being preferred.  One may include another
+// context in the current one as well, optionally with a
+// date and time.  Included contexts are included in the order
+// they are listed.
+//
+//context name {
+//	exten-name => {
+//		application(arg1,arg2,...);
+//
+// 	Timing list for includes is
+//
+//   <time range>|<days of week>|<days of month>|<months>
+//
+//	includes {
+//		daytime|9:00-17:00|mon-fri|*|*;
+//      };
+//
+// 	ignorepat can be used to instruct drivers to not cancel dialtone upon
+// 	receipt of a particular pattern.  The most commonly used example is
+// 	of course '9' like this:
+//
+//	ignorepat => 9;
+//
+// 	so that dialtone remains even after dialing a 9.
+//};
+
+
+//
+// Sample entries for extensions.conf
+//
+//
+context ael-dundi-e164-canonical {
+	//
+	// 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);
+};
+
+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
+};
+
+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 ael-dundi-e164-switch {
+	//
+	// Just a wrapper for the switch
+	//
+
+	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;
+	};
+	//
+};
+
+//
+// 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 ${exten}|1;
+	return;
+};
+
+//
+// Here are the entries you need to participate in the IAXTEL
+// call routing system.  Most IAXTEL numbers begin with 1-700, but
+// there are exceptions.  For more information, and to sign
+// up, please go to www.gnophone.com or www.iaxtel.com
+//
+context ael-iaxtel700 {
+	_91700XXXXXXX => Dial(IAX2/${IAXINFO-AEL}@iaxtel.com/${EXTEN:1}@iaxtel);
+};
+
+//
+// The SWITCH statement permits a server to share the dialplan with
+// another server. Use with care: Reciprocal switch statements are not
+// allowed (e.g. both A -> B and B -> A), and the switched server needs
+// to be on-line or else dialing can be severly delayed.
+//
+context ael-iaxprovider {
+	switches {
+	// IAX2/user:[key]@myserver/mycontext;
+	};
+};
+
+context ael-trunkint {
+	//
+	// International long distance through trunk
+	//
+	includes {
+		ael-dundi-e164-lookup;
+	};
+	_9011. => {
+		&ael-dundi-e164(${EXTEN:4});
+		Dial(${OUTBOUND-TRUNK}/${EXTEN:${OUTBOUND-TRUNKMSD}});
+	};
+};
+
+context ael-trunkld {
+	//
+	// Long distance context accessed through trunk
+	//
+	includes {
+		ael-dundi-e164-lookup;
+	};
+	_91NXXNXXXXXX => {
+		&ael-dundi-e164(${EXTEN:1});
+		Dial(${OUTBOUND-TRUNK}/${EXTEN:${OUTBOUND-TRUNKMSD}});
+	};
+};
+
+context ael-trunklocal {
+	//
+	// Local seven-digit dialing accessed through trunk interface
+	//
+	_9NXXXXXX => {
+		Dial(${OUTBOUND-TRUNK}/${EXTEN:${OUTBOUND-TRUNKMSD}});
+	};
+};
+
+context ael-trunktollfree {
+	//
+	// Long distance context accessed through trunk interface
+	//
+
+	_91800NXXXXXX => Dial(${OUTBOUND-TRUNK}/${EXTEN:${OUTBOUND-TRUNKMSD}});
+	_91888NXXXXXX => Dial(${OUTBOUND-TRUNK}/${EXTEN:${OUTBOUND-TRUNKMSD}});
+	_91877NXXXXXX => Dial(${OUTBOUND-TRUNK}/${EXTEN:${OUTBOUND-TRUNKMSD}});
+	_91866NXXXXXX => Dial(${OUTBOUND-TRUNK}/${EXTEN:${OUTBOUND-TRUNKMSD}});
+};
+
+context ael-international {
+	//
+	// 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;
+	};
+};
+
+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;
+	};
+};
+
+//
+// You can use an alternative switch type as well, to resolve
+// 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
+// variable substitution is not performed at load time
+// but is passed to the switch directly (presumably to
+// be substituted in the switch routine itself)
+//
+// lswitch => Loopback/12${EXTEN}@othercontext
+//
+// An "eswitch" is like a switch but the evaluation of
+// variable substitution is performed at runtime before
+// being passed to the switch routine.
+//
+// eswitch => IAX2/context@${CURSERVER}
+
+
+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;
+};
+
+context ael-demo {
+	s => {
+		Wait(1);
+		Answer();
+		Set(TIMEOUT(digit)=5);
+		Set(TIMEOUT(response)=10);
+restart:
+		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(${EXTEN}, "IAX2");
+	8500 => {
+		VoicemailMain();
+		goto s|instructions;
+	};
+	# => {
+		Playback(demo-thanks);
+		Hangup();
+	};
+	t => goto #|1;
+	i => Playback(invalid);
+};
+
+
+//
+// If you wish to use AEL for your default context, remove it
+// from extensions.conf (or change its name or comment it out)
+// and then uncomment the one here.
+//
+
+context ael-default {
+
+// By default we include the demo.  In a production system, you
+// probably don't want to have the demo there.
+
+	includes {
+		ael-demo;
+	};
+//
+// Extensions like the two below can be used for FWD, Nikotel, sipgate etc.
+// Note that you must have a [sipprovider] section in sip.conf whereas
+// the otherprovider.net example does not require such a peer definition
+//
+//_41X. => Dial(SIP/${EXTEN:2}@sipprovider,,r);
+//_42X. => Dial(SIP/user:passwd@${EXTEN:2}@otherprovider.net,30,rT);
+
+// Real extensions would go here. Generally you want real extensions to be
+// 4 or 5 digits long (although there is no such requirement) and start with a
+// single digit that is fairly large (like 6 or 7) so that you have plenty of
+// room to overlap extensions and menu options without conflict.  You can alias
+// them with names, too, and use global variables
+
+// 6245  => {
+//		hint(SIP/Grandstream1&SIP/Xlite1,Joe Schmoe); // Channel hints for presence
+// 		Dial(SIP/Grandstream1,20,rt);                 // permit transfer
+//        Dial(${HINT}/5245},20,rtT);                    // Use hint as listed
+//        switch(${DIALSTATUS}) {
+//        case BUSY:
+//                Voicemail(6245,b);
+//				return;
+//        default:
+//                Voicemail(6245,u);
+//				return;
+//        };
+//       };
+
+// 6361 => Dial(IAX2/JaneDoe,,rm);                // ring without time limit
+// 6389 => Dial(MGCP/aaln/1@192.168.0.14);
+// 6394 => Dial(Local/6275/n);                    // this will dial ${MARK}
+
+// 6275 => &ael-stdexten(6275,${MARK});           // assuming ${MARK} is something like DAHDI/2
+// mark => goto 6275|1;                          // alias mark to 6275
+// 6536 => &ael-stdexten(6236,${WIL});            // Ditto for wil
+// wil  => goto 6236|1;
+//
+// Some other handy things are an extension for checking voicemail via
+// voicemailmain
+//
+// 8500 => {
+//			VoicemailMain();
+//			Hangup();
+//	       };
+//
+// Or a conference room (you'll need to edit meetme.conf to enable this room)
+//
+// 8600 => Meetme(1234);
+//
+// Or playing an announcement to the called party, as soon it answers
+//
+// 8700 => Dial(${MARK},30,A(/path/to/my/announcemsg))
+//
+// For more information on applications, just type "show applications" at your
+// friendly Asterisk CLI prompt.
+//
+// 'show application <command>' will show details of how you
+// use that particular application in this file, the dial plan.
+//
+}

asterisk/extensions.conf

@@ -0,0 +1,858 @@
+; 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
+; - 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.
+;
+[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.
+;
+; XXX Not yet implemented XXX
+;
+static=yes
+;
+; if static=yes and writeprotect=no, you can save dialplan by
+; CLI command "dialplan save" too
+;
+writeprotect=no
+;
+; If autofallthrough is set, then if an extension runs out of
+; 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
+; (this is the original behavior of Asterisk 1.0 and earlier).
+;
+;autofallthrough=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.
+; With 50 extensions, the speedup is 1.32x
+; with 88 extensions, the speedup is 2.23x
+; with 138 extensions, the speedup is 3.44x
+; with 238 extensions, the speedup is 5.8x
+; 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
+; time, no matter the number of extensions.
+;
+; 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 desperate. If you have large dialplans (more than about 50 extensions
+; 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
+; on https://issues.asterisk.org. We have made good progress in providing
+; something compatible with the old matcher; help us finish the job!
+;
+; This value can be switched at runtime using the cli command "dialplan set extenpatternmatchnew true"
+; or "dialplan set extenpatternmatchnew false", so you can experiment to your hearts content.
+;
+;extenpatternmatchnew=no
+;
+; 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
+; through reloads, and even if deleted from the extensions.conf or
+; one of its included files, will remain set to the previous value.
+;
+; NOTE: A complication sets in, if you put your global variables into
+; the AEL file, instead of the extensions.conf file. With clearglobalvars
+; set, a "reload" will often leave the globals vars cleared, because it
+; is not unusual to have extensions.conf (which will have no globals)
+; load after the extensions.ael file (where the global vars are stored).
+; So, with "reload" in this particular situation, first the AEL file will
+; clear and then set all the global vars, then, later, when the extensions.conf
+; file is loaded, the global vars are all cleared, and then not set, because
+; they are not stored in the extensions.conf file.
+;
+clearglobalvars=no
+;
+; User context is where entries from users.conf are registered.  The
+; default value is 'default'
+;
+;userscontext=default
+;
+; You can include other config files, use the #include command
+; (without the ';'). Note that this is different from the "include" command
+; that includes contexts within other contexts. The #include command works
+; in all asterisk configuration files.
+;#include "filename.conf"
+;#include <filename.conf>
+;#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
+; 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.
+;#exec /opt/bin/build-extra-contexts.sh
+;#exec /opt/bin/build-extra-contexts.sh --foo="bar"
+;#exec </opt/bin/build-extra-contexts.sh --foo="bar">
+;#exec "/opt/bin/build-extra-contexts.sh --foo=\"bar\""
+;
+
+; The "Globals" category contains global variables that can be referenced
+; in the dialplan with the GLOBAL dialplan function:
+; ${GLOBAL(VARIABLE)}
+; ${${GLOBAL(VARIABLE)}} or ${text${GLOBAL(VARIABLE)}} or any hybrid
+; Unix/Linux environmental variables can be 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 chan_dahdi.conf) to dial, i.e. group 2, and how to choose a channel to use
+; in the specified group. The four possible options are:
+;
+; g: select the lowest-numbered non-busy 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
+
+;FREENUMDOMAIN=mydomain.com                     ; domain to send on outbound
+                                                ; freenum calls (uses outbound-freenum
+                                                ; context)
+
+;
+; WARNING WARNING WARNING WARNING
+; If you load any other extension configuration engine, such as pbx_ael.so,
+; your global variables may be overridden by that file.  Please take care to
+; use only one location to set global variables, and you will likely save
+; yourself a ton of grief.
+; WARNING WARNING WARNING WARNING
+;
+; Any category other than "General" and "Globals" represent
+; extension contexts, which are collections of extensions.
+;
+; Extension names may be numbers, letters, or combinations
+; thereof. If an extension name is prefixed by a '_'
+; character, it is interpreted as a pattern rather than a
+; literal.  In patterns, some characters have special meanings:
+;
+;   X - any digit from 0-9
+;   Z - any digit from 1-9
+;   N - any digit from 2-9
+;   [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
+;   . - wildcard, matches anything remaining (e.g. _9011. matches
+;	anything starting with 9011 excluding 9011 itself)
+;   ! - wildcard, causes the matching process to complete as soon as
+;       it can unambiguously determine that no other matches are possible
+;
+; For example, the extension _NXXXXXX would match normal 7 digit dialings,
+; while _1NXXNXXXXXX would represent an area code plus phone number
+; preceded by a one.
+;
+; Each step of an extension is ordered by priority, which must always start
+; with 1 to be considered a valid extension.  The priority "next" or "n" means
+; the previous priority plus one, regardless of whether the previous priority
+; was associated with the current extension or not.  The priority "same" or "s"
+; means the same as the previously specified priority, again regardless of
+; whether the previous entry was for the same extension.  Priorities may be
+; immediately followed by a plus sign and another integer to add that amount
+; (most useful with 's' or 'n').  Priorities may then also have an alias, or
+; label, in parentheses after their name which can be used in goto situations.
+;
+; Contexts contain several lines, one for each step of each extension.  One may
+; include another context in the current one as well, optionally with a date
+; and time.  Included contexts are included in the order they are listed.
+; Switches may also be included within a context.  The order of matching within
+; a context is always exact extensions, pattern match extensions, includes, and
+; switches.  Includes are always processed depth-first.  So for example, if you
+; would like a switch "A" to match before context "B", simply put switch "A" in
+; an included context "C", where "C" is included in your original context
+; before "B".
+;
+;[context]
+;exten => someexten,{priority|label{+|-}offset}[(alias)],application(arg1,arg2,...)
+;
+; Timing list for includes is
+;
+;   <time range>,<days of week>,<days of month>,<months>[,<timezone>]
+;
+; Note that ranges may be specified to wrap around the ends.  Also, minutes are
+; fine-grained only down to the closest even minute.
+;
+;include => daytime,9:00-17:00,mon-fri,*,*
+;include => weekend,*,sat-sun,*,*
+;include => weeknights,17:02-8:58,mon-fri,*,*
+;
+; ignorepat can be used to instruct drivers to not cancel dialtone upon receipt
+; of a particular pattern.  The most commonly used example is of course '9'
+; like this:
+;
+;ignorepat => 9
+;
+; so that dialtone remains even after dialing a 9.  Please note that ignorepat
+; only works with channels which receive dialtone from the PBX, such as DAHDI,
+; Phone, and VPB.  Other channels, such as SIP and MGCP, which generate their
+; own dialtone and converse with the PBX only after a number is complete, are
+; generally unaffected by ignorepat (unless DISA or another method is used to
+; generate a dialtone after answering the channel).
+;
+
+;
+; Sample entries for extensions.conf
+;
+;
+[dundi-e164-canonical]
+;include => stdexten
+;
+; List canonical entries here
+;
+;exten => 12564286000,1,Gosub(6000,stdexten(IAX2/foo))
+;exten => 12564286000,n,Goto(default,s,1)	; exited Voicemail
+;exten => _125642860XX,1,Dial(IAX2/otherbox/${EXTEN:7})
+
+[dundi-e164-customers]
+;
+; If you are an ITSP or Reseller, list your customers here.
+;
+;exten => _12564286000,1,Dial(SIP/customer1)
+;exten => _12564286001,1,Dial(IAX2/customer2)
+
+[dundi-e164-via-pstn]
+;
+; If you are freely delivering calls to the PSTN, list them here
+;
+;exten => _1256428XXXX,1,Dial(DAHDI/G2/${EXTEN:7}) ; Expose all of 256-428
+;exten => _1256325XXXX,1,Dial(DAHDI/G2/${EXTEN:7}) ; Ditto for 256-325
+
+[dundi-e164-local]
+;
+; Context to put your dundi IAX2 or SIP user in for
+; full access
+;
+include => dundi-e164-canonical
+include => dundi-e164-customers
+include => dundi-e164-via-pstn
+
+[dundi-e164-switch]
+;
+; Just a wrapper for the switch
+;
+switch => DUNDi/e164
+
+[dundi-e164-lookup]
+;
+; Locally to lookup, try looking for a local E.164 solution
+; then try DUNDi if we don't have one.
+;
+include => dundi-e164-local
+include => dundi-e164-switch
+;
+; 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,
+; you have to goto "s" to execute that extension.
+;
+; Note: In old versions of Asterisk the PBX in some cases defaulted to
+; extension "s" when a given extension was wrong (like in AMI originate).
+; This is no longer the case.
+;
+; For wildcard matches, see above - all pattern matches start with
+; an underscore.
+exten => s,1,Goto(${ARG1},1)
+include => dundi-e164-lookup
+
+;
+; Here are the entries you need to participate in the IAXTEL
+; call routing system.  Most IAXTEL numbers begin with 1-700, but
+; there are exceptions.  For more information, and to sign
+; up, please go to www.gnophone.com or www.iaxtel.com
+;
+[iaxtel700]
+exten => _91700XXXXXXX,1,Dial(IAX2/${GLOBAL(IAXINFO)}@iaxtel.com/${EXTEN:1}@iaxtel)
+
+;
+; The SWITCH statement permits a server to share the dialplan with
+; another server. Use with care: Reciprocal switch statements are not
+; allowed (e.g. both A -> B and B -> A), and the switched server needs
+; to be on-line or else dialing can be severly delayed.
+;
+[iaxprovider]
+;switch => IAX2/user:[key]@myserver/mycontext
+
+[trunkint]
+;
+; International long distance through trunk
+;
+exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
+exten => _9011.,n,Dial(${GLOBAL(TRUNK)}/${FILTER(0-9,${EXTEN:${GLOBAL(TRUNKMSD)}})})
+
+[trunkld]
+;
+; Long distance context accessed through trunk
+;
+exten => _91NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN:1})
+exten => _91NXXNXXXXXX,n,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
+
+[trunklocal]
+;
+; Local seven-digit dialing accessed through trunk interface
+;
+exten => _9NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
+
+[trunktollfree]
+;
+; Long distance context accessed through trunk interface
+;
+exten => _91800NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
+exten => _91888NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
+exten => _91877NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
+exten => _91866NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
+
+[international]
+;
+; Master context for international long distance
+;
+ignorepat => 9
+include => longdistance
+include => trunkint
+
+[longdistance]
+;
+; Master context for long distance
+;
+ignorepat => 9
+include => local
+include => trunkld
+
+[local]
+;
+; Master context for local, toll-free, and iaxtel calls only
+;
+ignorepat => 9
+include => default
+include => trunklocal
+include => iaxtel700
+include => trunktollfree
+include => iaxprovider
+
+;Include parkedcalls (or the context you define in features conf)
+;to enable call parking.
+include => parkedcalls
+;
+; You can use an alternative switch type as well, to resolve
+; extensions that are not known here, for example with remote
+; IAX switching you transparently get access to the remote
+; Asterisk PBX
+;
+; switch => IAX2/user:password@bigserver/local
+;
+; An "lswitch" is like a switch but is literal, in that
+; variable substitution is not performed at load time
+; but is passed to the switch directly (presumably to
+; be substituted in the switch routine itself)
+;
+; lswitch => Loopback/12${EXTEN}@othercontext
+;
+; An "eswitch" is like a switch but the evaluation of
+; variable substitution is performed at runtime before
+; being passed to the switch routine.
+;
+; eswitch => IAX2/context@${CURSERVER}
+
+; The following two contexts are a template to enable the ability to dial
+; ISN numbers. For more information about what an ISN number is, please see
+; http://www.freenum.org.
+;
+; This is the dialing hook.  use:
+; include => outbound-freenum
+
+[outbound-freenum]
+; We'll add more digits as needed. The purpose is to dial things
+; like extension numbers at domains (ITAD number) so we're matching
+; on lengths of 1 through 6 prior to the separator (the asterisk [*])
+;
+exten => _X*X!,1,Goto(outbound-freenum2,${EXTEN},1)
+exten => _XX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
+exten => _XXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
+exten => _XXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
+exten => _XXXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
+exten => _XXXXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
+
+[outbound-freenum2]
+; This is the handler which performs the dialing logic. It is called
+; from the [outbound-freenum] context
+;
+exten => _X!,1,Verbose(2,Performing ISN lookup for ${EXTEN})
+same => n,Set(SUFFIX=${CUT(EXTEN,*,2-)})                                ; make sure the suffix is all digits as well
+same => n,GotoIf($["${FILTER(0-9,${SUFFIX})}" != "${SUFFIX}"]?fn-CONGESTION,1)
+                                                                        ; filter out bad characters per the README-SERIOUSLY.best-practices.txt document
+same => n,Set(TIMEOUT(absolute)=10800)
+same => n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)})     ; perform our lookup with freenum.org
+same => n,GotoIf($["${isnresult}" != ""]?from)
+same => n,Set(DIALSTATUS=CONGESTION)
+same => n,Goto(fn-CONGESTION,1)
+same => n(from),Set(__SIPFROMUSER=${CALLERID(num)})
+same => n,GotoIf($["${GLOBAL(FREENUMDOMAIN)}" = ""]?dial)               ; check if we set the FREENUMDOMAIN global variable in [global]
+same => n,Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)})                 ;    if we did set it, then we'll use it for our outbound dialing domain
+same => n(dial),Dial(SIP/${isnresult},40)
+same => n,Goto(fn-${DIALSTATUS},1)
+
+exten => fn-BUSY,1,Busy()
+
+exten => _f[n]-.,1,NoOp(ISN: ${DIALSTATUS})
+same => n,Congestion()
+
+[macro-trunkdial]
+;
+; Standard trunk dial macro (hangs up on a dialstatus that should
+; terminate call)
+;   ${ARG1} - What to dial
+;
+exten => s,1,Dial(${ARG1})
+exten => s,n,Goto(s-${DIALSTATUS},1)
+exten => s-NOANSWER,1,Hangup
+exten => s-BUSY,1,Hangup
+exten => _s-.,1,NoOp
+
+[stdexten]
+;
+; Standard extension subroutine:
+;   ${EXTEN} - Extension
+;   ${ARG1} - Device(s) to ring
+;   ${ARG2} - Optional context in Voicemail
+;
+; Note that the current version will drop through to the next priority in the
+; case of their pressing '#'.  This gives more flexibility in what do to next:
+; you can prompt for a new extension, or drop the call, or send them to a
+; general delivery mailbox, or...
+;
+; The use of the LOCAL() function is purely for convenience.  Any variable
+; initially declared as LOCAL() will disappear when the innermost Gosub context
+; in which it was declared returns.  Note also that you can declare a LOCAL()
+; variable on top of an existing variable, and its value will revert to its
+; previous value (before being declared as LOCAL()) upon Return.
+;
+exten => _X.,50000(stdexten),NoOp(Start stdexten)
+exten => _X.,n,Set(LOCAL(ext)=${EXTEN})
+exten => _X.,n,Set(LOCAL(dev)=${ARG1})
+exten => _X.,n,Set(LOCAL(cntx)=${ARG2})
+exten => _X.,n,Set(LOCAL(mbx)=${ext}${IF($[!${ISNULL(${cntx})}]?@${cntx})})
+exten => _X.,n,Dial(${dev},20)				; Ring the interface, 20 seconds maximum
+exten => _X.,n,Goto(stdexten-${DIALSTATUS},1)		; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
+
+exten => stdexten-NOANSWER,1,Voicemail(${mbx},u)	; If unavailable, send to voicemail w/ unavail announce
+exten => stdexten-NOANSWER,n,Return()			; If they press #, return to start
+
+exten => stdexten-BUSY,1,Voicemail(${mbx},b)		; If busy, send to voicemail w/ busy announce
+exten => stdexten-BUSY,n,Return()			; If they press #, return to start
+
+exten => _stde[x]te[n]-.,1,Goto(stdexten-NOANSWER,1)	; Treat anything else as no answer
+
+exten => a,1,VoicemailMain(${mbx})			; If they press *, send the user into VoicemailMain
+exten => a,n,Return()
+
+[stdPrivacyexten]
+;
+; Standard extension subroutine:
+;   ${ARG1} - Extension
+;   ${ARG2} - Device(s) to ring
+;   ${ARG3} - Optional DONTCALL context name to jump to (assumes the s,1 extension-priority)
+;   ${ARG4} - Optional TORTURE context name to jump to (assumes the s,1 extension-priority)`
+;   ${ARG5} - Context in voicemail (if empty, then "default")
+;
+; See above note in stdexten about priority handling on exit.
+;
+exten => _X.,60000(stdPrivacyexten),NoOp(Start stdPrivacyexten)
+exten => _X.,n,Set(LOCAL(ext)=${ARG1})
+exten => _X.,n,Set(LOCAL(dev)=${ARG2})
+exten => _X.,n,Set(LOCAL(dontcntx)=${ARG3})
+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(${dev},20,p)			; Ring the interface, 20 seconds maximum, call screening
+						; option (or use P for databased call _X.creening)
+exten => _X.,n,Goto(stdexten-${DIALSTATUS},1)		; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
+
+exten => stdexten-NOANSWER,1,Voicemail(${mbx},u)	; If unavailable, send to voicemail w/ unavail announce
+exten => stdexten-NOANSWER,n,NoOp(Finish stdPrivacyexten NOANSWER)
+exten => stdexten-NOANSWER,n,Return()			; If they press #, return to start
+
+exten => stdexten-BUSY,1,Voicemail(${mbx},b)		; If busy, send to voicemail w/ busy announce
+exten => stdexten-BUSY,n,NoOp(Finish stdPrivacyexten BUSY)
+exten => stdexten-BUSY,n,Return()			; If they press #, return to start
+
+exten => stdexten-DONTCALL,1,Goto(${dontcntx},s,1)	; Callee chose to send this call to a polite "Don't call again" script.
+
+exten => stdexten-TORTURE,1,Goto(${tortcntx},s,1)	; Callee chose to send this call to a telemarketer torture script.
+
+exten => _stde[x]te[n]-.,1,Goto(stdexten-NOANSWER,1)	; Treat anything else as no answer
+
+exten => a,1,VoicemailMain(${mbx})		; If they press *, send the user into VoicemailMain
+exten => a,n,Return
+
+[macro-page];
+;
+; Paging macro:
+;
+;       Check to see if SIP device is in use and DO NOT PAGE if they are
+;
+;   ${ARG1} - Device to page
+
+exten => s,1,ChanIsAvail(${ARG1},s)			; s is for ANY call
+exten => s,n,GoToIf($[${AVAILSTATUS} = "1"]?autoanswer:fail)
+exten => s,n(autoanswer),Set(_ALERT_INFO="RA")			; This is for the PolyComs
+exten => s,n,SIPAddHeader(Call-Info: Answer-After=0)	; This is for the Grandstream, Snoms, and Others
+exten => s,n,NoOp()					; Add others here and Post on the Wiki!!!!
+exten => s,n,Dial(${ARG1})
+exten => s,n(fail),Hangup
+
+
+[demo]
+include => stdexten
+;
+; We start with what to do when a call first comes in.
+;
+exten => s,1,Wait(1)			; Wait a second, just for fun
+exten => s,n,Answer			; Answer the line
+exten => s,n,Set(TIMEOUT(digit)=5)	; Set Digit Timeout to 5 seconds
+exten => s,n,Set(TIMEOUT(response)=10)	; Set Response Timeout to 10 seconds
+exten => s,n(restart),BackGround(demo-congrats)	; Play a congratulatory message
+exten => s,n(instruct),BackGround(demo-instruct)	; Play some instructions
+exten => s,n,WaitExten			; Wait for an extension to be dialed.
+
+exten => 2,1,BackGround(demo-moreinfo)	; Give some more information.
+exten => 2,n,Goto(s,instruct)
+
+exten => 3,1,Set(CHANNEL(language)=fr)		; Set language to french
+exten => 3,n,Goto(s,restart)		; Start with the congratulations
+
+exten => 1000,1,Goto(default,s,1)
+;
+; We also create an example user, 1234, who is on the console and has
+; voicemail, etc.
+;
+exten => 1234,1,Playback(transfer,skip)		; "Please hold while..."
+					; (but skip if channel is not up)
+exten => 1234,n,Gosub(${EXTEN},stdexten(${GLOBAL(CONSOLE)}))
+exten => 1234,n,Goto(default,s,1)		; exited Voicemail
+
+exten => 1235,1,Voicemail(1234,u)		; Right to voicemail
+
+exten => 1236,1,Dial(Console/dsp)		; Ring forever
+exten => 1236,n,Voicemail(1234,b)		; Unless busy
+
+;
+; # for when they're done with the demo
+;
+exten => #,1,Playback(demo-thanks)	; "Thanks for trying the demo"
+exten => #,n,Hangup			; Hang them up.
+
+;
+; A timeout and "invalid extension rule"
+;
+exten => t,1,Goto(#,1)			; If they take too long, give up
+exten => i,1,Playback(invalid)		; "That's not valid, try again"
+
+;
+; Create an extension, 500, for dialing the
+; Asterisk demo.
+;
+exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
+exten => 500,n,Dial(IAX2/guest@pbx.digium.com/s@default)	; Call the Asterisk demo
+exten => 500,n,Playback(demo-nogo)	; Couldn't connect to the demo site
+exten => 500,n,Goto(s,6)		; Return to the start over message.
+
+;
+; Create an extension, 600, for evaluating echo latency.
+;
+exten => 600,1,Playback(demo-echotest)	; Let them know what's going on
+exten => 600,n,Echo			; Do the echo test
+exten => 600,n,Playback(demo-echodone)	; Let them know it's over
+exten => 600,n,Goto(s,6)		; Start over
+
+;
+;	You can use the Macro Page to intercom a individual user
+exten => 76245,1,Macro(page,SIP/Grandstream1)
+; or if your peernames are the same as extensions
+exten => _7XXX,1,Macro(page,SIP/${EXTEN})
+;
+;
+; System Wide Page at extension 7999
+;
+exten => 7999,1,Set(TIMEOUT(absolute)=60)
+exten => 7999,2,Page(Local/Grandstream1@page&Local/Xlite1@page&Local/1234@page/n,d)
+
+; Give voicemail at extension 8500
+;
+exten => 8500,1,VoicemailMain
+exten => 8500,n,Goto(s,6)
+;
+; Here's what a phone entry would look like (IXJ for example)
+;
+;exten => 1265,1,Dial(Phone/phone0,15)
+;exten => 1265,n,Goto(s,5)
+
+;
+;	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
+;	Local/{peername}@page
+;
+[page]
+exten => _X.,1,Macro(page,SIP/${EXTEN})
+
+;[mainmenu]
+;
+; Example "main menu" context with submenu
+;
+;exten => s,1,Answer
+;exten => s,n,Background(thanks)		; "Thanks for calling press 1 for sales, 2 for support, ..."
+;exten => s,n,WaitExten
+;exten => 1,1,Goto(submenu,s,1)
+;exten => 2,1,Hangup
+;include => default
+;
+;[submenu]
+;exten => s,1,Ringing					; Make them comfortable with 2 seconds of ringback
+;exten => s,n,Wait,2
+;exten => s,n,Background(submenuopts)	; "Thanks for calling the sales department.  Press 1 for steve, 2 for..."
+;exten => s,n,WaitExten
+;exten => 1,1,Goto(default,steve,1)
+;exten => 2,1,Goto(default,mark,2)
+
+[public]
+;
+; ATTENTION: If your Asterisk is connected to the internet and you do
+; not have allowguest=no in sip.conf, everybody out there may use your
+; public context without authentication.  In that case you want to
+; double check which services you offer to the world.
+;
+include => demo
+
+[default]
+;
+; By default we include the demo.  In a production system, you
+; probably don't want to have the demo there.
+;
+include => demo
+
+;
+; An extension like the one below can be used for FWD, Nikotel, sipgate etc.
+; Note that you must have a [sipprovider] section in sip.conf
+;
+;exten => _41X.,1,Dial(SIP/${FILTER(0-9,${EXTEN:2})}@sipprovider,,r)
+
+; Real extensions would go here. Generally you want real extensions to be
+; 4 or 5 digits long (although there is no such requirement) and start with a
+; single digit that is fairly large (like 6 or 7) so that you have plenty of
+; room to overlap extensions and menu options without conflict.  You can alias
+; them with names, too, and use global variables
+
+;exten => 6245,hint,SIP/Grandstream1&SIP/Xlite1(Joe Schmoe) ; Channel hints for presence
+;exten => 6245,1,Dial(SIP/Grandstream1,20,rt)	; permit transfer
+;exten => 6245,n(dial),Dial(${HINT},20,rtT)	; Use hint as listed
+;exten => 6245,n,Voicemail(6245,u)		; Voicemail (unavailable)
+;exten => 6245,s+1,Hangup			; s+1, same as n
+;exten => 6245,dial+101,Voicemail(6245,b)	; Voicemail (busy)
+;exten => 6361,1,Dial(IAX2/JaneDoe,,rm)		; ring without time limit
+;exten => 6389,1,Dial(MGCP/aaln/1@192.168.0.14)
+;exten => 6390,1,Dial(JINGLE/caller/callee) ; Dial via jingle using labels
+;exten => 6391,1,Dial(JINGLE/asterisk@digium.com/mogorman@astjab.org) ;Dial via jingle using asterisk as the transport and calling mogorman.
+;exten => 6394,1,Dial(Local/6275/n)		; this will dial ${MARK}
+
+;exten => 6275,1,Gosub(${EXTEN},stdexten(${MARK}))
+						; assuming ${MARK} is something like DAHDI/2
+;exten => 6275,n,Goto(default,s,1)		; exited Voicemail
+;exten => mark,1,Goto(6275,1)			; alias mark to 6275
+;exten => 6536,1,Gosub(${EXTEN},stdexten(${WIL}))
+						; Ditto for wil
+;exten => 6536,n,Goto(default,s,1)		; exited Voicemail
+;exten => wil,1,Goto(6236,1)
+
+;If you want to subscribe to the status of a parking space, this is
+;how you do it. Subscribe to extension 6600 in sip, and you will see
+;the status of the first parking lot with this extensions' help
+;exten => 6600,hint,park:701@parkedcalls
+;exten => 6600,1,noop
+;
+; You can also monitor the status of a queue by providing a hint for a
+; particular queue name.
+;exten => 8502,hint,Queue:markq
+;exten => 8502,1,Queue(markq)
+;
+
+;To subscribe to the availability of a free member in the 'markq' queue.
+;Note: '_avail' is added to the QueueName
+;exten => 8501,hint,Queue:markq_avail
+;exten => 8501,1,Queue(markq)
+
+; Some other handy things are an extension for checking voicemail via
+; voicemailmain
+;
+;exten => 8500,1,VoicemailMain
+;exten => 8500,n,Hangup
+;
+; Or a conference room (you'll need to edit meetme.conf to enable this room)
+;
+;exten => 8600,1,Meetme(1234)
+;
+; Or playing an announcement to the called party, as soon it answers
+;
+;exten = 8700,1,Dial(${MARK},30,A(/path/to/my/announcemsg))
+;
+
+; example of a compartmentalized company called "acme"
+;
+; this is the context that your incoming IAX/SIP trunk dumps you in...
+;[acme-incoming]
+;exten => s,1,Wait(1)
+;exten => s,n,Answer()
+;exten => s,n(menu),Playback(acme/vm-brief-menu)
+;exten => s,n(exten),Background(vm-enter-num-to-call)
+;exten => s,n,WaitExten(5)
+;exten => s,n(goodbye),Playback(vm-goodbye)
+;exten => s,n(end),Hangup()
+;
+;include  => acme-extens
+;
+;exten => i,1,Playback(vm-invalid)
+;exten => i,n,Goto(s,exten)			; optionally, transfer to operator
+;
+;exten => t,1,Goto(s,goodbye)
+;
+; this is the context our internal SIP hardphones use (see sip.conf)
+;
+;[acme-internal]
+;exten => s,1,Answer()
+;exten => s,n(exten),Background(vm-enter-num-to-call)
+;exten => s,n,WaitExten(5)
+;exten => s,n(goodbye),Playback(vm-goodbye)
+;exten => s,n(end),Hangup()
+;
+;include => trunkint
+;include => trunkld
+;include => trunklocal
+;
+;include => acme-extens
+;
+; you can test what your system sounds like to outside callers by dialing this
+;exten => 777,1,DISA(no-password,acme-incoming)
+;
+; grouping of acme's extensions... never used directly, always included.
+;
+;[acme-extens]
+;include => stdexten
+;exten => 111,1,Gosub(111,stdexten(SIP/pete_1,acme))
+;exten => 111,n,Goto(s,exten)
+;
+;exten => 112,1,Gosub(112,stdexten(SIP/nancy_1,acme))
+;exten => 112,n,Goto(s,end)
+;
+; end of acme example
+
+;
+; Time context: you can patch this in via the following.
+;
+; [acme-internal]
+; ...
+; exten => 777,1,Gosub(time)
+; exten => 777,n,Hangup()
+;
+; ...
+; include => time
+;
+; Note: if you're geographically spread out, you can have SIP extensions
+; specify their own local timezone in sip.conf as:
+;
+; [boi]
+; type=friend
+; context=acme-internal
+; callerid="Boise Ofc. <2083451111>"
+; ...
+; ; use system-wide default timezone of MST7MDT
+;
+; [lws]
+; type=friend
+; context=acme-internal
+; callerid="Lewiston Ofc. <2087431111>"
+; ...
+; setvar=timezone=PST8PDT
+;
+; "timezone" isn't a 'reserved' name in any way, and other places where
+; the timezone is significant (e.g. calls to "SayUnixTime()", etc) will
+; require modification as well.  Note that voicemail.conf already has
+; a mechanism for timezones.
+;
+
+[time]
+exten => _X.,30000(time),NoOp(Time: ${EXTEN} ${timezone})
+exten => _X.,n,Wait(0.25)
+exten => _X.,n,Answer()
+; the amount of delay is set for English; you may need to adjust this time
+; for other languages if there's no pause before the synchronizing beep.
+exten => _X.,n,Set(FUTURETIME=$[${EPOCH} + 12])
+exten => _X.,n,SayUnixTime(${FUTURETIME},Zulu,HNS)
+exten => _X.,n,SayPhonetic(z)
+; use the timezone associated with the extension (sip only), or system-wide
+; default if one hasn't been set.
+exten => _X.,n,SayUnixTime(${FUTURETIME},${timezone},HNS)
+exten => _X.,n,Playback(spy-local)
+exten => _X.,n,WaitUntil(${FUTURETIME})
+exten => _X.,n,Playback(beep)
+exten => _X.,n,Return()
+
+;
+; ANI context: use in the same way as "time" above
+;
+
+[ani]
+exten => _X.,40000(ani),NoOp(ANI: ${EXTEN})
+exten => _X.,n,Wait(0.25)
+exten => _X.,n,Answer()
+exten => _X.,n,Playback(vm-from)
+exten => _X.,n,SayDigits(${CALLERID(ani)})
+exten => _X.,n,Wait(1.25)
+exten => _X.,n,SayDigits(${CALLERID(ani)})	; playback again in case of missed digit
+exten => _X.,n,Return()
+
+; For more information on applications, just type "core show applications" at your
+; friendly Asterisk CLI prompt.
+;
+; "core show application <command>" will show details of how you
+; 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.

asterisk/extensions.lua

@@ -0,0 +1,241 @@
+
+
+CONSOLE = "Console/dsp" -- Console interface for demo
+--CONSOLE = "DAHDI/1"
+--CONSOLE = "Phone/phone0"
+
+IAXINFO = "guest"       -- IAXtel username/password
+--IAXINFO = "myuser:mypass"
+
+TRUNK = "DAHDI/G2"
+TRUNKMSD = 1
+-- TRUNK = "IAX2/user:pass@provider"
+
+
+--
+-- Extensions are expected to be defined in a global table named 'extensions'.
+-- The 'extensions' table should have a group of tables in it, each
+-- representing a context.  Extensions are defined in each context.  See below
+-- for examples.
+--
+-- Extension names may be numbers, letters, or combinations thereof. If
+-- an extension name is prefixed by a '_' character, it is interpreted as
+-- a pattern rather than a literal.  In patterns, some characters have
+-- special meanings:
+--
+--   X - any digit from 0-9
+--   Z - any digit from 1-9
+--   N - any digit from 2-9
+--   [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
+--   . - wildcard, matches anything remaining (e.g. _9011. matches
+--       anything starting with 9011 excluding 9011 itself)
+--   ! - wildcard, causes the matching process to complete as soon as
+--       it can unambiguously determine that no other matches are possible
+--
+-- For example the extension _NXXXXXX would match normal 7 digit
+-- dialings, while _1NXXNXXXXXX would represent an area code plus phone
+-- number preceded by a one.
+--
+-- If your extension has special characters in it such as '.' and '!' you must
+-- explicitly make it a string in the tabale definition:
+--
+--   ["_special."] = function;
+--   ["_special!"] = function;
+--
+-- 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.
+--
+-- One context may be included in another context using the 'includes'
+-- 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.
+--
+--   v = channel.var_name
+--   v = channel["var_name"]
+--   v.value
+--   v:get()
+--
+--   channel.var_name = "value"
+--   channel["var_name"] = "value"
+--   v:set("value")
+--
+--   channel.func_name(1,2,3):set("value")
+--   value = channel.func_name(1,2,3):get()
+--
+--   channel["func_name(1,2,3)"]:set("value")
+--   channel["func_name(1,2,3)"] = "value"
+--   value = channel["func_name(1,2,3)"]:get()
+--
+-- Note the use of the ':' operator to access the get() and set()
+-- methods.
+--
+-- 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.
+--
+--    app.Dial("DAHDI/1")
+--    app.dial("DAHDI/1")
+--    app["dial"]("DAHDI/1")
+--
+-- More examples can be found below.
+--
+-- An autoservice is automatically run while lua code is executing.  The
+-- autoservice can be stopped and restarted using the autoservice_stop() and
+-- autoservice_start() functions.  The autservice should be running before
+-- starting long running operations. The autoservice will automatically be
+-- stopped before executing applications and dialplan functions and will be
+-- restarted afterwards.  The autoservice_status() function can be used to
+-- check the current status of the autoservice and will return true if an
+-- autoservice is currently running.
+--
+-- Note about naming conflicts:
+-- Lua allows you to refer to table entries using the '.' notation,
+-- I.E. app.goto(something), only if the entry doesn't conflict with an Lua
+-- reserved word.  In the 'goto' example, with Lua 5.1 or earlier, 'goto' is
+-- not a reserved word so you'd be calling the Asterisk dialplan application
+-- 'goto'.  Lua 5.2 however, introduced the 'goto' control statement which
+-- makes 'goto' a reserved word.  This casues the interpreter to fail parsing
+-- the file and pbx_lua.so will fail to load.  The same applies to any use of
+-- Lua tables including extensions, channels and any tables you create.
+--
+-- There are two ways around this:  Since Lua is case-sensitive, you can use
+-- capitalized names, I.E. app.Goto(something) to refer to the Asterisk apps,
+-- functions, etc. Or you can use the full syntax, I.E. app["goto"](something).
+-- Both syntaxes are backwards compatible with earlier Lua versions.  To make
+-- your Lua dialplans easier to maintain and to reduce the chance of future
+-- conflicts you may want to use the app["goto"](something) syntax for all
+-- table accesses.
+--
+
+function outgoing_local(c, e)
+	app.dial("DAHDI/1/" .. e, "", "")
+end
+
+function demo_instruct()
+	app.background("demo-instruct")
+	app.waitexten()
+end
+
+function demo_congrats()
+	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()
+
+	channel.TIMEOUT("digit"):set(5)
+	channel.TIMEOUT("response"):set(10)
+	-- app.set("TIMEOUT(digit)=5")
+	-- app.set("TIMEOUT(response)=10")
+
+	demo_congrats(context, exten)
+end
+
+function demo_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()
+--	See the naming conflict note above.
+			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"};
+	};
+
+	public = {
+		-- ATTENTION: If your Asterisk is connected to the internet and you do
+		-- not have allowguest=no in sip.conf, everybody out there may use your
+		-- public context without authentication.  In that case you want to
+		-- double check which services you offer to the world.
+		--
+		include = {"demo"};
+	};
+
+	["local"] = {
+		["_NXXXXXX"] = outgoing_local;
+	};
+}
+
+hints = {
+	demo = {
+		[1000] = "SIP/1000";
+		[1001] = "SIP/1001";
+	};
+
+	default = {
+		["1234"] = "SIP/1234";
+	};
+}
+

asterisk/extensions_minivm.conf

@@ -0,0 +1,159 @@
+; MINI-VOICEMAIL dialplan example
+; ---------------------------------------------------------------------------------------
+; ASTERISK_FILE_VERSION(__FILE__, "$Revision: 197535 $")
+;
+;
+; This is an example on how to use the Mini-Voicemail system to build
+; voicemail systems.
+;
+;.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-
+; A macro to test the MINIVMACCOUNT dialplan function
+; 	Currently, accountcode and pincode is not used in the application
+; 	They where added to be used in dialplan scripting
+;
+;
+[macro-minivmfunctest]
+exten => s,1,set(account=${ARGV1})
+exten => minivm,n,verbose(1,-------------------- Minivm Function test - Accoutn ${account}  -------------)
+exten => s,n,verbose(1,---- Has account:       ${MINIVMACCOUNT(${account}:hasaccount)})
+exten => s,n,verbose(1,---- Fullname:          ${MINIVMACCOUNT(${account}:fullname)})
+exten => s,n,verbose(1,---- Email:             ${MINIVMACCOUNT(${account}:email)})
+exten => s,n,verbose(1,---- Pager:             ${MINIVMACCOUNT(${account}:pager)})
+exten => s,n,verbose(1,---- E-mail template:   ${MINIVMACCOUNT(${account}:etemplate)})
+exten => s,n,verbose(1,---- Pager template:    ${MINIVMACCOUNT(${account}:ptemplate)})
+exten => s,n,verbose(1,---- Account code:      ${MINIVMACCOUNT(${account}:accountcode)})
+exten => s,n,verbose(1,---- Path:              ${MINIVMACCOUNT(${account}:path)})
+exten => s,n,verbose(1,---- Pincode:           ${MINIVMACCOUNT(${account}:pincode)})
+exten => s,n,verbose(1,---- Time zone:         ${MINIVMACCOUNT(${account}:timezone)})
+exten => s,n,verbose(1,---- Language:          ${MINIVMACCOUNT(${account}:language)})
+; This requires setvar=customerclass=gold in the account configuration
+exten => s,n,verbose(1,---- Var:customerclass: ${MINIVMACCOUNT(${account}:customerclass)})
+
+[minivm-scenario1]
+; minivmtest tests the dialplan function MINIVMACCOUNT
+; Check the output in the console with verbose set
+exten => minivmtest,1,answer
+exten => minivmtest,n,wait(0.5)
+exten => minivmtest,n,set(ACCOUNT=do-not-spam-me@example.com)
+exten => minivmtest,n,macro(minivmfunctest, ${ACCOUNT})
+exten => minivmtest,n,playback(beep)
+exten => minivmtest,n,hangup
+
+;.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-
+; "minivm" tests a full scenario
+; Remember that users may hangup
+; This works both for users with accounts in minivm.conf and by just giving an e-mail address
+; without configuring an account
+exten => minivm,1,answer
+exten => minivm,n,wait(0.5)	; Wait for Voip channels to settle
+exten => minivm,n,set(account=oej@example.com)
+exten => minivm,n,noop(------------------------------------------- Minivm Greet   -------------)
+exten => minivm,n,minivmgreet(${account})
+exten => minivm,n,verbose(1,-- MINIVM_GREET_STATUS = ${MINIVM_GREET_STATUS} )
+exten => minivm,n,noop(------------------------------------------- Minivm Record  -------------)
+exten => minivm,n,minivmRecord(${account},b)
+exten => minivm,n,goto(minivmcleanup,1)
+
+; Cleanup after recording or hangup
+exten => minivmcleanup,1,noop(------------------------------------------- Minivm Notify  -------------)
+;Increment voicemail counter with 1. The counter will be used in the e-mail message
+;and in the filename
+exten => minivmcleanup,n,gotoif($[${MINIVM_RECORD_STATUS} != SUCCESS]?minivmrecordfailure,1)
+exten => minivmcleanup,n,set(MINIVMCOUNTER(${account}:voicemailcounter:inc)=1)
+exten => minivmcleanup,n,set(MVM_COUNTER = ${MINIVMCOUNTER(${account}:voicemailcounter)})
+exten => minivmcleanup,n,minivmNotify(${account})
+exten => minivmcleanup,n,verbose(1,-- MINIVM_NOTIFY_STATUS = ${MINIVM_NOTIFY_STATUS} )
+; Now, clean up after sending voicemail
+exten => minivmcleanup,n,noop(------------------------------------------- Minivm Delete  -------------)
+exten => minivmcleanup,n,minivmdelete()
+exten => minivmcleanup,n,verbose(1,-- MINIVM_DELETE_STATUS = ${MINIVM_DELETE_STATUS} )
+
+;Recording failed
+exten => minivmrecordfailure,1,playback(vm-sorry)
+exten => minivmrecordfailure,n,wait(1)
+exten => minivmrecordfailure,n,hangup
+
+; If the user hangs up during the recording, we need to clean up
+; And send notifications
+exten => h,1,gotoif($[x${MINIVM_DELETE_STATUS} != x] ?h,stop)
+exten => h,n,noop(------------------------------------------- HANGUP during voicemail recording   -------------)
+exten => h,n,goto(minivmcleanup,1)
+exten => h,n(stop),noop(---Minivm DONE----)
+
+;.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-
+; Extension to record a greeting message
+; Call this like:
+;	macro(recordgreetings,alice@atlanta.example.com)
+;
+[macro-recordgreetings]
+exten => s,1,answer
+exten => s,n,wait(0.5)
+exten => s,n,set(account=${ARGV1])
+; This file give extra options not available here, needs to be edited
+; Change of password does not work
+exten => s,n(menu),background(vm-options)
+exten => 1,1,setvar(option=u)
+exten => 1,n,macro(minivmrec,${account},${option})
+exten => 1,n,goto(menu)
+exten => 2,1,setvar(option=b)
+exten => 2,n,macro(minivmrec,${account},${option})
+exten => 2,n,goto(menu)
+exten => 3,1,setvar(option=n)
+exten => 3,n,macro(minivmrec,${account},${option})
+exten => 3,n,goto(menu)
+exten => 4,1,setvar(option=t)
+exten => 4,n,macro(minivmrec,${account},${option})
+exten => 4,n,goto(menu)
+exten => *,1,playback(vm-thankyou)
+exten => *,n,wait(1)
+exten => *,n,hangup
+
+exten => i,1,playback(invalid)
+exten => i,n,goto(menu)
+
+[macro-minivmrec]
+exten => s,1,gotoif(${MINIVMACCOUNT(${account}:hasaccount)}?record)
+; Account is not configured in minivm.conf or realtime
+; Phony message, add something useful here
+exten => s,n,playback(privacy-incorrect)
+exten => s,n,macroreturn
+exten => record,1,minivmappmess(${ARGV1},${ARGV2})
+exten => record,n,noop(Recording status: ${MINIVM_APPMESS_STATUS})
+exten => record,n,macroreturn
+
+;.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-
+; To set a counter and use a template for voicemail to users without acounts
+; use something like this
+;
+; email address is in the "account" channel variable. Set from ast_db or a script
+; based on called ID
+
+exten => sendvoicemail,1,answer
+exten => sendvoicemail,n,wait(0.5)
+exten => sendvoicemail,n,set(domain=${CUT(account,@,2)})
+exten => sendvoicemail,n,set(country=${CUT(domain,.,2)})
+exten => sendvoicemail,n,minivmgreet(${account})
+exten => sendvoicemail,n,minivmRecord(${account},b)
+exten => sendvoicemail,n,goto(sendvmcleanup)
+
+exten => sendvmcleanup,1,gotoif($[${MINIVM_RECORD_STATUS} != SUCCESS]?done)
+; The counter is set in the domain directory, so we don't create one directory per user
+; The counter has the email in the name of the counter, increase it
+; Set the MVM_COUNTER variable that we use in the template
+exten => sendvmcleanup,n,set(MINIVMCOUNTER(${account}:voicemailcounter:inc)=1)
+exten => sendvmcleanup,n,set(MVM_COUNTER = ${MINIVMCOUNTER(${account}:voicemailcounter)})
+; Increase a domain counter too, to see how many voicemails are sent to this domain
+; This is just for statistics
+exten => sendvmcleanup,n,set(MINIVMCOUNTER(${domain}:${domain}-all:inc) = 1)
+
+; Send voicemail in e-mail with country-specific template
+; The template need to be defined in minivm.conf
+;
+exten => sendvmcleanup,n,minivmNotify(${account}, ${country}_email)
+exten => sendvmcleanup,n,minivmDelete()
+
+exten => sendvmcleanup,n(done),wait(0.5)
+exten => sendvmcleanup,n,hangup
+
+exten => h,1,gotoif($[${MINIVM_RECORD_STATUS} = SUCCESS]?sendvmcleanup,1))
+

asterisk/features.conf

@@ -0,0 +1,238 @@
+;
+; Sample Call Features (parking, transfer, etc) configuration
+;
+
+[general]
+
+; OPTIONS THAT CAN ONLY BE SET IN THE GENERAL CONTEXT
+
+;featuredigittimeout = 1000     ; Max time (ms) between digits for
+                                ; feature activation  (default is 1000 ms)
+
+; Pickup Options
+;
+;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)
+
+; Transfer Options
+;
+;transferdigittimeout => 3      ; Number of seconds to wait between digits when transferring a call
+                                ; (default is 3 seconds)
+;xfersound = beep               ; to indicate an attended transfer is complete
+;xferfailsound = beeperr        ; to indicate a failed transfer
+;atxfernoanswertimeout = 15     ; Timeout for answer on attended transfer default is 15 seconds.
+;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.
+;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.
+                                ; For atxferdropcall=no to work properly, you also need to
+                                ; define ATXFER_NULL_TECH in main/features.c.  The reason the
+                                ; code is not enabled by default is spelled out in the comment
+                                ; block near the top of main/features.c describing ATXFER_NULL_TECH.
+
+; Parking Options
+;
+; These options apply to all parking lots, including the default lot defined in
+; the general context.
+;
+;courtesytone = beep            ; Sound file to play to when someone picks up a parked call
+                                ; and also when the Touch Monitor is activated/deactivated.
+                                ; Default is no tone.
+;parkedplay = caller            ; Who to play courtesytone to when picking up a parked call.
+                                ; One of: parked, caller, both  (default is caller)
+;parkeddynamic = yes            ; Enables dynamically created parkinglots. (default is no)
+;adsipark = yes                 ; if you want ADSI parking announcements
+
+
+; OPTIONS THAT CAN BE SET PER PARKING LOT
+;
+; If these options are set in the general context, they will *only* apply to the default
+; parking lot. If set in any other lot, they will *only* apply to that lot.
+
+parkext => 700                  ; What extension to dial to park. This option can take any alphanumeric string.
+                                ; (default is 700 for all lots)
+;parkext_exclusive=yes          ; Specify that the parkext created for this parking lot
+                                ; will only access this parking lot. (default is no for all lots)
+parkpos => 701-750              ; What extensions to park calls on.
+                                ; (default is 701-750 for the default lot and blank for custom lots)
+                                ; These need to be numeric, as Asterisk starts from the start position
+                                ; and increments with one for the next parked call. Hence, leading zeros
+                                ; and non-numerical characters will be ignored.
+context => parkedcalls          ; Which context parked calls are in (default is parkedcalls for the default lot and blank for custom lots)
+;parkinghints = no              ; Add hints priorities automatically for parking slots (default is no for all lots).
+;parkingtime => 45              ; Number of seconds a call can be parked before returning.
+                                ; (default is 45 for all lots)
+;parkedcalltransfers = caller   ; Enables or disables DTMF based transfers when picking up a parked call.
+                                ; one of: callee, caller, both, no (default is no for all lots)
+;parkedcallreparking = caller   ; Enables or disables DTMF based parking when picking up a parked call.
+                                ; one of: callee, caller, both, no (default is no for all lots)
+;parkedcallhangup = caller      ; Enables or disables DTMF based hangups when picking up a parked call.
+                                ; one of: callee, caller, both, no (default is no for all lots)
+;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 for all lots)
+;findslot => next               ; Continue to the 'next' free parking space.
+                                ; (Default is 'first' for all lots)
+;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.
+                                ; (Default is 'default' for the default parking lot. Non-default parkinglots
+                                ; don't specify a class so the class is picked by the channel put on hold
+                                ; which has the 'default' class by default.)
+;comebacktoorigin = yes         ; Setting this option configures the behavior of call parking when the
+                                ; parked call times out (See the parkingtime option).  The default value is 'yes'.
+                                ;
+                                ; 'yes' - When the parked call times out, attempt to send the call back to the peer
+                                ;         that parked this call. This is done by saving off the name of the channel
+                                ;         that parked the call. The call will return to the context 'park-dial' and
+                                ;         an extension created based on the name of the channel that originally parked
+                                ;         the call.  This extension will be created automatically to do a Dial() to the
+                                ;         device that originally parked the call for comebacktodialtime seconds. If the
+                                ;         call is not answered, the call will proceed to 'park-dial,t,1'.
+                                ;
+                                ; 'no'  - This option is useful for performing custom dialplan functionality prior to
+                                ;         sending the call back to the extension that initially parked the call, or to
+                                ;         an entirely different destination.
+                                ;
+                                ;         When the parked call times out, send it back to the dialplan.  The location
+                                ;         will be defined by the comebackcontext option. The extension will be built from
+                                ;         the saved channel name that parked the call. For example, if a SIP peer named
+                                ;         '0004F2040001' parked this call, the extension will be 'SIP_0004F2040001'.
+                                ;         (Note that an underscore is used here because the '/' character has a special
+                                ;         meaning in extension names for CallerID matching.)  If this extension does not
+                                ;         exist, the call will be sent to the 's' extension, instead.  Finally, if the 's'
+                                ;         extension of 'parkedcallstimeout' does not exist, the call will fall back to the
+                                ;         's' extension of the 'default' context.
+                                ;
+                                ;         Additionally, in this example an extension of 'SIP_0004F2040001' will be
+                                ;         created in the 'park-dial' context.  This extension will be set up to do a
+                                ;         Dial() to 'SIP/0004F2040001'.
+                                ;
+                                ;         During the timeout procedure, the following variables are set
+                                ;         PARKINGSLOT - extension that the call was parked in prior to timing out
+                                ;         PARKEDLOT - name of the lot that the call was parked in prior to timing out
+                                ;         PARKER - name of the device that parked the call
+
+;comebackdialtime = 30          ; When a parked call times out, this is the number of seconds to dial the device that
+                                ; originally parked the call.  It is also available as a channel variable COMEBACKDIALTIME
+                                ; after a parked call has timed out.
+                                ; The default value is 30 seconds.
+;comebackcontext = parkedcallstimeout
+                                ; The context a timed out call will return to if comebcktoorigin=no.
+                                ; The default value is 'parkedcallstimeout'.
+
+
+; EXAMPLE NON-DEFAULT PARKING LOT DEFINITION
+;
+; You can set parkinglot with the CHANNEL dialplan function
+; or by setting 'parkinglot' directly in the channel configuration file.
+;
+; (Note: Leading '0's and any non-numerical characters on parkpos extensions
+;  will be ignored.  Parkext on the other hand can be any string.)
+;
+;[parkinglot_edvina]
+;context => edvinapark
+;parkext => 799
+;parkpos => 800-850
+;findslot => next
+;comebacktoorigin = no
+;comebackdialtime = 90
+;comebackcontext = edvinapark-timeout ; Make sure you create the context!
+;parkedmusicclass = edvina
+
+; EXAMPLE DTMF FEATURE MAP
+;
+; 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
+; chan_local in combination with Answer to accomplish it.
+
+[featuremap]
+;blindxfer => #1                ; Blind transfer  (default is #) -- Make sure to set the T and/or t option in the Dial() or Queue() app call!
+;disconnect => *0               ; Disconnect  (default is *) -- Make sure to set the H and/or h option in the Dial() or Queue() app call!
+                                ; Note that Agents via chan_agent use * to disconnect a call by default. If you are wanting to use * 
+                                ; to abort a transfer you will either need to change the disconnect code here or set endcall to no in agents.conf 
+;automon => *1                  ; One Touch Record a.k.a. Touch Monitor -- Make sure to set the W and/or w option in the Dial() or Queue() app call!
+;atxfer => *2                   ; Attended transfer  -- Make sure to set the T and/or t option in the Dial() or Queue()  app call!
+;parkcall => #72                ; Park call (one step parking)  -- Make sure to set the K and/or k option in the Dial() app call!
+;automixmon => *3               ; One Touch Record a.k.a. Touch MixMonitor -- Make sure to set the X and/or x option in the Dial() or Queue() app call!
+
+; EXAMPLE DYNAMIC FEATURES APPLICATION MAP
+;
+
+[applicationmap]
+; Note that the DYNAMIC_FEATURES channel variable must be set to use the features
+; defined here.  The value of DYNAMIC_FEATURES should be the names of the features
+; to allow the channel to use separated by '#'.  For example:
+;
+;    Set(__DYNAMIC_FEATURES=myfeature1#myfeature2#myfeature3)
+;
+; (Note: The two leading underscores allow these feature settings to be set on
+;  on the outbound channels, as well.  Otherwise, only the original channel
+;  will have access to these features.)
+;
+; The syntax for declaring a dynamic feature is any of the following:
+;
+;<FeatureName> => <DTMF_sequence>,<ActivateOn>[/<ActivatedBy>],<Application>[,<AppArguments>[,MOH_Class]]
+;<FeatureName> => <DTMF_sequence>,<ActivateOn>[/<ActivatedBy>],<Application>[,"<AppArguments>"[,MOH_Class]]
+;<FeatureName> => <DTMF_sequence>,<ActivateOn>[/<ActivatedBy>],<Application>([<AppArguments>])[,MOH_Class]
+
+;
+;  FeatureName   -> This is the name of the feature used when setting the
+;                   DYNAMIC_FEATURES variable to enable usage of this feature.
+;  DTMF_sequence -> This is the key sequence used to activate this feature.
+;  ActivateOn    -> This is the channel of the call that the application will be executed
+;                   on. Valid values are "self" and "peer". "self" means run the
+;                   application on the same channel that activated the feature. "peer"
+;                   means run the application on the opposite channel from the one that
+;                   has activated the feature.
+;  ActivatedBy   -> This is which channel is allowed to activate this feature. Valid
+;                   values are "caller", "callee", and "both". "both" is the default.
+;                   The "caller" is the channel that executed the Dial application, while
+;                   the "callee" is the channel called by the Dial application.
+;  Application   -> This is the application to execute.
+;  AppArguments  -> These are the arguments to be passed into the application.  If you need
+;                   commas in your arguments, you should use either the second or third
+;                   syntax, above.
+;  MOH_Class     -> This is the music on hold class to play while the idle
+;                   channel waits for the feature to complete. If left blank,
+;                   no music will be played.
+;
+
+;
+; IMPORTANT NOTE: The applicationmap is not intended to be used for all Asterisk
+;   applications. When applications are used in extensions.conf, they are executed
+;   by the PBX core. In this case, these applications are executed outside of the
+;   PBX core, so it does *not* make sense to use any application which has any
+;   concept of dialplan flow. Examples of this would be things like Macro, Goto,
+;   Background, WaitExten, and many more.
+;
+; Enabling these features means that the PBX needs to stay in the media flow and
+; media will not be re-directed if DTMF is sent in the media stream.
+;
+; Example Usage:
+;
+;testfeature => #9,peer,Playback,tt-monkeys  ;Allow both the caller and callee to play
+;                                            ;tt-monkeys to the opposite channel
+;
+; Set arbitrary channel variables, based upon CALLERID number (Note that the application
+; argument contains commas)
+;retrieveinfo => #8,peer,Set(ARRAY(CDR(mark),CDR(name))=${ODBC_FOO(${CALLERID(num)})})
+;
+;pauseMonitor   => #1,self/callee,Pausemonitor     ;Allow the callee to pause monitoring
+;                                                  ;on their channel
+;unpauseMonitor => #3,self/callee,UnPauseMonitor   ;Allow the callee to unpause monitoring
+;                                                  ;on their channel
+
+; Dynamic Feature Groups:
+;   Dynamic feature groups are groupings of features defined in [applicationmap]
+;   that can have their own custom key mappings.  To give a channel access to a dynamic
+;   feature group, add the group name to the value of the DYNAMIC_FEATURES variable.
+;
+; example:
+; [myGroupName]         ; defines the group named myGroupName
+; testfeature => #9     ; associates testfeature with the group and the keycode '#9'.
+; pauseMonitor =>       ; associates pauseMonitor with the group and uses the keycode specified
+;                       ; in the [applicationmap].

asterisk/festival.conf

@@ -0,0 +1,35 @@
+;
+; Festival Configuration
+;
+[general]
+;
+; Host which runs the festival server (default : localhost);
+;
+;host=localhost
+;
+; Port on host where the festival server runs (default : 1314)
+;
+;port=1314
+;
+; Use cache (yes, no - defaults to no)
+;
+;usecache=yes
+;
+; 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).
+; THIS DIRECTORY *MUST* EXIST and must be writable from the asterisk process.
+; Defaults to /tmp/
+;
+;cachedir=/var/lib/asterisk/festivalcache/
+;
+; 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.
+;
+;festivalcommand=(tts_textasterisk "%s" 'file)(quit)\n
+;
+;
+

asterisk/followme.conf

@@ -0,0 +1,91 @@
+; Find-Me / Follow-Me Configuration File
+[general]
+;
+featuredigittimeout=>5000
+; The number of ms to wait for a digit input for the callee on whether to take the call or
+; not before we consider them "done" entering digits.
+;
+takecall=>1
+; The global default keypress for the callee to take taking the current call. This can be
+; a single digit or multiple digits. Default is "1".
+; Note this string must not be a substring of declinecall.
+;
+declinecall=>2
+; The global default keypress for the callee to decline taking the current call. This can
+; be a single digit or multiple digits. Default is "2".
+; Note this string must not be a substring of takecall.
+;
+call_from_prompt=>followme/call-from
+; The global default for the 'Incoming call from' message.
+;
+norecording_prompt=>followme/no-recording
+; The global default for the 'You have an incoming call' message when the caller elects
+; not to leave their name or the option isn't set for them to do so.
+;
+options_prompt=>followme/options
+; The global default for the 'Press 1 to accept this call or press 2 to decline it' message.
+;
+pls_hold_prompt=>followme/pls-hold-while-try
+; The global default for 'Please hold while we try and connect your call' message.
+;
+status_prompt=>followme/status
+; The global default for 'The party you're calling isn't at their desk' message.
+;
+sorry_prompt=>followme/sorry
+; The global default for 'I'm sorry, but we were unable to locate your party' message.
+;
+;
+[default]
+musicclass=>default
+; The moh class that should be used for the caller while they are waiting to be connected.
+context=>default
+; The context to dial the numbers from
+number=>01233456,25
+; The a follow-me number to call. The format is:
+; number=> <number to call[&2nd #[&...]]>[,<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
+; 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
+; numbers in the same step is supported with this application if you'd like to dial
+; multiple numbers in the same followme step.
+;
+; The timeout value is the amount of time allowed between the time the dialing step
+; starts and the callee answers.  The callee then has until the timeout of the last
+; step to make a choice on whether to take the call or not.  That being the case,
+; you may want to make the timeout on the last step longer to give enough time to
+; make the choice to accept or not.
+takecall=>1
+; The keypress for the callee to take taking the current call. This can be
+; a single digit or multiple digits. Default is the global default.
+; Note this string must not be a substring of declinecall.
+;
+declinecall=>2
+; The keypress for the callee to decline taking the current call. This can
+; be a single digit or multiple digits. Default is the global default.
+; Note this string must not be a substring of takecall.
+;
+call_from_prompt=>followme/call-from
+; The 'Incoming call from' message prompt. Default is the global default.
+;
+norecording_prompt=>followme/no-recording
+; The 'You have an incoming call' message prompt when the caller elects
+; not to leave their name or the option isn't set for them to do so. Default
+; is the global default.
+;
+options_prompt=>followme/options
+; The 'Press 1 to accept this call or press 2 to decline it' message prompt.
+; Default is the global default.
+;
+pls_hold_prompt=>followme/pls-hold-while-try
+; The 'Please hold while we try and connect your call' message prompt.
+; Default is the global default.
+;
+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
+; The 'I'm sorry, but we were unable to locate your party' message prompt. Default
+; is the global default.
+
+

asterisk/func_odbc.conf

@@ -0,0 +1,96 @@
+;
+; func_odbc.conf
+;
+; Each context is a separately defined function.  By convention, all
+; functions are entirely uppercase, so the defined contexts should also
+; be all-uppercase, but there is nothing that enforces this.  All functions
+; are case-sensitive, however.
+;
+; For substitution, you have ${ARG1}, ${ARG2} ... ${ARGn}
+; for the arguments to each SQL statement.
+;
+; In addition, for write statements, you have ${VAL1}, ${VAL2} ... ${VALn}
+; parsed, just like arguments, for the values.  In addition, if you want the
+; whole value, never mind the parsing, you can get that with ${VALUE}.
+;
+;
+; If you have data which may potentially contain single ticks, you may wish
+; to use the dialplan function SQL_ESC() to escape the data prior to its
+; inclusion in the SQL statement.
+;
+;
+; The following options are available in this configuration file:
+;
+; readhandle   A comma-separated list of DSNs (from res_odbc.conf) to use when
+;              executing the readsql statement.  Each DSN is tried, in
+;              succession, until the statement succeeds.  You may specify up to
+;              5 DSNs per function class.  If not specified, it will default to
+;              the value of writehandle or dsn, if specified.
+; writehandle  A comma-separated list of DSNs (from res_odbc.conf) to use when
+;              executing the writesql statement.  The same rules apply as to
+;              readhandle.  "dsn" is a synonym for "writehandle".
+; readsql      The statement to execute when reading from the function class.
+; writesql     The statement to execute when writing to the function class.
+; insertsql    The statement to execute when writing to the function class
+;              succeeds, but initially indicates that 0 rows were affected.
+; prefix       Normally, all function classes are prefixed with "ODBC" to keep
+;              them uniquely named.  You may choose to change this prefix, which
+;              may be useful to segregate a collection of certain function
+;              classes from others.
+; escapecommas This option may be used to turn off the default behavior of
+;              escaping commas which occur within a field.  If commas are
+;              escaped (the default behavior), then fields containing commas
+;              will be treated as a single value when assigning to ARRAY() or
+;              HASH().  If commas are not escaped, then values will be separated
+;              at the comma within fields.  Please note that turning this option
+;              off is incompatible with the functionality of HASH().
+; synopsis     Appears in the synopsis field for the command
+;              'core show function <function name>'
+; mode         This option may be set to 'multirow' to allow the function
+;              specified to return more than a single row.  However, this
+;              changes the way that func_odbc normally works.  Instead of the
+;              invocation of the function returning a row, it returns an opaque
+;              ID, which may be passed to ODBC_FETCH() to return each row in
+;              turn.  ODBC_FETCH_STATUS returns SUCCESS or FAILURE, to indicate
+;              whether any results were stored, and you should call ODBC_Finish
+;              on the ID to clean up any remaining results when you are done
+;              with the query.  Also, the variable ODBCROWS is set initially,
+;              which may be used in an iterative fashion to return each row in
+;              the result.
+;              Please note that multirow queries are isolated to the channel,
+;              and rows may not be fetched outside of the channel where the
+;              query was initially performed.  Additionally, as the results are
+;              associated with a channel, mode=multirow is incompatible with
+;              the global space.
+; rowlimit     Rowlimit limits the total number of rows which can be stored for
+;              that query.  For mode=multirow, otherwise, func_odbc will
+;              attempt to store all rows in the resultset, up to the maximum
+;              amount of memory.  In normal mode, rowlimit can be set to allow
+;              additional rows to be fetched, rather than just the first one.
+;              These additional rows can be returned by using the name of the
+;              function which was called to retrieve the first row as an
+;              argument to ODBC_FETCH().
+
+
+; ODBC_SQL - Allow an SQL statement to be built entirely in the dialplan
+[SQL]
+dsn=mysql1
+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.
+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
+
+; ODBC_PRESENCE - Retrieve and update presence
+[PRESENCE]
+dsn=mysql1
+readsql=SELECT location FROM presence WHERE id='${SQL_ESC(${ARG1})}'
+writesql=UPDATE presence SET location='${SQL_ESC(${VAL1})}' WHERE id='${SQL_ESC(${ARG1})}'
+

asterisk/gtalk.conf

@@ -0,0 +1,27 @@
+[general]
+;context=default		; Context to dump call into
+;bindaddr=0.0.0.0		; Address to bind to
+;externip=127.0.0.1		; Set your external ip if you are behind a NAT.
+;stunaddr=mystunserver.com	; Get your external ip from a STUN server.
+				; Note, if the STUN query is successful, this will
+				; replace any value placed in externip;
+;allowguest=yes			; Allow calls from people not in list of peers
+;disallow=all
+;allow=gsm
+;allow=ulaw
+;parkinglot=soccer		; Sets the default parking lot for call parking
+				; Parkinglots are configured in features.conf
+
+;[guest]			; special account for options on guest account
+;disallow=all
+;allow=ulaw
+;context=guest
+
+;[ogorman]
+;username=ogorman@gmail.com	; username of the peer your 
+				; calling or accepting calls from
+;disallow=all
+;allow=ulaw
+;context=default
+;connection=asterisk		; client or component in jabber.conf for the
+				; call to leave on.

asterisk/h323.conf

@@ -0,0 +1,210 @@
+; The NuFone Network's
+; Open H.323 driver configuration
+;
+[general]
+port = 1720
+;bindaddr = 1.2.3.4 	; this SHALL contain a single, valid IP address for this machine
+;
+; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
+;tos_audio=ef		; Sets TOS for RTP audio packets.
+;cos_audio=5		; Sets 802.1p priority for RTP audio packets.
+;
+; You may specify a global default AMA flag for iaxtel calls.  It must be
+; one of 'default', 'omit', 'billing', or 'documentation'.  These flags
+; are used in the generation of call detail records.
+;
+;amaflags = default
+;
+; You may specify a default account for Call Detail Records in addition
+; to specifying on a per-user basis
+;
+;accountcode=lss0101
+;
+; You can fine tune codecs here using "allow" and "disallow" clauses
+; with specific codecs.  Use "all" to represent all formats.
+;
+;disallow=all
+;allow=all		; turns on all installed codecs
+;disallow=g723.1	; Hm...  Proprietary, don't use it...
+;allow=gsm		; Always allow GSM, it's cool :)
+;allow=ulaw		; see https://wiki.asterisk.org/wiki/display/AST/RTP+Packetization
+			; for framing options
+;autoframing=yes	; Set packetization based on the remote endpoint's (ptime)
+			; preferences. Defaults to no.
+;
+; User-Input Mode (DTMF)
+;
+; valid entries are:   rfc2833, inband, cisco, h245-signal
+; default is rfc2833
+;dtmfmode=rfc2833
+;
+; Default RTP Payload to send RFC2833 DTMF on.  This is used to
+; interoperate with broken gateways which cannot successfully
+; negotiate a RFC2833 payload type in the TerminalCapabilitySet.
+; To specify required payload type, put it after colon in dtmfmode
+; option like
+;dtmfmode=rfc2833:101
+; or
+;dtmfmode=cisco:121
+;
+; 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
+;gatekeeper = DISABLE
+;
+;
+; Tell Asterisk whether or not to accept Gatekeeper
+; routed calls or not. Normally this should always
+; be set to yes, unless you want to have finer control
+; over which users are allowed access to Asterisk.
+; Default: YES
+;
+;AllowGKRouted = yes
+;
+; When the channel works without gatekeeper, there is possible to
+; reject calls from anonymous (not listed in users) callers.
+; Default is to allow anonymous calls.
+;
+;AcceptAnonymous = yes
+;
+; Optionally you can determine a user by Source IP versus its H.323 alias.
+; Default behavour is to determine user by H.323 alias.
+;
+;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
+; use user authentication at all.
+;
+;context=default
+;
+; Use this option to help Cisco (or other) gateways to setup backward voice
+; path to pass inband tones to calling user (see, for example,
+; http://www.cisco.com/warp/public/788/voip/ringback.html)
+;
+; Add PROGRESS information element to SETUP message sent on outbound calls
+; to notify about required backward voice path. Valid values are:
+;   0 - don't add PROGRESS information element (default);
+;   1 - call is not end-end ISDN, further call progress information can
+;        possibly be available in-band;
+;   3 - origination address is non-ISDN (Cisco accepts this value only);
+;   8 - in-band information or an appropriate pattern is now available;
+;progress_setup = 3
+;
+; Add PROGRESS information element (IE) to ALERT message sent on incoming
+; calls to notify about required backwared voice path. Valid values are:
+;   0 - don't add PROGRESS IE (default);
+;   8 - in-band information or an appropriate pattern is now available;
+;progress_alert = 8
+;
+; Generate PROGRESS message when H.323 audio path has established to create
+; backward audio path at other end of a call.
+;progress_audio = yes
+;
+; Specify how to inject non-standard information into H.323 messages. When
+; the channel receives messages with tunneled information, it automatically
+; enables the same option for all further outgoing messages independedly on
+; options has been set by the configuration. This behavior is required, for
+; example, for Cisco CallManager when Q.SIG tunneling is enabled for a
+; gateway where Asterisk lives.
+; The option can be used multiple times, one option per line.
+;tunneling=none               ; Totally disable tunneling (default)
+;tunneling=cisco              ; Enable Cisco-specific tunneling
+;tunneling=qsig               ; Enable tunneling via Q.SIG messages
+;
+; Specify how to pass hold notification to remote party. Default is to
+; use H.450.4 supplementary service message.
+;hold=none                    ; Do not pass hold/retrieve notifications
+;hold=notify                  ; Use H.225 NOTIFY message
+;hold=q931only                ; Use stripped H.225 NOTIFY message (Q.931 part
+;                             ; only, usable for Cisco CallManager)
+;hold=h450                    ; Pass notification as H.450.4 supplementary
+;                             ; service
+;
+;------------------------------ 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.
+
+; jbforce = no                ; Forces the use of a jitterbuffer on the receive side of a H323
+                              ; 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.
+
+; 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.
+
+; jblog = no                  ; Enables jitterbuffer frame logging. Defaults to "no".
+;-----------------------------------------------------------------------------------
+;
+; H.323 Alias definitions
+;
+; Type 'h323' will register aliases to the endpoint
+; and Gatekeeper, if there is one.
+;
+; Example: if someone calls time@your.asterisk.box.com
+; Asterisk will send the call to the extension 'time'
+; in the context default
+;
+;   [default]
+;   exten => time,1,Answer
+;   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
+; 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
+; routed to the H.323 alias 'time'.
+;
+;[time]
+;type=h323
+;e164=18102341212
+;context=default
+;
+;[det-gw]
+;type=h323
+;prefix=1248,1313
+;context=detroit
+;
+;
+; Inbound H.323 calls from BillyBob would land in the incoming
+; context with a maximum of 4 concurrent incoming calls
+;
+;
+; Note: If keyword 'incominglimit' are omitted Asterisk will not
+; enforce any maximum number of concurrent calls.
+;
+;[BillyBob]
+;type=user
+;host=192.168.1.1
+;context=incoming
+;incominglimit=4
+;h245Tunneling=no
+;
+;
+; Outbound H.323 call to Larry using SlowStart
+;
+;[Larry]
+;type=peer
+;host=192.168.2.1
+;fastStart=no
+
+
+

asterisk/http.conf

@@ -0,0 +1,86 @@
+;
+; Asterisk Builtin mini-HTTP server
+;
+;
+; Note about Asterisk documentation:
+;   If Asterisk was installed from a tarball, then the HTML documentation should
+;   be installed in the static-http/docs directory which is
+;   (/var/lib/asterisk/static-http/docs) on linux by default.  If the Asterisk
+;   HTTP server is enabled in this file by setting the "enabled", "bindaddr",
+;   and "bindport" options, then you should be able to view the documentation
+;   remotely by browsing to:
+;       http://<server_ip>:<bindport>/static/docs/index.html
+;
+[general]
+;
+; Whether HTTP/HTTPS interface is enabled or not.  Default is no.
+; This also affects manager/rawman/mxml access (see manager.conf)
+;
+;enabled=yes
+;
+; Address to bind to, both for HTTP and HTTPS. You MUST specify
+; a bindaddr in order for the HTTP server to run. There is no
+; default value.
+;
+bindaddr=127.0.0.1
+;
+; Port to bind to for HTTP sessions (default is 8088)
+;
+;bindport=8088
+;
+; Prefix allows you to specify a prefix for all requests
+; to the server.  The default is blank.  If uncommented
+; all requests must begin with /asterisk
+;
+;prefix=asterisk
+;
+; sessionlimit specifies the maximum number of httpsessions that will be
+; allowed to exist at any given time. (default: 100)
+;
+;sessionlimit=100
+;
+; session_inactivity specifies the number of milliseconds to wait for
+; more data over the HTTP connection before closing it.
+;
+; Default: 30000
+;session_inactivity=30000
+;
+; Whether Asterisk should serve static content from http-static
+; Default is no.
+;
+;enablestatic=yes
+;
+; Redirect one URI to another.  This is how you would set a
+; 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:
+;
+;redirect = / /static/config/index.html
+;
+; HTTPS support. In addition to enabled=yes, you need to
+; explicitly enable tls, define the port to use,
+; and have a certificate somewhere.
+;tlsenable=yes          ; enable tls - default no.
+;tlsbindaddr=0.0.0.0:8089    ; address and port to bind to - default is bindaddr and port 8089.
+;
+;tlscertfile=</path/to/certificate.pem>  ; path to the certificate file (*.pem) only.
+;tlsprivatekey=</path/to/private.pem>    ; path to private key file (*.pem) only.
+; If no path is given for tlscertfile or tlsprivatekey, default is to look in current
+; directory. If no tlsprivatekey is given, default is to search tlscertfile for private key.
+;
+; To produce a certificate you can e.g. use openssl. This places both the cert and
+; private in same .pem file.
+; openssl req -new -x509 -days 365 -nodes -out /tmp/foo.pem -keyout /tmp/foo.pem
+;
+; The post_mappings section maps URLs to real paths on the filesystem.  If a
+; POST is done from within an authenticated manager session to one of the
+; configured POST mappings, then any files in the POST will be placed in the
+; configured directory.
+;
+;[post_mappings]
+;
+; In this example, if the prefix option is set to "asterisk", then using the
+; POST URL: /asterisk/uploads will put files in /var/lib/asterisk/uploads/.
+;uploads = /var/lib/asterisk/uploads/
+;

asterisk/iax.conf

@@ -0,0 +1,668 @@
+;
+; Inter-Asterisk eXchange v2 (IAX2) Channel Driver configuration
+;
+; This configuration is read when the chan_iax2.so module is loaded, and is
+; re-read when the module is reloaded, such as when invoking the CLI command:
+;
+;     *CLI> iax2 reload
+;
+
+; General settings, like port number to bind to, and an option address (the
+; default is to bind to all local addresses).
+
+[general]
+
+; Listener Addresses
+;
+; Use the 'bindaddr' and 'bindport' options to specify on which address and port
+; the IAX2 channel driver will listen for incoming requests.
+;
+;
+
+;bindport=4569           ; The default port to listen on
+                         ; 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=192.168.0.1    ; You can specify 'bindaddr' more than once to bind to
+                         ; multiple addresses, but the first will be the
+                         ; default.
+
+;
+; Set 'iaxcompat' to yes if you plan to use layered switches or some other
+; scenario which may cause some delay when doing a lookup in the dialplan. It
+; incurs a small performance hit to enable it. This option causes Asterisk to
+; spawn a separate thread when it receives an IAX2 DPREQ (Dialplan Request)
+; instead of blocking while it waits for a response.
+;
+; Accepted values: yes, no
+; Default value:   no
+;
+;iaxcompat=yes
+;
+
+;
+; Disable UDP checksums (if nochecksums is set, then no checkums will
+; be calculated/checked on systems supporting this feature)
+;
+; Accepted values: yes, no
+; Default value:   no
+;
+;nochecksums=yes
+;
+
+;
+; 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.
+;
+; Accepted values: yes, no
+; Default value:   no
+;
+;delayreject=yes
+;
+
+;
+; You may specify a global default AMA flag for iaxtel calls.  These flags are
+; used in the generation of call detail records.
+;
+; Accepted values: default, omit, billing, documentation
+; Default value:   default
+;
+;amaflags=billing
+;
+
+;
+; ADSI (Analog Display Services Interface) can be enabled if you have (or may
+; have) ADSI compatible CPE equipment.
+;
+; Accepted values: yes, no
+; Default value:   no
+;
+;adsi=yes
+;
+
+;
+; Whether or not to perform an SRV lookup on outbound calls.
+;
+; Accepted values: yes, no
+; Default value:   no
+;
+;srvlookup=yes
+;
+
+;
+; You may specify a default account for Call Detail Records (CDRs) in addition to
+; specifying on a per-user basis.
+;
+; Accepted values: Any string value up to 19 characters in length
+; Default value:   <empty>
+;
+;accountcode=lss0101
+;
+
+;
+; You may specify a global default language for users.  This can be specified
+; also on a per-user basis.  If omitted, will fallback to English (en).
+;
+; Accepted values: A language tag such as 'en' or 'es'
+; Default value:   en
+;
+;language=en
+;
+
+;
+; This option specifies a preference for which music-on-hold class this channel
+; should listen to when put on hold if the music class has not been set on the
+; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
+; channel putting this one on hold did not suggest a music class.
+;
+; If this option is set to "passthrough", then the hold message will always be
+; passed through as signalling instead of generating hold music locally.
+;
+; This option may be specified globally, or on a per-user or per-peer basis.
+;
+; Accepted values: passthrough, or any music-on-hold class name
+; Default value:   <empty>
+;
+;mohinterpret=default
+;
+
+;
+; The 'mohsuggest' 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.
+;
+;mohsuggest=default
+;
+
+;
+; Specify bandwidth of low, medium, or high to control which codecs are used
+; in general.
+;
+bandwidth=low
+;
+
+;
+; You can also fine tune codecs here using "allow" and "disallow" clauses with
+; specific codecs.  Use "all" to represent all formats.
+;
+;allow=all
+;disallow=g723.1
+disallow=lpc10
+;allow=gsm
+;
+
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+; Jitter Buffer
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+
+;
+; You can adjust several parameters relating to the jitter buffer.  The jitter
+; buffer's function is to compensate for varying network delay.
+;
+; All of the jitter buffer settings are in milliseconds.  The jitter buffer
+; works for INCOMING audio only - the outbound audio will be dejittered by the
+; jitter buffer at the other end.
+;
+; jitterbuffer=yes|no: global default as to whether you want
+; the jitter buffer at all.
+;
+; 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
+; themselves, so this option will force * to always jitterbuffer, even in this
+; case.
+;
+; maxjitterbuffer: a maximum size for the jitter buffer.
+; Setting a reasonable maximum here will prevent the call delay
+; from rising to silly values in extreme situations; you'll hear
+; SOMETHING, even though it will be jittery.
+;
+; resyncthreshold: when the jitterbuffer notices a significant change in delay
+; that continues over a few frames, it will resync, assuming that the change in
+; delay was caused by a timestamping mix-up. The threshold for noticing a
+; change in delay is measured as twice the measured jitter plus this resync
+; threshold.
+; Resyncing can be disabled by setting this parameter to -1.
+;
+; maxjitterinterps: the maximum number of interpolation frames the jitterbuffer
+; should return in a row. Since some clients do not send CNG/DTX frames to
+; indicate silence, the jitterbuffer will assume silence has begun after
+; returning this many interpolations. This prevents interpolating throughout
+; a long silence.
+;
+; jittertargetextra: number of milliseconds by which the new jitter buffer
+; will pad its size. the default is 40, so without modification, the new
+; jitter buffer will set its size to the jitter value plus 40 milliseconds.
+; increasing this value may help if your network normally has low jitter,
+; but occasionally has spikes.
+;
+
+jitterbuffer=no
+forcejitterbuffer=no
+;maxjitterbuffer=1000
+;maxjitterinterps=10
+;resyncthreshold=1000
+;jittertargetextra=40
+
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+; IAX2 Encryption
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+
+;
+; Enable IAX2 encryption.  The default is 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.  The default is no.
+;
+;forceencryption=yes
+;
+
+; This option defines the maximum payload in bytes an IAX2 trunk can support at
+; a given time.  The best way to explain this is to provide an example.  If the
+; maximum number of calls to be supported is 800, and each call transmits 20ms
+; frames of audio using ulaw:
+;
+;     (8000hz / 1000ms) * 20ms * 1 byte per sample = 160 bytes per frame
+;
+; The maximum load in bytes is:
+;
+;     (160 bytes per frame) * (800 calls) = 128000 bytes
+;
+; Once this limit is reached, calls may be dropped or begin to lose audio.
+; Depending on the codec in use and number of channels to be supported this value
+; may need to be raised, but in most cases the default value is large enough.
+;
+; trunkmaxsize = 128000 ; defaults to 128000 bytes, which supports up to 800
+                        ; calls of ulaw at 20ms a frame.
+
+; With a large amount of traffic on IAX2 trunks, there is a risk of bad voice
+; quality when allowing the Linux system to handle fragmentation of UDP packets.
+; Depending on the size of each payload, allowing the OS to handle fragmentation
+; may not be very efficient. This setting sets the maximum transmission unit for
+; IAX2 UDP trunking. The default is 1240 bytes which means if a trunk's payload
+; is over 1240 bytes for every 20ms it will be broken into multiple 1240 byte
+; messages.  Zero disables this functionality and let's the OS handle
+; fragmentation.
+;
+; trunkmtu = 1240    ; trunk data will be sent in 1240 byte messages.
+
+; trunkfreq sets how frequently trunk messages are sent in milliseconds. This
+; value is 20ms by default, which means the trunk will send all the data queued
+; to it in the past 20ms.  By increasing the time between sending trunk messages,
+; the trunk's payload size will increase as well.  Note, depending on the size
+; set by trunkmtu, messages may be sent more often than specified.  For example
+; if a trunk's message size grows to the trunkmtu size before 20ms is reached
+; that message will be sent immediately.  Acceptable values are between 10ms and
+; 1000ms.
+;
+; trunkfreq=20    ; How frequently to send trunk msgs (in ms). This is 20ms by
+                  ; default.
+
+; Should we send timestamps for the individual sub-frames within trunk frames?
+; There is a small bandwidth use for these (less than 1kbps/call), but they
+; ensure that frame timestamps get sent end-to-end properly.  If both ends of
+; all your trunks go directly to TDM, _and_ your trunkfreq equals the frame
+; length for your codecs, you can probably suppress these.  The receiver must
+; also support this feature, although they do not also need to have it enabled.
+;
+; trunktimestamps=yes
+
+; Minimum and maximum amounts of time that IAX2 peers can request as a
+; registration expiration interval (in seconds).
+; minregexpire = 60
+; maxregexpire = 60
+
+; IAX2 helper threads
+
+; Establishes the number of iax helper threads to handle I/O.
+; iaxthreadcount = 10
+
+; Establishes the number of extra dynamic threads that may be spawned to handle I/O
+; iaxmaxthreadcount = 100
+
+;
+; We can register with another IAX2 server to let him know where we are
+; in case we have a dynamic IP address for example
+;
+; Register with tormenta using username marko and password secretpass
+;
+;register => marko:secretpass@tormenta.linux-support.net
+;
+; Register joe at remote host with no password
+;
+;register => joe@remotehost:5656
+;
+; Register marko at tormenta.linux-support.net using RSA key "torkey"
+;
+;register => marko:[torkey]@tormenta.linux-support.net
+;
+; 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
+; calls arrive at the "s" extension of "default" context.
+;
+;register => user:pass@iaxtel.com
+;
+; Sample Registration for IAX2 + FWD
+;
+; To register using IAX2 with FWD, it must be enabled by visiting the URL
+; http://www.fwdnet.net/index.php?section_id=112
+;
+; Note that you need an extension in you default context which matches
+; your free world dialup number.  Please replace "FWDNumber" with your
+; FWD number and "passwd" with your password.
+;
+;register => FWDNumber:passwd@iax.fwdnet.net
+;
+; Through the use of the res_stun_monitor module, Asterisk has the ability to detect when the
+; perceived external network address has changed.  When the stun_monitor is installed and
+; configured, chan_iax will renew all outbound registrations when the monitor detects any sort
+; of network change has occurred. By default this option is enabled, but only takes effect once
+; res_stun_monitor is configured.  If res_stun_monitor is enabled and you wish to not
+; generate all outbound registrations on a network change, use the option below to disable
+; this feature.
+;
+; subscribe_network_change_event = yes ; on by default
+;
+; You can enable authentication debugging to increase the amount of
+; debugging traffic.
+;
+;authdebug = yes
+;
+; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
+;tos=ef
+;cos=5
+;
+; 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.  The actual extension is the 'regexten' parameter of the registering
+; peer or its name if 'regexten' is not provided.  More than one regexten
+; may be supplied if they are separated by '&'.  Patterns may be used in
+; regexten.
+;
+;regcontext=iaxregistrations
+;
+; If we don't get ACK to our NEW within 2000ms, and autokill is set to yes,
+; then we cancel the whole thing (that's enough time for one retransmission
+; only).  This is used to keep things from stalling for a long time for a host
+; that is not available, but would be ill advised for bad connections.  In
+; addition to 'yes' or 'no' you can also specify a number of milliseconds.
+; See 'qualify' for individual peers to turn on for just a specific peer.
+;
+autokill=yes
+;
+; codecpriority controls the codec negotiation of an inbound IAX2 call.
+; 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:
+;
+; caller   - Consider the callers preferred order ahead of the host's.
+; host     - Consider the host's preferred order ahead of the caller's.
+; disabled - Disable the consideration of codec preference altogether.
+;            (this is the original behaviour before preferences were added)
+; reqonly  - Same as disabled, only do not consider capabilities if
+;            the requested format is not available the call will only
+;            be accepted if the requested format is available.
+;
+; The default value is 'host'
+;
+;codecpriority=host
+;
+; allowfwdownload controls whether this host will serve out firmware to
+; IAX2 clients which request it.  This has only been used for the IAXy,
+; and it has been recently proven that this firmware distribution method
+; can be used as a source of traffic amplification attacks.  Also, the
+; IAXy firmware has not been updated for at least 18 months, so unless
+; you are provisioning IAXys in a secure network, we recommend that you
+; leave this option to the default, off.
+;
+;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)
+
+;rtsavesysname=yes  ; Save systemname in realtime database at registration
+                    ; Default = 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'.
+
+;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.
+
+;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)
+
+;parkinglot=edvina     ; Default parkinglot for IAX2 peers and users
+                       ; This can also be configured per device
+                       ; Parkinglots are defined in features.conf
+
+;
+; The following two options are used to disable call token validation for the
+; purposes of interoperability with IAX2 endpoints that do not yet support it.
+;
+; Call token validation can be set as optional for a single IP address or IP
+; address range by using the 'calltokenoptional' option. 'calltokenoptional' is
+; only a global option.
+;
+;calltokenoptional=209.16.236.73/255.255.255.0
+;
+; By setting 'requirecalltoken=no', call token validation becomes optional for
+; that peer/user.  By setting 'requirecalltoken=auto', call token validation
+; is optional until a call token supporting peer registers successfully using
+; call token validation.  This is used as an indication that from now on, we
+; can require it from this peer.  So, requirecalltoken is internally set to yes.
+; requirecalltoken may only be used in peer/user/friend definitions,
+; not in the global scope.
+; By default, 'requirecalltoken=yes'.
+;
+;requirecalltoken=no
+;
+
+;
+; These options are used to limit the amount of call numbers allocated to a
+; single IP address.  Before changing any of these values, it is highly encouraged
+; to read the user guide associated with these options first.  In most cases, the
+; default values for these options are sufficient.
+;
+; The 'maxcallnumbers' option limits the amount of call numbers allowed for each
+; individual remote IP address.  Once an IP address reaches it's call number
+; limit, no more new connections are allowed until the previous ones close.  This
+; option can be used in a peer definition as well, but only takes effect for
+; the IP of a dynamic peer after it completes registration.
+;
+;maxcallnumbers=512
+;
+; The 'maxcallnumbers_nonvalidated' is used to set the combined number of call
+; numbers that can be allocated for connections where call token  validation
+; has been disabled.  Unlike the 'maxcallnumbers' option, this limit is not
+; separate for each individual IP address.  Any connection resulting in a
+; non-call token validated call number being allocated contributes to this
+; limit.  For use cases, see the call token user guide.  This option's
+; default value of 8192 should be sufficient in most cases.
+;
+;maxcallnumbers_nonvalidated=1024
+;
+; The [callnumberlimits] section allows custom call number limits to be set
+; for specific IP addresses and IP address ranges.  These limits take precedence
+; over the global 'maxcallnumbers' option, but may still be overridden by a
+; peer defined 'maxcallnumbers' entry.  Note that these limits take effect
+; for every individual address within the range, not the range as a whole.
+;
+;[callnumberlimits]
+;10.1.1.0/255.255.255.0 = 24
+;10.1.2.0/255.255.255.0 = 32
+;
+
+; The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and '-' not
+; in square brackets.  For example, the Caller*ID value 555.5555 becomes 5555555
+; when this option is enabled.  Disabling this option results in no modification
+; of the Caller*ID value, which is necessary when the Caller*ID represents something
+; that must be preserved.  This option can only be used in the [general] section.
+; By default this option is on.
+;
+;shrinkcallerid=yes     ; on by default
+
+; Guest sections for unauthenticated connection attempts.  Just specify an
+; empty secret, or provide no secret section.
+;
+[guest]
+type=user
+context=public
+callerid="Guest IAX User"
+
+;
+; Trust Caller*ID Coming from iaxtel.com
+;
+[iaxtel]
+type=user
+context=default
+auth=rsa
+inkeys=iaxtel
+
+;
+; Trust Caller*ID Coming from iax.fwdnet.net
+;
+[iaxfwd]
+type=user
+context=default
+auth=rsa
+inkeys=freeworlddialup
+
+;
+; Trust Caller*ID delivered over DUNDi/e164
+;
+;[dundi]
+;type=user
+;dbsecret=dundi/secret
+;context=dundi-e164-local
+
+;
+; Further user sections may be added, specifying a context and a secret used
+; for connections with that given authentication name.  Limited IP based
+; access control is allowed by use of "permit", "deny", and "acl" keywords.
+; Multiple rules are permitted. Multiple permitted contexts may be specified,
+; in which case the first will be the default.  You can also override
+; Caller*ID so that when you receive a call you set the Caller*ID to be what
+; you want instead of trusting what the remote user provides
+;
+; There are three authentication methods that are supported:  md5, plaintext,
+; and rsa.  The least secure is "plaintext", which sends passwords cleartext
+; 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
+; 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
+; while private keys are named /var/lib/asterisk/keys/<name>.key.  Private
+; keys should always be 3DES encrypted.
+;
+;
+; NOTE: All hostnames and IP addresses in this file are for example purposes
+;       only; you should not expect any of them to actually be available for
+;       your use.
+;
+;[markster]
+;type=user
+;context=default
+;context=local
+;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.
+;dbsecret=mysecrets/place    ; Secrets can be stored in astdb, too
+;transfer=no                 ; Disable IAX2 native transfer
+;transfer=mediaonly          ; When doing IAX2 native transfers, transfer only
+                             ; the media stream
+;jitterbuffer=yes            ; Override the global setting and enable the jitter
+                             ; buffer for this user
+;maxauthreq=10               ; Set the maximum number of outstanding AUTHREQs
+                             ; waiting for replies. If this limit is reached,
+                             ; any further authentication will be blocked, until
+                             ; the pending requests expire or a reply is
+                             ; received.
+;callerid="Mark Spencer" <(256) 428-6275>
+;deny=0.0.0.0/0.0.0.0
+;accountcode=markster0101
+;permit=209.16.236.73/255.255.255.0
+;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.  The default is on.
+
+;
+; Peers may also be specified, with a secret and a remote hostname.
+;
+[demo]
+type=peer
+username=asterisk
+secret=supersecret
+host=216.207.245.47
+description=Demo System At Digium    ; Description of this peer, as listed by
+                                     ; 'iax2 show peers'
+;sendani=no
+;host=asterisk.linux-support.net
+;port=5036
+;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.  The default
+                            ; is 'no.'
+;qualifyfreqok = 60000      ; How frequently to ping the peer when everything
+                            ; seems to be OK, in milliseconds.
+;qualifyfreqnotok = 10000   ; How frequently to ping the peer when it's 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.  The default is 'on.'
+
+; Peers can remotely register as well, so that they can be mobile.  Default
+; IPs 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
+; another PBX.
+;connectedline=yes ; Set if connected line and redirecting information updates
+;                  ; are passed between Asterisk servers for this peer.
+;                  ; yes - Sending and receiving updates are enabled.
+;                  ; send - Only send updates.
+;                  ; receive - Only process received updates.
+;                  ; no - Sending and receiving updates are disabled.
+;                  ; Default is "no".
+;                  ;
+;                  ; Note: Because of an incompatibility between Asterisk v1.4
+;                  ; and Asterisk v1.8 or later, this option must be set
+;                  ; to "no" toward the Asterisk v1.4 peer.  A symptom of the
+;                  ; incompatibility is the call gets disconnected unexpectedly.
+
+
+;[dynamichost]
+;host=dynamic
+;secret=mysecret
+;mailbox=1234		; Notify about mailbox 1234
+;inkeys=key1:key2
+;peercontext=local	; Default context to request for calls to peer
+;defaultip=216.207.245.34
+;callerid="Some Host" <(256) 428-6011>
+
+;[biggateway]
+;type=peer
+;host=192.168.0.1
+;description=Gateway to PSTN
+;context=*
+;secret=myscret
+;trunk=yes			; Use IAX2 trunking with this host
+;timezone=America/New_York	; Set a timezone for the date/time IE
+
+;
+; Friends are a shortcut for creating a user and a peer with the same values.
+;
+
+;[marko]
+;type=friend
+;host=dynamic
+;regexten=1234
+;secret=moofoo   ; Multiple secrets may be specified. For a "user", all
+;secret=foomoo   ; specified entries will be accepted as valid. For a "peer",
+;secret=shazbot  ; only the last specified secret will be used.
+;context=default
+;permit=0.0.0.0/0.0.0.0
+;acl=example_named_acl
+
+;
+; With immediate=yes, an IAX2 phone or a phone on an IAXy acts as a hot-line
+; which goes immediately to the s extension when picked up.  Useful for
+; elevator phones, manual service, or other similar applications.
+;
+;[manual]
+;type=friend
+;host=dynamic
+;immediate=yes  ; go immediately to s extension when picked up
+;secret=moofoo	; when immediate=yes is specified, secret is required
+;context=number-please ; we start at the s extension in this context
+;

asterisk/iaxprov.conf

@@ -0,0 +1,81 @@
+;
+; IAX2 Provisioning Information
+;
+; Contains provisioning information for templates and for specific service
+; entries.
+;
+; 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).
+;
+; Templates which begin with 'si-' are used for provisioning units with
+; specific service identifiers.  For example the entry "si-000364000126"
+; would be used when the device with the corresponding service identifier of
+; "000364000126" attempts to register or make a call.
+;
+[default]
+;
+; The port number the device should use to bind to.  The default is 4569.
+;
+;port=4569
+;
+; server is our PRIMARY server for registration and placing calls
+;
+;server=192.168.69.3
+;
+; altserver is the BACKUP server for registration and placing calls in the
+; event the primary server is unavailable.
+;
+;altserver=192.168.69.4
+;
+; port is the port number to use for IAX2 outbound.  The connections to the
+; server and altserver -- default is of course 4569.
+;serverport=4569
+;
+; language is the preferred language for the device
+;
+;language=en
+;
+; codec is the requested codec.  The iaxy supports ulaw and adpcm
+;
+codec=ulaw
+;
+; flags is a comma separated list of flags which the device should
+; use and may contain any of the following keywords:
+;
+;  "register"  - Register with server
+;  "secure"    - Do not accept calls / provisioning not originated by the server
+;  "heartbeat" - Generate status packets on port 9999 sent to 255.255.255.255
+;  "debug"     - Output extra debugging to port 9999
+;
+;  Note that use can use += and -= to adjust parameters
+;
+flags=register,heartbeat
+;
+; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of this parameter.
+;tos=ef
+;
+; Example iaxy provisioning
+;
+;[si-000364000126]
+;user=iaxy
+;pass=bitsy
+;flags += debug
+
+;[si-000364000127]
+;user=iaxy2
+;pass=bitsy2
+;template=si-000364000126
+;flags += debug
+
+;
+;[*]
+;
+;  If specified, the '*' provisioning is used for all devices which do not
+;  have another provisioning entry within the file.  If unspecified, no
+;  provisioning will take place for devices which have no entry.  DO NOT
+;  USE A '*' PROVISIONING ENTRY UNLESS YOU KNOW WHAT YOU'RE DOING.
+;
+;template=default
+

asterisk/indications.conf

@@ -0,0 +1,736 @@
+;
+; indications.conf
+;
+; Configuration file for location specific tone indications
+;
+
+;
+; NOTE:
+;    When adding countries to this file, please keep them in alphabetical
+;    order according to the 2-character country codes!
+;
+; The [general] category is for certain global variables.
+; All other categories are interpreted as location specific indications
+;
+
+[general]
+country=us		; default location
+
+
+; [example]
+; description = string
+;      The full name of your country, in English.
+; ringcadence = num[,num]*
+;      List of durations the physical bell rings.
+; dial = tonelist
+;      Set of tones to be played when one picks up the hook.
+; busy = tonelist
+;      Set of tones played when the receiving end is busy.
+; congestion = tonelist
+;      Set of tones played when there is some congestion (on the network?)
+; callwaiting = tonelist
+;      Set of tones played when there is a call waiting in the background.
+; dialrecall = tonelist
+;      Not well defined; many phone systems play a recall dial tone after hook
+;      flash.
+; record = tonelist
+;      Set of tones played when call recording is in progress.
+; info = tonelist
+;      Set of tones played with special information messages (e.g., "number is
+;      out of service")
+; 'name' = tonelist
+;      Every other variable will be available as a shortcut for the "PlayList" command
+;      but will not be used automatically by Asterisk.
+;
+;
+; The tonelist itself is defined by a comma-separated sequence of elements.
+; Each element consist of a frequency (f) with an optional duration (in ms)
+; attached to it (f/duration). The frequency component may be a mixture of two
+; frequencies (f1+f2) or a frequency modulated by another frequency (f1*f2).
+; The implicit modulation depth is fixed at 90%, though.
+; If the list element starts with a !, that element is NOT repeated,
+; therefore, only if all elements start with !, the tonelist is time-limited,
+; all others will repeat indefinitely.
+;
+; concisely:
+;   element = [!]freq[+|*freq2][/duration]
+;   tonelist = element[,element]*
+;
+
+[at]
+description = Austria
+ringcadence = 1000,5000
+; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
+dial = 420
+busy = 420/400,0/400
+ring = 420/1000,0/5000
+congestion = 420/200,0/200
+callwaiting = 420/40,0/1960
+dialrecall = 420
+; RECORDTONE - not specified
+record = 1400/80,0/14920
+info = 950/330,1450/330,1850/330,0/1000
+stutter = 380+420
+
+[au]
+description = Australia
+; Reference http://www.acif.org.au/__data/page/3303/S002_2001.pdf
+; Normal Ring
+ringcadence = 400,200,400,2000
+; Distinctive Ring 1 - Forwarded Calls
+; 400,400,200,200,400,1400
+; Distinctive Ring 2 - Selective Ring 2 + Operator + Recall
+; 400,400,200,2000
+; Distinctive Ring 3 - Multiple Subscriber Number 1
+; 200,200,400,2200
+; Distinctive Ring 4 - Selective Ring 1 + Centrex
+; 400,2600
+; Distinctive Ring 5 - Selective Ring 3
+; 400,400,200,400,200,1400
+; Distinctive Ring 6 - Multiple Subscriber Number 2
+; 200,400,200,200,400,1600
+; Distinctive Ring 7 - Multiple Subscriber Number 3 + Data Privacy
+; 200,400,200,400,200,1600
+; Tones
+dial = 413+438
+busy = 425/375,0/375
+ring = 413+438/400,0/200,413+438/400,0/2000
+; XXX Congestion: Should reduce by 10 db every other cadence XXX
+congestion = 425/375,0/375,420/375,0/375
+callwaiting = 425/200,0/200,425/200,0/4400
+dialrecall = 413+438
+; Record tone used for Call Intrusion/Recording or Conference
+record = !425/1000,!0/15000,425/360,0/15000
+info = 425/2500,0/500
+; Other Australian Tones
+; The STD "pips" indicate the call is not an untimed local call
+std = !525/100,!0/100,!525/100,!0/100,!525/100,!0/100,!525/100,!0/100,!525/100
+; Facility confirmation tone (eg. Call Forward Activated)
+facility = 425
+; Message Waiting "stutter" dialtone
+stutter = 413+438/100,0/40
+; Ringtone for calls to Telstra mobiles
+ringmobile = 400+450/400,0/200,400+450/400,0/2000
+
+[bg]
+; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
+description = Bulgaria
+ringcadence = 1000,4000
+;
+dial = 425
+busy = 425/500,0/500
+ring = 425/1000,0/4000
+congestion = 425/250,0/250
+callwaiting = 425/150,0/150,425/150,0/4000
+dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
+record = 1400/425,0/15000
+info = 950/330,1400/330,1800/330,0/1000
+stutter = 425/1500,0/100
+
+[br]
+description = Brazil
+ringcadence = 1000,4000
+dial = 425
+busy = 425/250,0/250
+ring = 425/1000,0/4000
+congestion = 425/250,0/250,425/750,0/250
+callwaiting = 425/50,0/1000
+; Dialrecall not used in Brazil standard (using UK standard)
+dialrecall = 350+440
+; Record tone is not used in Brazil, use busy tone
+record = 425/250,0/250
+; Info not used in Brazil standard (using UK standard)
+info = 950/330,1400/330,1800/330
+stutter = 350+440
+
+[be]
+description = Belgium
+; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
+ringcadence = 1000,3000
+dial = 425
+busy = 425/500,0/500
+ring = 425/1000,0/3000
+congestion = 425/167,0/167
+callwaiting = 1400/175,0/175,1400/175,0/3500
+; DIALRECALL - not specified
+dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
+; RECORDTONE - not specified
+record = 1400/500,0/15000
+info = 900/330,1400/330,1800/330,0/1000
+stutter = 425/1000,0/250
+
+[ch]
+description = Switzerland
+; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
+ringcadence = 1000,4000
+dial = 425
+busy = 425/500,0/500
+ring = 425/1000,0/4000
+congestion = 425/200,0/200
+callwaiting = 425/200,0/200,425/200,0/4000
+; DIALRECALL - not specified
+dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
+; RECORDTONE - not specified
+record = 1400/80,0/15000
+info = 950/330,1400/330,1800/330,0/1000
+stutter = 425+340/1100,0/1100
+
+[cl]
+description = Chile
+; According to specs from Telefonica CTC Chile
+ringcadence = 1000,3000
+dial = 400
+busy = 400/500,0/500
+ring = 400/1000,0/3000
+congestion = 400/200,0/200
+callwaiting = 400/250,0/8750
+dialrecall = !400/100,!0/100,!400/100,!0/100,!400/100,!0/100,400
+record = 1400/500,0/15000
+info = 950/333,1400/333,1800/333,0/1000
+stutter = !400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,400
+
+[cn]
+description = China
+; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
+ringcadence = 1000,4000
+dial = 450
+busy = 450/350,0/350
+ring = 450/1000,0/4000
+congestion = 450/700,0/700
+callwaiting = 450/400,0/4000
+dialrecall = 450
+record = 950/400,0/10000
+info = 450/100,0/100,450/100,0/100,450/100,0/100,450/400,0/400
+; STUTTER - not specified
+stutter = 450+425
+
+[cz]
+description = Czech Republic
+; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
+ringcadence = 1000,4000
+dial = 425/330,0/330,425/660,0/660
+busy = 425/330,0/330
+ring = 425/1000,0/4000
+congestion = 425/165,0/165
+callwaiting = 425/330,0/9000
+; DIALRECALL - not specified
+dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425/330,0/330,425/660,0/660
+; RECORDTONE - not specified
+record = 1400/500,0/14000
+info = 950/330,0/30,1400/330,0/30,1800/330,0/1000
+; STUTTER - not specified
+stutter = 425/450,0/50
+
+[de]
+description = Germany
+; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
+ringcadence = 1000,4000
+dial = 425
+busy = 425/480,0/480
+ring = 425/1000,0/4000
+congestion = 425/240,0/240
+callwaiting = !425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,0
+; DIALRECALL - not specified
+dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
+; RECORDTONE - not specified
+record = 1400/80,0/15000
+info = 950/330,1400/330,1800/330,0/1000
+stutter = 425+400
+
+[dk]
+description = Denmark
+; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
+ringcadence = 1000,4000
+dial = 425
+busy = 425/500,0/500
+ring = 425/1000,0/4000
+congestion = 425/200,0/200
+callwaiting = !425/200,!0/600,!425/200,!0/3000,!425/200,!0/200,!425/200,0
+; DIALRECALL - not specified
+dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
+; RECORDTONE - not specified
+record = 1400/80,0/15000
+info = 950/330,1400/330,1800/330,0/1000
+; STUTTER - not specified
+stutter = 425/450,0/50
+
+[ee]
+description = Estonia
+; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
+ringcadence = 1000,4000
+dial = 425
+busy = 425/300,0/300
+ring = 425/1000,0/4000
+congestion = 425/200,0/200
+; CALLWAIT not in accordance to ITU
+callwaiting = 950/650,0/325,950/325,0/30,1400/1300,0/2600
+; DIALRECALL - not specified
+dialrecall = 425/650,0/25
+; RECORDTONE - not specified
+record = 1400/500,0/15000
+; INFO not in accordance to ITU
+info = 950/650,0/325,950/325,0/30,1400/1300,0/2600
+; 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
+
+[es]
+description = Spain
+ringcadence = 1500,3000
+dial = 425
+busy = 425/200,0/200
+ring = 425/1500,0/3000
+congestion = 425/200,0/200,425/200,0/200,425/200,0/600
+callwaiting = 425/175,0/175,425/175,0/3500
+dialrecall = !425/200,!0/200,!425/200,!0/200,!425/200,!0/200,425
+record = 1400/500,0/15000
+info = 950/330,0/1000
+dialout = 500
+; 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
+
+
+[fi]
+description = Finland
+ringcadence = 1000,4000
+dial = 425
+busy = 425/300,0/300
+ring = 425/1000,0/4000
+congestion = 425/200,0/200
+callwaiting = 425/150,0/150,425/150,0/8000
+dialrecall = 425/650,0/25
+record = 1400/500,0/15000
+info = 950/650,0/325,950/325,0/30,1400/1300,0/2600
+stutter = 425/650,0/25
+
+[fr]
+description = France
+; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
+ringcadence = 1500,3500
+; Dialtone can also be 440+330
+dial = 440
+busy = 440/500,0/500
+ring = 440/1500,0/3500
+; CONGESTION - not specified
+congestion = 440/250,0/250
+callwait = 440/300,0/10000
+; DIALRECALL - not specified
+dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
+; RECORDTONE - not specified
+record = 1400/500,0/15000
+info = !950/330,!1400/330,!1800/330
+stutter = !440/100,!0/100,!440/100,!0/100,!440/100,!0/100,!440/100,!0/100,!440/100,!0/100,!440/100,!0/100,440
+
+[gr]
+description = Greece
+ringcadence = 1000,4000
+dial = 425/200,0/300,425/700,0/800
+busy = 425/300,0/300
+ring = 425/1000,0/4000
+congestion = 425/200,0/200
+callwaiting = 425/150,0/150,425/150,0/8000
+dialrecall = 425/650,0/25
+record = 1400/400,0/15000
+info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0
+stutter = 425/650,0/25
+
+[hu]
+description = Hungary
+; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
+ringcadence = 1250,3750
+dial = 425
+busy = 425/300,0/300
+ring = 425/1250,0/3750
+congestion = 425/300,0/300
+callwaiting = 425/40,0/1960
+dialrecall = 425+450
+; RECORDTONE - not specified
+record = 1400/400,0/15000
+info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0
+stutter = 350+375+400
+
+[il]
+description = Israel
+ringcadence = 1000,3000
+dial = 414
+busy = 414/500,0/500
+ring = 414/1000,0/3000
+congestion = 414/250,0/250
+callwaiting = 414/100,0/100,414/100,0/100,414/600,0/3000
+dialrecall = !414/100,!0/100,!414/100,!0/100,!414/100,!0/100,414
+record = 1400/500,0/15000
+info = 1000/330,1400/330,1800/330,0/1000
+stutter = !414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,414
+
+
+[in]
+description = India
+ringcadence = 400,200,400,2000
+dial = 400*25
+busy = 400/750,0/750
+ring = 400*25/400,0/200,400*25/400,0/2000
+congestion = 400/250,0/250
+callwaiting = 400/200,0/100,400/200,0/7500
+dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
+record = 1400/500,0/15000
+info = !950/330,!1400/330,!1800/330,0/1000
+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,!0/100,400*25
+
+[it]
+description = Italy
+; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
+ringcadence = 1000,4000
+dial = 425/200,0/200,425/600,0/1000
+busy = 425/500,0/500
+ring = 425/1000,0/4000
+congestion = 425/200,0/200
+callwaiting = 425/400,0/100,425/250,0/100,425/150,0/14000
+dialrecall = 470/400,425/400
+record = 1400/400,0/15000
+info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0
+stutter = 470/400,425/400
+
+[lt]
+description = Lithuania
+ringcadence = 1000,4000
+dial = 425
+busy = 425/350,0/350
+ring = 425/1000,0/4000
+congestion = 425/200,0/200
+callwaiting = 425/150,0/150,425/150,0/4000
+; DIALRECALL - not specified
+dialrecall = 425/500,0/50
+; RECORDTONE - not specified
+record = 1400/500,0/15000
+info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0
+; 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
+
+[jp]
+description = Japan
+ringcadence = 1000,2000
+dial = 400
+busy = 400/500,0/500
+ring = 400+15/1000,0/2000
+congestion = 400/500,0/500
+callwaiting = 400+16/500,0/8000
+dialrecall = !400/200,!0/200,!400/200,!0/200,!400/200,!0/200,400
+record = 1400/500,0/15000
+info = !950/330,!1400/330,!1800/330,0
+stutter = !400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,400
+
+[mx]
+description = Mexico
+ringcadence = 2000,4000
+dial = 425
+busy = 425/250,0/250
+ring = 425/1000,0/4000
+congestion = 425/250,0/250
+callwaiting = 425/200,0/600,425/200,0/10000
+dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
+record = 1400/500,0/15000
+info = 950/330,0/30,1400/330,0/30,1800/330,0/1000
+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,!0/100,425
+
+[my]
+description = Malaysia
+ringcadence = 2000,4000
+dial = 425
+busy = 425/500,0/500
+ring = 425/400,0/200,425/400,0/2000
+congestion = 425/500,0/500
+; 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
+
+[nl]
+description = Netherlands
+; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
+ringcadence = 1000,4000
+; Most of these 425's can also be 450's
+dial = 425
+busy = 425/500,0/500
+ring = 425/1000,0/4000
+congestion = 425/250,0/250
+callwaiting = 425/500,0/9500
+; DIALRECALL - not specified
+dialrecall = 425/500,0/50
+; RECORDTONE - not specified
+record = 1400/500,0/15000
+info = 950/330,1400/330,1800/330,0/1000
+stutter = 425/500,0/50
+
+[no]
+description = Norway
+ringcadence = 1000,4000
+dial = 425
+busy = 425/500,0/500
+ring = 425/1000,0/4000
+congestion = 425/200,0/200
+callwaiting = 425/200,0/600,425/200,0/10000
+dialrecall = 470/400,425/400
+record = 1400/400,0/15000
+info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0
+stutter = 470/400,425/400
+
+[nz]
+description = New Zealand
+; Reference = http://www.telepermit.co.nz/TNA102.pdf
+ringcadence = 400,200,400,2000
+dial = 400
+busy = 400/500,0/500
+ring = 400+450/400,0/200,400+450/400,0/2000
+congestion = 400/250,0/250
+callwaiting = !400/200,!0/3000,!400/200,!0/3000,!400/200,!0/3000,!400/200
+dialrecall = !400/100,!0/100,!400/100,!0/100,!400/100,!0/100,400
+record = 1400/425,0/15000
+info = 400/750,0/100,400/750,0/100,400/750,0/100,400/750,0/400
+stutter = !400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,400
+unobtainable = 400/75,0/100,400/75,0/100,400/75,0/100,400/75,0/400
+
+[ph]
+
+; reference http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
+
+description = Philippines
+ringcadence = 1000,4000
+dial = 425
+busy = 480+620/500,0/500
+ring = 425+480/1000,0/4000
+congestion = 480+620/250,0/250
+callwaiting = 440/300,0/10000
+; DIALRECALL - not specified
+dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
+; RECORDTONE - not specified
+record = 1400/500,0/15000
+; INFO - not specified
+info = !950/330,!1400/330,!1800/330,0
+; STUTTER - not specified
+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,!0/100,425
+
+
+[pl]
+description = Poland
+ringcadence = 1000,4000
+dial = 425
+busy = 425/500,0/500
+ring = 425/1000,0/4000
+congestion = 425/500,0/500
+callwaiting = 425/150,0/150,425/150,0/4000
+; DIALRECALL - not specified
+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
+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
+
+[pt]
+description = Portugal
+ringcadence = 1000,5000
+dial = 425
+busy = 425/500,0/500
+ring = 425/1000,0/5000
+congestion = 425/200,0/200
+callwaiting = 440/300,0/10000
+dialrecall = 425/1000,0/200
+record = 1400/500,0/15000
+info = 950/330,1400/330,1800/330,0/1000
+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
+
+[ru]
+; References:
+;	http://www.minsvyaz.ru/site.shtml?id=1806
+;	http://www.aboutphone.info/lib/gost/45-223-2001.html
+description = Russian Federation / ex Soviet Union
+ringcadence = 1000,4000
+dial = 425
+busy = 425/350,0/350
+ring = 425/1000,0/4000
+congestion = 425/175,0/175
+callwaiting = 425/200,0/5000
+record = 1400/400,0/15000
+info = 950/330,1400/330,1800/330,0/1000
+dialrecall = 425/400,0/40
+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
+
+[se]
+description = Sweden
+ringcadence = 1000,5000
+dial = 425
+busy = 425/250,0/250
+ring = 425/1000,0/5000
+congestion = 425/250,0/750
+callwaiting = 425/200,0/500,425/200,0/9100
+dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
+record = 1400/500,0/15000
+info = !950/332,!0/24,!1400/332,!0/24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/332,0
+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
+; stutter = 425/320,0/20        ; Real swedish standard, not used for now
+
+[sg]
+description = Singapore
+; Singapore
+; 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
+ring        = 425*24/400,0/200,425*24/400,0/2000     ; modulation should be 100%, not 90%
+busy        = 425/750,0/750
+congestion  = 425/250,0/250
+callwaiting = 425*24/300,0/200,425*24/300,0/3200
+stutter     = !425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,425
+info        = 950/330,1400/330,1800/330,0/1000       ; not currently in use acc. to reference
+dialrecall  = 425*24/500,0/500,425/500,0/2500        ; unspecified in IDA reference, use repeating Holding Tone A,B
+record      = 1400/500,0/15000                       ; unspecified in IDA reference, use 0.5s tone every 15s
+; additionally defined in reference
+nutone      = 425/2500,0/500
+intrusion   = 425/250,0/2000
+warning     = 425/624,0/4376                         ; end of period tone, warning
+acceptance  = 425/125,0/125
+holdinga    = !425*24/500,!0/500                     ; followed by holdingb
+holdingb    = !425/500,!0/2500
+
+[th]
+description = Thailand
+ringcadence = 1000,4000
+; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
+dial = 400*50
+busy = 400/500,0/500
+ring = 420/1000,0/5000
+congestion = 400/300,0/300
+callwaiting = 1000/400,10000/400,1000/400
+; DIALRECALL - not specified - use special dial tone instead.
+dialrecall = 400*50/400,0/100,400*50/400,0/100
+; RECORDTONE - not specified
+record = 1400/500,0/15000
+; INFO - specified as an announcement - use special information tones instead
+info = 950/330,1400/330,1800/330
+; STUTTER - not specified
+stutter = !400/200,!0/200,!400/600,!0/200,!400/200,!0/200,!400/600,!0/200,!400/200,!0/200,!400/600,!0/200,!400/200,!0/200,!400/600,!0/200,400
+
+[uk]
+description = United Kingdom
+ringcadence = 400,200,400,2000
+; These are the official tones taken from BT SIN350. The actual tones
+; used by BT include some volume differences so sound slightly different
+; from Asterisk-generated ones.
+dial = 350+440
+; Special dial is the intermittent dial tone heard when, for example,
+; you have a divert active on the line
+specialdial = 350+440/750,440/750
+; Busy is also called "Engaged"
+busy = 400/375,0/375
+; "Congestion" is the Beep-bip engaged tone
+congestion = 400/400,0/350,400/225,0/525
+; "Special Congestion" is not used by BT very often if at all
+specialcongestion = 400/200,1004/300
+unobtainable = 400
+ring = 400+450/400,0/200,400+450/400,0/2000
+callwaiting = 400/100,0/4000
+; BT seem to use "Special Call Waiting" rather than just "Call Waiting" tones
+specialcallwaiting = 400/250,0/250,400/250,0/250,400/250,0/5000
+; "Pips" used by BT on payphones. (Sounds wrong, but this is what BT claim it
+; is and I've not used a payphone for years)
+creditexpired = 400/125,0/125
+; These two are used to confirm/reject service requests on exchanges that
+; don't do voice announcements.
+confirm = 1400
+switching = 400/200,0/400,400/2000,0/400
+; This is the three rising tones Doo-dah-dee "Special Information Tone",
+; usually followed by the BT woman saying an appropriate message.
+info = 950/330,0/15,1400/330,0/15,1800/330,0/1000
+; Not listed in SIN350
+record = 1400/500,0/60000
+stutter = 350+440/750,440/750
+
+[us]
+description = United States / North America
+ringcadence = 2000,4000
+dial = 350+440
+busy = 480+620/500,0/500
+ring = 440+480/2000,0/4000
+congestion = 480+620/250,0/250
+callwaiting = 440/300,0/10000
+dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
+record = 1400/500,0/15000
+info = !950/330,!1400/330,!1800/330,0
+stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
+
+[us-old]
+description = United States Circa 1950/ North America
+ringcadence = 2000,4000
+dial = 600*120
+busy = 500*100/500,0/500
+ring = 420*40/2000,0/4000
+congestion = 500*100/250,0/250
+callwaiting = 440/300,0/10000
+dialrecall = !600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,600*120
+record = 1400/500,0/15000
+info = !950/330,!1400/330,!1800/330,0
+stutter = !600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,600*120
+
+[tw]
+description = Taiwan
+; http://nemesis.lonestar.org/reference/telecom/signaling/dialtone.html
+; http://nemesis.lonestar.org/reference/telecom/signaling/busy.html
+; http://www.iproducts.com.tw/ee/kylink/06ky-1000a.htm
+; http://www.pbx-manufacturer.com/ky120dx.htm
+; http://www.nettwerked.net/tones.txt
+; http://www.cisco.com/univercd/cc/td/doc/product/tel_pswt/vco_prod/taiw_sup/taiw2.htm
+;
+; busy tone 480+620Hz 0.5 sec. on ,0.5 sec. off
+; reorder tone 480+620Hz 0.25 sec. on,0.25 sec. off
+; ringing tone 440+480Hz 1 sec. on ,2 sec. off
+;
+ringcadence = 1000,4000
+dial = 350+440
+busy = 480+620/500,0/500
+ring = 440+480/1000,0/2000
+congestion = 480+620/250,0/250
+callwaiting = 350+440/250,0/250,350+440/250,0/3250
+dialrecall = 300/1500,0/500
+record = 1400/500,0/15000
+info = !950/330,!1400/330,!1800/330,0
+stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
+
+[ve]
+; 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
+dial = 425
+busy = 425/500,0/500
+ring = 425/1000,0/4000
+congestion = 425/250,0/250
+callwaiting = 400+450/300,0/6000
+dialrecall = 425
+record = 1400/500,0/15000
+info = !950/330,!1440/330,!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
+
+
+[za]
+description = South Africa
+; http://www.cisco.com/univercd/cc/td/doc/product/tel_pswt/vco_prod/safr_sup/saf02.htm
+; (definitions for other countries can also be found there)
+; Note, though, that South Africa uses two switch types in their network --
+; Alcatel switches -- mainly in the Western Cape, and Siemens elsewhere.
+; The former use 383+417 in dial, ringback etc.  The latter use 400*33
+; I've provided both, uncomment the ones you prefer
+ringcadence = 400,200,400,2000
+; dial/ring/callwaiting for the Siemens switches:
+dial = 400*33
+ring = 400*33/400,0/200,400*33/400,0/2000
+callwaiting = 400*33/250,0/250,400*33/250,0/250,400*33/250,0/250,400*33/250,0/250
+; dial/ring/callwaiting for the Alcatel switches:
+; dial = 383+417
+; ring = 383+417/400,0/200,383+417/400,0/2000
+; callwaiting = 383+417/250,0/250,383+417/250,0/250,383+417/250,0/250,383+417/250,0/250
+congestion = 400/250,0/250
+busy = 400/500,0/500
+dialrecall = 350+440
+; XXX Not sure about the RECORDTONE
+record = 1400/500,0/10000
+info = 950/330,1400/330,1800/330,0/330
+stutter = !400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,400*33

asterisk/jabber.conf

@@ -0,0 +1,39 @@
+[general]
+;debug=yes				; Enable debugging (disabled 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'.
+;autoregister=yes			; Auto register users from buddy list.
+;collection_nodes=yes			; Enable support for XEP-0248 for use with
+					; distributed device state.  Default is 'no'.
+;pubsub_autocreate=yes			; Whether or not the PubSub server supports/is using
+					; auto-create for nodes.  If it is, we have to
+					; explicitly pre-create nodes before publishing them.
+					; Default is 'no'.
+;auth_policy=accept			; Auto accept users' subscription requests (default).
+					; Set to deny for auto denial.
+;[asterisk]
+;type=client				; Client or Component connection
+;serverhost=astjab.org			; Route to server for example, talk.google.com
+;pubsub_node=pubsub.astjab.org		; Node to use for publishing events via PubSub
+;username=asterisk@astjab.org/asterisk	; Username with optional resource.
+;secret=blah				; Password
+;priority=1				; Resource priority
+;port=5222				; Port to use defaults to 5222
+;usetls=yes				; Use tls or not
+;usesasl=yes				; Use sasl or not
+;buddy=mogorman@astjab.org		; Manual addition of buddy to list.
+					; For distributed events, these buddies are
+					; automatically added in the whitelist as
+					; 'owners' of the node(s).
+;distribute_events=yes			; Whether or not to distribute events using
+					; this connection.  Default is 'no'.
+;status=available			; One of: chat, available, away, xaway, or dnd
+;statusmessage="I am available"		; Have custom status message for Asterisk
+;timeout=5				; Timeout (in seconds) on the message stack, defaults to 5.
+					; Messages stored longer than this value will be deleted by Asterisk.
+					; This option applies to incoming messages only, which are intended to
+					; be processed by the JABBER_RECEIVE dialplan function.
+;sendtodialplan=yes			; Send incoming messages into the dialplan.  Off by default.
+;context=messages			; Dialplan context to send incoming messages to.  If not set,
+					; "default" will be used.

asterisk/jingle.conf

@@ -0,0 +1,20 @@
+;[general]
+;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
+;
+;[guest]			;;special account for options on guest account
+;disallow=all
+;allow=ulaw
+;context=guest
+;
+;[ogorman]
+;username=ogorman@gmail.com	;;username of the peer your
+				;;calling or accepting calls from
+;disallow=all
+;allow=ulaw
+;context=default
+;connection=asterisk		;;client or component in jabber.conf
+				;;for the call to leave on.
+;

asterisk/logger.conf

@@ -0,0 +1,127 @@
+;
+; Logging Configuration
+;
+; In this file, you configure logging to files or to
+; the syslog system.
+;
+; "logger reload" at the CLI will reload configuration
+; of the logging system.
+
+[general]
+;
+; Customize the display of debug message time stamps
+; this example is the ISO 8601 date format (yyyy-mm-dd HH:MM:SS)
+;
+; 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
+;
+;
+; This makes Asterisk write callids to log messages
+; (defaults to yes)
+;use_callids = no
+;
+; This appends the hostname to the name of the log files.
+;appendhostname = yes
+;
+; This determines whether or not we log queue events to a file
+; (defaults to yes).
+;queue_log = no
+;
+; Determines whether the queue_log always goes to a file, even
+; when a realtime backend is present (defaults to no).
+;queue_log_to_file = yes
+;
+; Set the queue_log filename
+; (defaults to queue_log)
+;queue_log_name = queue_log
+;
+; Log rotation strategy:
+; sequential:  Rename archived logs in order, such that the newest
+;              has the highest sequence number [default].  When
+;              exec_after_rotate is set, ${filename} will specify
+;              the new archived logfile.
+; rotate:  Rotate all the old files, such that the oldest has the
+;          highest sequence number [this is the expected behavior
+;          for Unix administrators].  When exec_after_rotate is
+;          set, ${filename} will specify the original root filename.
+; timestamp:  Rename the logfiles using a timestamp instead of a
+;             sequence number when "logger rotate" is executed.
+;             When exec_after_rotate is set, ${filename} will
+;             specify the new archived logfile.
+;rotatestrategy = rotate
+;
+; Run a system command after rotating the files.  This is mainly
+; useful for rotatestrategy=rotate. The example allows the last
+; two archive files to remain uncompressed, but after that point,
+; they are compressed on disk.
+;
+; exec_after_rotate=gzip -9 ${filename}.2
+;
+;
+; For each file, specify what to log.
+;
+; For console logging, you set options at start of
+; Asterisk with -v for verbose and -d for debug
+; See 'asterisk -h' for more information.
+;
+; Directory for log files is configures in asterisk.conf
+; option astlogdir
+;
+[logfiles]
+;
+; Format is "filename" and then "levels" of debugging to be included:
+;    debug
+;    notice
+;    warning
+;    error
+;    verbose(<level>)
+;    dtmf
+;    fax
+;    security
+;
+; Special filename "console" represents the root console
+;
+; Filenames can either be relative to the standard Asterisk log directory
+; (see 'astlogdir' in asterisk.conf), or absolute paths that begin with
+; '/'.
+;
+; Verbose takes an optional argument, in the form of an integer level.
+; Verbose messages with higher levels will not be logged to the file.  If
+; the verbose level is not specified, it will log verbose messages following
+; the current level of the root console.
+;
+; Special level name "*" means all levels, even dynamic levels registered
+; by modules after the logger has been initialized (this means that loading
+; and unloading modules that create/remove dynamic logger levels will result
+; in these levels being included on filenames that have a level name of "*",
+; without any need to perform a 'logger reload' or similar operation).
+; Note that there is no value in specifying both "*" and specific level names
+; for a filename; the "*" level means all levels.  The only exception is if
+; you need to specify a specific verbose level. e.g, "verbose(3),*".
+;
+; We highly recommend that you DO NOT turn on debug mode if you are simply
+; running a production system.  Debug mode turns on a LOT of extra messages,
+; most of which you are unlikely to understand without an understanding of
+; the underlying code.  Do NOT report debug messages as code issues, unless
+; you have a specific issue that you are attempting to debug.  They are
+; messages for just that -- debugging -- and do not rise to the level of
+; something that merit your attention as an Asterisk administrator.  Debug
+; messages are also very verbose and can and do fill up logfiles quickly;
+; this is another reason not to have debug mode on a production system unless
+; you are in the process of debugging a specific issue.
+;
+;debug => debug
+;security => security
+console => notice,warning,error
+;console => notice,warning,error,debug
+messages => notice,warning,error
+;full => notice,warning,error,debug,verbose,dtmf,fax
+
+;syslog keyword : This special keyword logs to syslog facility
+;
+;syslog.local0 => notice,warning,error
+;

asterisk/manager.conf

@@ -0,0 +1,153 @@
+;
+; AMI - The Asterisk Manager Interface
+;
+; Third party application call management support and PBX event supervision
+;
+; Use the "manager show commands" at the CLI to list available manager commands
+; and their authorization levels.
+;
+; "manager show command <command>" will show a help text.
+;
+; ---------------------------- 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
+; 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
+; default is 60 seconds.
+;
+[general]
+enabled = no
+;webenabled = yes
+
+port = 5038
+bindaddr = 0.0.0.0
+
+; Parameters that control AMI over TLS. ("enabled" must be set too).
+; You can open a connection to this socket with e.g.
+;
+;	openssl s_client -connect my_host:5039
+;
+;tlsenable=no		; set to YES to enable it
+;tlsbindaddr=0.0.0.0:5039		; address and port to bind to, default to bindaddr and port 5039
+;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.
+;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.
+;                               ; The default is yes.
+;
+;displayconnects = yes
+;
+; Add a Unix epoch timestamp to events (not action responses)
+;
+;timestampevents = yes
+
+;brokeneventsaction = yes   ; Restore previous behavior that caused the events
+                            ; action to not return a response in certain
+                            ; circumstances.  Defaults to 'no'.
+
+;
+; Display certain channel variables every time a channel-oriented
+; event is emitted:
+;
+;channelvars = var1,var2,var3
+
+; debug = on	; enable some debugging info in AMI messages (default off).
+		; Also accessible through the "manager debug" CLI command.
+
+; authtimeout specifies the maximum number of seconds a client has to
+; authenticate.  If the client does not authenticate beofre this timeout
+; expires, the client will be disconnected. (default: 30 seconds)
+
+;authtimeout = 30
+
+; authlimit specifies the maximum number of unauthenticated sessions that will
+; be allowed to connect at any given time.
+
+;authlimit = 50
+
+;httptimeout = 60
+; a) httptimeout sets the Max-Age of the http cookie
+; 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
+;    a http session alive after completing a successful action
+
+
+;[mark]
+;secret = mysecret
+;deny=0.0.0.0/0.0.0.0
+;permit=209.16.236.73/255.255.255.0
+;acl=named_acl_example               ; use a named ACL from acl.conf
+;
+; 
+;setvar=PBXACCOUNT=edvina
+; The setvar option defines channel variables that will be set when this account
+; originates a call. You can define multiple setvar= commands for one manager
+; user.
+;
+;eventfilter=Event: Newchannel
+;eventfilter=!Channel: DAHDI.*
+; The eventfilter option is used to whitelist or blacklist events per user.
+; A filter consists of a (basic/old-style and unanchored) regular expression
+; that is run on the entire event data. If the first character of the filter
+; is an exclamation mark (!), the filter is appended to the blacklist instead
+; of the whitelist. After first checking the read access below, the regular
+; expression filters are processed as follows:
+; - If no filters are configured all events are reported as normal.
+; - If there are white filters only: implied black all filter processed first,
+; then white filters.
+; - If there are black filters only: implied white all filter processed first,
+; then black filters.
+; - If there are both white and black filters: implied black all filter processed
+; first, then white filters, and lastly black filters.
+
+;
+; If the device connected via this user accepts input slowly,
+; the timeout for writes to it can be increased to keep it
+; from being disconnected (value is in milliseconds)
+;
+; writetimeout = 100
+;
+;displayconnects = yes	; Display on CLI user login/logoff
+;
+; 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
+; following classes exist:
+;
+; all       - All event classes below (including any we may have missed).
+; system    - General information about the system and ability to run system
+;             management commands, such as Shutdown, Restart, and Reload.
+; call      - Information about channels and ability to set information in a
+;             running channel.
+; log       - Logging information.  Read-only. (Defined but not yet used.)
+; verbose   - Verbose information.  Read-only. (Defined but not yet used.)
+; agent     - Information about queues and agents and ability to add queue
+;             members to a queue.
+; user      - Permission to send and receive UserEvent.
+; config    - Ability to read and write configuration files.
+; command   - Permission to run CLI commands.  Write-only.
+; dtmf      - Receive DTMF events.  Read-only.
+; reporting - Ability to get information about the system.
+; cdr       - Output of cdr_manager, if loaded.  Read-only.
+; dialplan  - Receive NewExten and VarSet events.  Read-only.
+; originate - Permission to originate new calls.  Write-only.
+; agi       - Output AGI commands executed.  Input AGI command to execute.
+; cc        - Call Completion events.  Read-only.
+; aoc       - Permission to send Advice Of Charge messages and receive Advice
+;           - Of Charge events.
+; test      - Ability to read TestEvent notifications sent to the Asterisk Test
+;             Suite.  Note that this is only enabled when the TEST_FRAMEWORK
+;             compiler flag is defined.
+; message   - Permissions to send out of call messages. Write-only
+;
+;read = system,call,log,verbose,agent,user,config,dtmf,reporting,cdr,dialplan
+;write = system,call,agent,user,config,command,reporting,originate,message
+

asterisk/meetme.conf

@@ -0,0 +1,45 @@
+;
+; Configuration file for MeetMe simple conference rooms for Asterisk of course.
+;
+; This configuration file is read every time you call app meetme()
+
+[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.
+;
+; Conferences may be scheduled from realtime?
+;schedule=yes
+;
+; Update realtime when members login/out of the conference
+;logmembercount=no
+;
+; How much earlier than the start time should we allow participants to
+; join the conference (in seconds)?
+;fuzzystart=300
+;
+; If the participants join too early, how much time should we allow
+; to tell them that they've joined too early, rather than telling them
+; the conference simply doesn't exist (in seconds)?
+;earlyalert=3600
+;
+; How many seconds before the scheduled end of the conference should
+; the participants be warned?
+;endalert=120
+;
+[rooms]
+;
+; 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 => 2345,9938

asterisk/mgcp.conf

@@ -0,0 +1,142 @@
+;
+; MGCP Configuration for Asterisk
+;
+[general]
+;port = 2427
+;bindaddr = 0.0.0.0
+
+; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
+;tos=cs3		; Sets TOS for signaling packets.
+;tos_audio=ef		; Sets TOS for RTP audio packets.
+;cos=3			; Sets 802.1p priority for signaling packets.
+;cos_audio=5		; Sets 802.1p priority for RTP audio packets.
+
+;---------------------- DIGIT TIMEOUTS ----------------------------
+firstdigittimeout = 30000 ; default 16000 = 16s
+gendigittimeout = 10000   ; default  8000 = 8s 
+matchdigittimeout = 5000  ; defaults 3000 = 3s
+
+;------------------------------ 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.
+
+; jbforce = no                ; Forces the use of a jitterbuffer on the receive side of a MGCP
+                              ; 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.
+
+; 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.
+
+; jbtargetextra = 40          ; This option only affects the jb when 'jbimpl = adaptive' is set.
+                              ; The option represents the number of milliseconds by which the new
+                              ; jitter buffer will pad its size. the default is 40, so without
+                              ; modification, the new jitter buffer will set its size to the jitter
+                              ; value plus 40 milliseconds. increasing this value may help if your
+                              ; network normally has low jitter, but occasionally has spikes.
+
+; jblog = no                  ; Enables jitterbuffer frame logging. Defaults to "no".
+;-----------------------------------------------------------------------------------
+
+;[dlinkgw]
+;host = 192.168.0.64
+;context = default
+;directmedia = no
+;line => aaln/2
+;line => aaln/1
+
+;; 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 
+;; *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> 
+;callgroup=0      ; in the range from 0 to 63
+;pickupgroup=0    ; in the range from 0 to 63
+;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 
+;
+
+;[dph100]
+;
+; Supporting the DPH100M requires defining DLINK_BUGGY_FIRMWARE in
+; chan_mgcp.c in addition to enabling the slowsequence mode due to
+; bugs in the D-Link firmware
+;
+;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.
+;slowsequence=yes	; The DPH100M does not follow MGCP standards for sequencing
+;line => aaln/1
+
+; known to work with wave7optics FTTH LMGs
+;[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'
+;context	= local
+;host		= 192.168.1.20
+;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
+;cancallforward = yes
+;directmedia	= no
+;transfer	= no
+;dtmfmode	= inband
+;setvar=one=1           ; Set channel variables associated with this incoming line
+;setvar=two=2
+;line => aaln/1			; now lets save this config to line1 aka aaln/1
+;clearvars=all          ; Reset list of variables back to none
+;callerid 	= "Duane Cox" <456>	; now lets setup line 2
+;callwaiting	= no
+;callreturn	= yes
+;cancallforward = yes
+;directmedia	= no
+;transfer	= no
+;dtmfmode	= inband
+;line => aaln/2			; now lets save this config to line2 aka aaln/2
+
+; PacketCable
+;[sbv5121e-mta.test.local]
+;host        = 10.0.1.3
+;callwaiting = 1
+;canreinvite = 1
+;dtmfmode    = rfc2833
+;amaflags    = BILLING
+;ncs         = yes ; Use NCS 1.0 signalling
+;pktcgatealloc = yes ; Allocate DQOS gate on CMTS
+;hangupongateremove = yes ; Hangup the channel if the CMTS close the gate
+;callerid    = 3622622225
+;accountcode = test-3622622225
+;line        = aaln/1
+;callerid    = 3622622226
+;accountcode = test-3622622226
+;line        = aaln/2

asterisk/minivm.conf

@@ -0,0 +1,218 @@
+;
+; Mini-Voicemail Configuration
+; for the MiniVM set of applications
+;
+; MiniVM consists of the following dialplan applications
+;   MinivmGreet       Play personal prompts for busy/unavailable/temporary messages or default prompts
+;   MinivmRecord      Record voice prompts to account directory or default directory
+;   MinivmNotify      Notify via e-mail or pager - with or without attachment
+;   MinivmDelete      Delete voice prompt (filename as argument or channel variable set by MinivmRecord)
+;
+; MiniVM works without accounts (just give e-mail address as argument) or with accounts in
+; this configuration file or realtime. The idea is to build voicemail as building blocks so that
+; a complete and adaptive voicemail system can be built in the dialplan
+;
+;------------------------------ Variables to use in subject, from and message body ------------------
+; 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
+;
+; Note: The emailbody config row can only be up to 512 characters due to a
+;       limitation in the Asterisk configuration subsystem.
+;	To create longer mails, use the templatefile option when creating the template
+;----------------------------------------------------------------------------------------------------
+
+[general]
+; Default format for storing and sending voicemail
+; (only one format. Can also be set on a per-mailbox level)
+format=wav49
+;format=gsm
+;
+;Turn on logfile with the following syntax. One line per voicemail received
+;with minivmRecord()
+; Mailbox:domain:macrocontext:exten:priority:callerchan:callerid:origdate:origtime:duration:durationstatus:accountcode
+;logfile=/var/log/asterisk/minivm.log
+; Who the e-mail notification should appear to come from
+serveremail=asterisk
+;serveremail=asterisk@asterisk.example.com
+; Minimum length of a voicemail message in seconds for the message to be kept
+; The default is no minimum.
+;minmessage=3
+; How many seconds of silence before we end the recording
+maxsilence=10
+; Silence threshold (what we consider silence: the lower, the more sensitive)
+silencethreshold=128
+; How long greeting messages (busy/unavailable/temp/name) are allowed to be, in seconds
+;maxgreet=120
+; If you need to have an external program, i.e. /usr/bin/myapp called when a
+; voicemail is received by the server. The arguments are
+;
+; 	<app> <username@domain> <callerid-number> <callerid-name>
+;
+;externnotify=/usr/bin/myapp
+; The character set for voicemail messages can be specified here
+;charset=ISO-8859-1
+; Skip the "[PBX]:" string from the message title
+;pbxskip=yes
+; Change the From: string
+
+; You can override the default program to send e-mail if you wish, too
+; This is used both for e-mail and pager messages
+;mailcmd=/usr/sbin/sendmail -t
+;
+;--------------Default e-mail message template (used if no templates are used) ------
+;fromstring=The Asterisk PBX
+;
+
+;emailsubject=[PBX]: New message ${MVM_COUNER} in mailbox ${VM_MAILBOX}
+; The following definition is very close to the default, but the default shows
+; just the CIDNAME, if it is not null, otherwise just the CIDNUM, or "an unknown
+; caller", if they are both null.
+;emailbody=Dear ${MVM_NAME}:\n\n\tjust wanted to let you know you were just left a ${MVM_DUR} long message (number ${MVM_COUNTER})\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
+;
+; Set the date format on outgoing mails. Valid arguments can be found on the
+; strftime(3) man page
+;
+; Default
+emaildateformat=%A, %B %d, %Y at %r
+; 24h date format
+;emaildateformat=%A, %d %B %Y at %H:%M:%S
+;
+;--------------Default pager message template (used if no templates are used) ------
+; You can also change the Pager From: string, the pager body and/or subject.
+; The above defined variables also can be used here
+;pagerfromstring=The Asterisk PBX
+;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:
+; '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"
+; N             Minute, with 00 pronounced as "hundred" (US military time)
+; P or p        AM or PM
+; Q             "today", "yesterday" or ABdY
+;               (*note: not standard strftime value)
+; q             "" (for today), "yesterday", weekday, or ABdY
+;               (*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
+
+[zonemessages]
+eastern=America/New_York|'vm-received' Q 'digits/at' IMp
+central=America/Chicago|'vm-received' Q 'digits/at' IMp
+central24=America/Chicago|'vm-received' q 'digits/at' H N 'hours'
+military=Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p'
+
+;----------------------- Message body templates---------------------
+; [template-name]	; "template-" is a verbatim marker
+; fromaddress = Your Friendly Asterisk Server
+; fromemail = asteriskvm@digium.com
+; subject = <string>
+; 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
+;				; See your O/S documentation for proper settings for setlocale()
+; templatefile = <filename>	; File name (relative to Asterisk configuration directory,
+				; or absolute
+; messagebody = Format		; Message body definition with variables
+;
+[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
+dateformat=%A, %d %B %Y at %H:%M:%S
+locale=sv_SE
+
+[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
+dateformat=%A, %B %d, %Y at %r
+
+;[template-sv_SE_pager]
+;templatefile = templates/pager_sv_se.txt
+;subject = Du har fått voicemail
+;charset=iso-8859-1
+;attachmedia=no
+;locale=sv_SE
+
+;[template-nb_NO_email]
+;templatefile = templates/email_nb_NO.txt
+;subject = Du har fått voicemail
+;charset=iso-8859-1
+;locale=nb_NO
+
+;[template-en_US_email_southern]
+;templatefile = templates/email_en_US.txt
+;subject = Y'all got voicemail, honey!
+;charset=ascii
+
+;[template-en_UK_email]
+;templatefile = templates/email_en_us.txt
+;subject = Dear old chap, you've got an electronic communique
+;charset=ascii
+
+;----------------------- Mailbox accounts --------------------------
+;Template for mailbox definition - all options
+;
+;	[username@domain]		; Has to be unique within domain (MWM_USERNAME, MWM_DOMAIN)
+;	etemplate = sv_SE		; Email template from [templates]
+;	ptemplate = en_US		; Pager template from [templates]
+;	email = userpart@domain		; Extra e-mail address (overrides mailbox name)
+;	pager = pageremail@domain	; E-mail address for pager messages
+;	fullname = Mark Spencer		; Full name  (MWM_NAME)
+;	options =			; E-mail options, se below
+;       accountcode =			; Account code (read in dialplan function MINIVMACCOUNT)
+;       pincode =			; Numeric pin code (read in dialplan function MINIVMACCOUNT)
+;	timezone=se			; Time zone
+;	serveremail = asterisk@digium.com	; Who to send email from (overrides template if set)
+;	externnotify = <application>	; External application for this account
+;	volgain =			; Volume gain setting (requires "sox")
+;	setvar=SERVICENAME=Voop.com Networks ; Extra variables to use in template
+
+; Remember that you can use Asterisk Configuration Templates (ACT)
+
+;	[template@example.com](!)		; Declare template
+;	setvar=customerdomain=example.com
+;	setvar=customerclass=gold
+;	etemplate = sv_se_email
+; 	serveremail = voicemail@example.com
+
+;	[user2@example.com](template@example.com)	; Declare user2 account using template
+;	fullname = Olle E. Johansson
+;					; User inherits everything from template

asterisk/misdn.conf

@@ -0,0 +1,537 @@
+;
+; chan_misdn sample config
+;
+
+; general section:
+;
+; for debugging and general setup, things that are not bound to port groups
+;
+
+[general]
+;
+; Sets the Path to the misdn-init.conf (for nt_ptp mode checking)
+;
+misdn_init=/etc/misdn-init.conf
+
+; set debugging flag:
+;   0 - No Debug
+;   1 - mISDN Messages and * - Messages, and * - State changes
+;   2 - Messages + Message specific Informations (e.g. bearer capability)
+;   3 - very Verbose, the above + lots of Driver specific infos
+;   4 - even more Verbose than 3
+;
+; default value: 0
+;
+debug=0
+
+
+
+; set debugging file and flags for mISDNuser (NT-Stack)
+;
+; flags can be or'ed with the following values:
+;
+; DBGM_NET        0x00000001
+; DBGM_MSG        0x00000002
+; DBGM_FSM        0x00000004
+; DBGM_TEI        0x00000010
+; DBGM_L2         0x00000020
+; DBGM_L3         0x00000040
+; DBGM_L3DATA     0x00000080
+; DBGM_BC         0x00000100
+; DBGM_TONE       0x00000200
+; DBGM_BCDATA     0x00000400
+; DBGM_MAN        0x00001000
+; DBGM_APPL       0x00002000
+; DBGM_ISDN       0x00004000
+; DBGM_SOCK       0x00010000
+; DBGM_CONN       0x00020000
+; DBGM_CDATA      0x00040000
+; DBGM_DDATA      0x00080000
+; DBGM_SOUND      0x00100000
+; DBGM_SDATA      0x00200000
+; DBGM_TOPLEVEL   0x40000000
+; DBGM_ALL        0xffffffff
+;
+
+ntdebugflags=0
+ntdebugfile=/var/log/misdn-nt.log
+
+
+; some pbx systems do cut the L1 for some milliseconds, to avoid
+; dropping running calls, we can set this flag to yes and tell
+; mISDNuser not to drop the calls on L2_RELEASE
+ntkeepcalls=no
+
+; the big trace
+;
+; default value: [not set]
+;
+;tracefile=/var/log/asterisk/misdn.log
+
+
+; set to yes if you want mISDN_dsp to bridge the calls in HW
+;
+; default value: yes
+;
+bridging=no
+
+
+; stops dialtone after getting first digit on nt Port
+;
+; default value: yes
+;
+stop_tone_after_first_digit=yes
+
+; whether to append overlapdialed Digits to Extension or not
+;
+; default value: yes
+;
+append_digits2exten=yes
+
+;;; CRYPTION STUFF
+
+; Whether to look for dynamic crypting attempt
+;
+; default value: no
+;
+dynamic_crypt=no
+
+; crypt_prefix, what is used for crypting Protocol
+;
+; default value: [not set]
+;
+crypt_prefix=**
+
+; Keys for cryption, you reference them in the dialplan
+; later also in dynamic encr.
+;
+; default value: [not set]
+;
+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.
+
+; jbforce = no                ; Forces the use of a jitterbuffer on the receive side of a SIP
+                              ; 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.
+
+; 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.
+
+; jbtargetextra = 40          ; This option only affects the jb when 'jbimpl = adaptive' is set.
+                              ; The option represents the number of milliseconds by which the new
+                              ; jitter buffer will pad its size. the default is 40, so without
+                              ; modification, the new jitter buffer will set its size to the jitter
+                              ; value plus 40 milliseconds. increasing this value may help if your
+                              ; network normally has low jitter, but occasionally has spikes.
+
+; jblog = no                  ; Enables jitterbuffer frame logging. Defaults to "no".
+;-----------------------------------------------------------------------------------
+
+; users sections:
+;
+; name your sections as you wish but not "general" or "default" !
+; the sections are Groups, you can dial out in extensions.conf
+; with Dial(mISDN/g:extern/101) where extern is a section name,
+; chan_misdn tries every port in this section to find a
+; new free channel
+;
+; The default section is not a group section, it just contains config elements
+; which are inherited by group sections.
+;
+[default]
+
+; define your default context here
+;
+; default value: default
+;
+context=misdn
+
+; language
+;
+; default value: en
+;
+language=en
+
+;
+; 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.
+;
+musicclass=default
+
+;
+; Either if we should produce DTMF Tones ourselves
+;
+senddtmf=yes
+
+;
+; If we should generate Ringing for chan_sip and others
+;
+far_alerting=no
+
+
+;
+; Here you can list which bearer capabilities should be allowed:
+;   all                  - allow any bearer capability
+;   speech               - allow speech
+;   3_1khz               - allow 3.1KHz audio
+;   digital_unrestricted - allow unrestricted digital
+;   digital_restricted   - allow restricted digital
+;   video                - allow video
+;
+; Example:
+; allowed_bearers=speech,3_1khz
+;
+allowed_bearers=all
+
+; Incoming number prefixes for the indicated Type-Of-Number.  These are
+; inserted before any number (caller, dialed, connected, redirecting,
+; redirection) received from the ISDN link if that number has the
+; corresponding Type-Of-Number.
+; See the dialplan options.
+;
+; default values:
+;    unknownprefix=
+;    internationalprefix=00
+;    nationalprefix=0
+;    netspecificprefix=
+;    subscriberprefix=
+;    abbreviatedprefix=
+;
+;unknownprefix=
+internationalprefix=00
+nationalprefix=0
+;netspecificprefix=
+;subscriberprefix=
+;abbreviatedprefix=
+
+; set rx/tx gains between -8 and 8 to change the RX/TX Gain
+;
+; default values: rxgain: 0
+;                 txgain: 0
+;
+rxgain=0
+txgain=0
+
+; some telcos especially in NL seem to need this set to yes, also in
+; switzerland this seems to be important
+;
+; default value: no
+;
+te_choose_channel=no
+
+
+
+;
+; Monitors L1 of the port.  If L1 is down it tries
+; to bring it up.  The polling timeout is given in seconds.
+; Setting the value to 0 disables monitoring L1 of the port.
+;
+; default value: 0
+;
+; This option is only read at chan_misdn loading time.
+; You need to unload and load chan_misdn to change the
+; value.  An asterisk restart will also do the trick.
+;
+l1watcher_timeout=0
+
+;
+; This option defines, if chan_misdn should check the L1 on  a PMP
+; before making a group call on it. The L1 may go down for PMP Ports
+; so we might need this.
+; But be aware! a broken or plugged off cable might be used for a group call
+; as well, since chan_misdn has no chance to distinguish if the L1 is down
+; because of a lost Link or because the Provider shut it down...
+;
+; default: no
+;
+pmp_l1_check=no
+
+
+;
+; in PMP this option defines which cause should be sent out to
+; the 3. caller. chan_misdn does not support callwaiting on TE
+; PMP side. This allows to modify the RELEASE_COMPLETE cause
+; at least.
+;
+reject_cause=16
+
+
+;
+; Send Setup_Acknowledge on incoming calls anyway (instead of PROCEEDING),
+; this requests additional Infos, so we can waitfordigits
+; without much issues. This works only for PTP Ports
+;
+; default value: no
+;
+need_more_infos=no
+
+
+;
+; set this to yes if you want to disconnect calls when a timeout occurs
+; for example during the overlapdial phase
+;
+nttimeout=no
+
+; Set the method to use for channel selection:
+;   standard     - Use the first free channel starting from the lowest number.
+;   standard_dec - Use the first free channel starting from the highest number.
+;   round_robin  - Use the round robin algorithm to select a channel. Use this
+;                  if you want to balance your load.
+;
+; default value: standard
+;
+method=standard
+
+
+; specify if chan_misdn should collect digits before going into the
+; dialplan, you can choose yes=4 Seconds, no, or specify the amount
+; of seconds you need;
+;
+overlapdial=yes
+
+;
+; dialplan means Type Of Number in ISDN Terms
+; There are different types of the dialplan:
+;
+; dialplan -> for outgoing call's dialed number
+; localdialplan -> for outgoing call's callerid
+;       (if -1 is set use the value from the asterisk channel)
+; cpndialplan -> for incoming call's connected party number sent to caller
+;       (if -1 is set use the value from the asterisk channel)
+;
+; dialplan options:
+;
+; 0 - unknown
+; 1 - International
+; 2 - National
+; 3 - Network-Specific
+; 4 - Subscriber
+; 5 - Abbreviated
+;
+; default value: 0
+;
+dialplan=0
+localdialplan=0
+cpndialplan=0
+
+
+
+;
+; turn this to no if you don't mind correct handling of Progress Indicators
+;
+early_bconnect=yes
+
+
+;
+; turn this on if you like to send Tone Indications to a Incoming
+; isdn channel on a TE Port. Rarely used, only if the Telco allows
+; you to send indications by yourself, normally the Telco sends the
+; indications to the remote party.
+;
+; default: no
+;
+incoming_early_audio=no
+
+; uncomment the following to get into s extension at extension conf
+; there you can use DigitTimeout if you can't or don't want to use
+; isdn overlap dial.
+; note: This will jump into the s exten for every exten!
+;
+; default value: no
+;
+;always_immediate=no
+
+;
+; set this to yes if you want to generate your own dialtone
+; with always_immediate=yes, else chan_misdn generates the dialtone
+;
+; default value: no
+;
+nodialtone=no
+
+
+; uncomment the following if you want callers which called exactly the
+; base number (so no extension is set) jump to the s extension.
+; if the user dials something more it jumps to the correct extension
+; instead
+;
+; default value: no
+;
+;immediate=no
+
+; uncomment the following to have hold and retrieve support
+;
+; default value: no
+;
+;hold_allowed=yes
+
+; Pickup and Callgroup
+;
+; default values: not set = 0
+; range: 0-63
+;
+;callgroup=1
+;pickupgroup=1
+
+; Named pickup groups and named call groups
+;
+; give a name to groups and configure any number of groups
+;
+;namedcallgroup=engineering,sales,netgroup,protgroup
+;namedpickupgroup=sales
+
+; Set the outgoing caller id to the value.
+;callerid="name" <number>
+
+;
+; these are the exact isdn screening and presentation indicators
+; if -1 is given for either value the presentation indicators are used
+; from asterisks CALLERPRES function.
+; s=0, p=0 -> callerid presented
+; s=1, p=1 -> callerid restricted (the remote end does not see it!)
+;
+; default values s=-1, p=-1
+presentation=-1
+screen=-1
+
+; Incoming calls will have a caller ID tag set to this value
+;
+;incoming_cid_tag = "asterisk"
+
+; With this set, you can automatically append the MSN of a party
+; to the cid_tag. Incoming calls have the dialed number appended
+; to the tag, and outgoing calls have the caller number appended
+; to the tag. An '_' is used to separate the tag from the
+; MSN.
+; Default is no.
+;
+;append_msn_to_cid_tag = no
+
+; Select what to do with outgoing COLP information on this port.
+;
+; 0 - Send out COLP information unaltered. (default)
+; 1 - Force COLP to restricted on all outgoing COLP information.
+; 2 - Do not send COLP information.
+outgoing_colp=0
+
+; Put a display ie in the CONNECT message containing the following
+; information if it is available (nt port only):
+;
+; 0 - Do not put the connected line information in the display ie.
+; 1 - Put the available connected line name in the display ie.
+; 2 - Put the available connected line number in the display ie.
+; 3 - Put the available connected line name and number in the display ie.
+;
+display_connected=0
+
+; Put a display ie in the SETUP message containing the following
+; information if it is available (nt port only):
+;
+; 0 - Do not put the caller information in the display ie.
+; 1 - Put the available caller name in the display ie.
+; 2 - Put the available caller number in the display ie.
+; 3 - Put the available caller name and number in the display ie.
+;
+display_setup=0
+
+; This enables echo cancellation with the given number of taps.
+; Be aware: Move this setting only to outgoing portgroups!
+; A value of zero turns echo cancellation off.
+;
+; possible values are: 0,32,64,128,256,yes(=128),no(=0)
+;
+; default value: no
+;
+;echocancel=no
+
+;
+; chan_misdns jitterbuffer, default 4000
+;
+jitterbuffer=4000
+
+;
+; change this threshold to enable dejitter functionality
+;
+jitterbuffer_upper_threshold=0
+
+
+;
+; change this to yes, if you want to bridge a mISDN data channel to
+; another channel type or to an application.
+;
+hdlc=no
+
+
+;
+; defines the maximum amount of incoming calls per port for
+; this group. Calls which exceed the maximum will be marked with
+; the channel variable MAX_OVERFLOW. It will contain the amount of
+; overflowed calls
+;
+max_incoming=-1
+
+;
+; defines the maximum amount of outgoing calls per port for this group
+; exceeding calls will be rejected
+;
+max_outgoing=-1
+
+;
+; Enable/disable the call-completion retention option support (ptp only).
+;
+; Note: To use the CCBS/CCNR supplementary service feature and other
+; supplementary services using FACILITY messages requires a
+; modified version of mISDN from:
+; http://svn.digium.com/svn/thirdparty/mISDN/trunk
+; http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
+;
+cc_request_retention=yes
+
+[intern]
+; define your ports, e.g. 1,2 (depends on mISDN-driver loading order)
+ports=1,2
+; context where to go to when incoming Call on one of the above ports
+context=Intern
+
+[internPP]
+;
+; adding the postfix 'ptp' to a port number is obsolete now, chan_misdn
+; parses /etc/misdn-init.conf and sets the ptp mode to the corresponding
+; configs. For backwards compatibility you can still set ptp here.
+;
+ports=3
+
+[first_extern]
+; again port defs
+ports=4
+; again a context for incoming calls
+context=Extern1
+; msns for te ports, listen on those numbers on the above ports, and
+; indicate the incoming calls to asterisk
+; here you can give a comma separated list or simply an '*' for
+; any msn.
+msns=*
+
+; here an example with given msns
+[second_extern]
+ports=5
+context=Extern2
+callerid="Asterisk" <1234>
+msns=102,144,101,104

asterisk/modules.conf

@@ -0,0 +1,45 @@
+;
+; Asterisk configuration file
+;
+; Module Loader configuration file
+;
+
+[modules]
+autoload=yes
+;
+; Any modules that need to be loaded before the Asterisk core has been
+; initialized (just after the logger has been initialized) can be loaded
+; using 'preload'. This will frequently be needed if you wish to map all
+; module configuration files into Realtime storage, since the Realtime
+; driver will need to be loaded before the modules using those configuration
+; files are initialized.
+;
+; An example of loading ODBC support would be:
+;preload => res_odbc.so
+;preload => res_config_odbc.so
+;
+; Uncomment the following if you wish to use the Speech Recognition API
+;preload => res_speech.so
+;
+; If you want Asterisk to fail if a module does not load, then use
+; the "require" keyword. Asterisk will exit with a status code of 2
+; if a required module does not load.
+;
+; require = chan_sip.so
+; If you want you can combine with preload
+; preload-require = res_odbc.so
+;
+; If you want, load the GTK console right away.
+;
+noload => pbx_gtkconsole.so
+;load => pbx_gtkconsole.so
+;
+load => res_musiconhold.so
+;
+; Load one of: chan_oss, alsa, or console (portaudio).
+; By default, load chan_oss only (automatically).
+;
+noload => chan_alsa.so
+;noload => chan_oss.so
+noload => chan_console.so
+;

asterisk/motif.conf

@@ -0,0 +1,92 @@
+; Sample configuration file for chan_motif
+
+; Transports
+;
+; There are three different transports and protocol derivatives supported by chan_motif. They are in order of preference:
+; Jingle using ICE-UDP, Google Jingle, and Google-V1.
+;
+; Jingle as defined in XEP-0166 supports the widest range of features. It is referred to as "ice-udp" in this file. This is
+; the specification that Jingle clients implement.
+;
+; Google Jingle follows the Jingle specification for signaling but uses a custom transport for media. It is supported
+; by the Google Talk Plug-in in Gmail and by some other Jingle clients. It is referred to as "google" in this file.
+;
+; Google-V1 is the original Google Talk signaling protocol which uses an initial preliminary version of Jingle.
+; It also uses the same custom transport as Google Jingle for media. It is supported by Google Voice, some other Jingle
+; clients, and the Windows Google Talk client. It is referred to as "google-v1" in this file.
+;
+; Incoming sessions will automatically switch to the correct transport once it has been determined.
+;
+; Outgoing sessions are capable of determining if the target is capable of Jingle or a Google transport if the target is
+; in the roster. Unfortunately it is not possible to differentiate between a Google Jingle or Google-V1 capable resource
+; until a session initiate attempt occurs. If a resource is determined to use a Google transport it will initially use
+; Google Jingle but will fall back to Google-V1 if required.
+;
+; If an outgoing session attempt fails due to failure to support the given transport chan_motif will fall back in preference
+; order listed at the beginning of this document until all transports have been exhausted.
+;
+
+; Dialing and Resource Selection Strategy
+;
+; Placing a call through an endpoint can be accomplished using the following dial string:
+;
+; Motif/<endpoint name>/<target>
+;
+; When placing an outgoing call through an endpoint the requested target is searched for in the roster list. If present
+; the first Jingle or Google Jingle capable resource is specifically targetted. Since the capabilities of the resource are
+; known the outgoing session initation will disregard the configured transport and use the determined one.
+;
+; If the target is not found in the roster the target will be used as-is and a session will be initiated using the
+; transport specified in this configuration file. If no transport has been specified the endpoint defaults to ice-udp.
+;
+
+; Video Support
+;
+; Support for video does not need to be explicitly enabled. Configuring any video codec on your endpoint will
+; automatically enable it.
+
+; DTMF
+;
+; The only supported method for DTMF is RFC2833. This is always enabled on audio streams and negotiated if possible.
+
+; Incoming Calls
+;
+; Incoming calls will first look for the extension matching the name of the endpoint in the configured context. If
+; no such extension exists the call will automatically fall back to the "s" extension.
+
+; CallerID
+;
+; The incoming caller id number is populated with the username of the caller and the name is populated with the full
+; identity of the caller. If you would like to perform authentication or filtering of incoming calls it is recommended
+; that you use these fields to do so.
+;
+; Outgoing caller id can *not* be set.
+;
+; IMPORTANT: Multiple endpoints using the same connection is *NOT* supported. Doing so may result in broken calls.
+
+; Default template for endpoints, to be included in their definition
+[default](!)
+disallow=all
+allow=ulaw
+allow=h264
+context=incoming-motif ; Default context that incoming sessions will land in
+
+;maxicecandidates = 10 ; Maximum number of ICE candidates we will offer
+;maxpayloads = 30      ; Maximum number of payloads we will offer
+
+; Sample configuration entry for Jingle
+[jingle-endpoint](default)
+transport=ice-udp               ; Change the default protocol of outgoing sessions to Jingle ICE-UDP
+allow=g722                      ; Add G.722 as an allowed format since the other side may support it
+connection=local-jabber-account ; Connection to accept traffic on and send traffic out
+accountcode=jingle              ; Account code for CDR purposes
+
+; Sample configuration entry for Google Talk
+[gtalk-endpoint](default)
+transport=google         ; Since this is a Google Talk endpoint we want to offer Google Jingle for outgoing sessions
+connection=gtalk-account
+
+; Sample configuration entry for Google Voice
+[gvoice](default)
+transport=google-v1       ; Google Voice uses the original Google Talk protocol
+connection=gvoice-account

asterisk/musiconhold.conf

@@ -0,0 +1,89 @@
+;
+; Music on Hold -- Sample Configuration
+;
+[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
+
+
+; valid mode options:
+; files		-- read files from a directory in any Asterisk supported
+;		   media format
+; quietmp3 	-- default
+; mp3 		-- loud
+; mp3nb		-- unbuffered
+; quietmp3nb	-- quiet unbuffered
+; custom	-- run a custom application (See examples below)
+
+; =========
+; File-based (native) music on hold
+; =========
+;
+; This plays files directly from the specified directory, no external
+; processes are required. Files are played in normal sorting order
+; (same as a sorted directory listing), and no volume or other
+; sound adjustments are available. If the file is available in
+; the same format as the channel's codec, then it will be played
+; without transcoding (same as Playback would do in the dialplan).
+; Files can be present in as many formats as you wish, and the
+; 'best' format will be chosen at playback time.
+;
+; The path specified can be either an absolute path (starts with '/'),
+; or a relative path; relative paths are interpreted as being relative
+; to the 'astdatalibdir' in asterisk.conf, which defaults to
+; /var/lib/asterisk.
+;
+; NOTE:
+; If you are not using "autoload" in modules.conf, then you
+; must ensure that the format modules for any formats you wish
+; to use are loaded _before_ res_musiconhold. If you do not do
+; this, res_musiconhold will skip the files it is not able to
+; understand when it loads.
+;
+
+[default]
+mode=files
+directory=moh
+;
+;[native-random]
+;mode=files
+;directory=moh
+;digit=#        ; If this option is set for a class, then when callers are
+;               ; listening to music on hold, they can press this digit, and
+;               ; they will switch to listening to this music class.
+;announcement=queue-thankyou	;If this option is set for a class, then
+;               ; when callers get put on hold, the specified sound will be
+;               ; be played to them. Also, if using modes that Asterisk
+;               ; controls the playlist for (files, mp3, etc), the same
+;               ; sound will also be played between MOH songs.
+;sort=random    ; Sort the files in random order
+
+;[native-alphabetical]
+;mode=files
+;directory=moh
+;sort=alpha     ; Sort the files in alphabetical order.  If this option is
+;               ; not specified, the sort order is undefined.
+
+; =========
+; Other (non-native) playback methods
+; =========
+
+;[manual]
+;mode=custom
+; Note that with mode=custom, a directory is not required, such as when reading
+; from a stream.
+;directory=/var/lib/asterisk/mohmp3
+;application=/usr/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s
+
+;[ulawstream]
+;mode=custom
+;application=/usr/bin/streamplayer 192.168.100.52 888
+;format=ulaw
+
+; mpg123 on Solaris does not always exit properly; madplay may be a better
+; choice
+;[solaris]
+;mode=custom
+;directory=/var/lib/asterisk/mohmp3
+;application=/site/sw/bin/madplay -Q -o raw:- --mono -R 8000 -a -12

asterisk/muted.conf

@@ -0,0 +1,39 @@
+#
+# Sample muted configuration file
+#
+# Copyright (C) 2004 Digium, Inc.
+#
+# First you have the host, username, and password
+# we use to connect to the asterisk system
+#
+# What is this? Well, haven't you ever wished you could automatically
+# turn down the volume on your stereo, CDPlayer, etc, when a call comes in,
+# and then return it to normal when the call ends? Well, this is a possible
+# mechanism to make this happen!
+# You have to fire up the new utils/muted, which runs as a daemon in the
+# background. This daemon connects to asterisk via a manager interface, and
+# also reads this config file from /etc/muted.conf. when the channels mentioned
+# are activated, it tweaks the sound levels on the sound card(s).
+# So, depending on the sound card, you may be able to run all your sound
+# generating devices thru your sound card, and use this mechanism to quiet
+# them down while you are on the phone. If anyone figures out how to make
+# this work with kids, please inform!!
+#
+host localhost
+user user
+pass pass
+#
+# List each channel we're supposed to watch
+#
+channel DAHDI/1
+channel DAHDI/2
+channel SIP/mark
+#
+# Mute level is the percentage of the current volume we should
+# lower the music to.
+#
+mutelevel 20
+#
+# Smooth fade makes the fadein/fadeout nicer sounding
+#
+smoothfade

asterisk/ooh323.conf

@@ -0,0 +1,204 @@
+; ---------------------------------------------------------------------------------
+; ---  ******* IMPORTANT NOTE ***********
+; ---
+; ---  This module is currently unsupported.  Use it at your own risk.
+; ---
+; ---------------------------------------------------------------------------------
+
+; Objective System's H323 Configuration example for Asterisk
+; ooh323c driver configuration
+;
+; [general] section defines global parameters
+;
+; This is followed by profiles which can be of three types - user/peer/friend
+; Name of the user profile should match with the h323id of the user device.
+; For peer/friend profiles, host ip address must be provided as "dynamic" is
+; not supported as of now.
+;
+; Syntax for specifying a H323 device in extensions.conf is
+; For Registered peers/friends profiles:
+;        OOH323/name where name is the name of the peer/friend profile.
+;
+; For unregistered H.323 phones:
+;        OOH323/ip[:port] OR if gk is used OOH323/alias where alias can be any H323
+;                          alias
+;
+; For dialing into another asterisk peer at a specific exten
+;       OOH323/exten/peer OR OOH323/exten@ip
+;
+; Domain name resolution is not yet supported.
+;
+; When a H.323 user calls into asterisk, his H323ID is matched with the profile
+; name and context is determined to route the call
+;
+; The channel driver will register all global aliases and aliases defined in
+; peer profiles with the gatekeeper, if one exists. So, that when someone
+; outside our pbx (non-user) calls an extension, gatekeeper will route that
+; call to our asterisk box, from where it will be routed as per dial plan.
+
+
+[general]
+;Define the asetrisk server h323 endpoint
+
+;The port asterisk should listen for incoming H323 connections.
+;Default - 1720
+;port=1720
+
+;The dotted IP address asterisk should listen on for incoming H323
+;connections
+;Default - tries to find out local ip address on it's own
+bindaddr=0.0.0.0
+
+;This parameter indicates whether channel driver should register with
+;gatekeeper as a gateway or an endpoint.
+;Default - no
+;gateway=no
+
+;Whether asterisk should use fast-start and tunneling for H323 connections.
+;Default - yes
+;faststart=no
+;h245tunneling=no
+
+;Whether media wait for connect
+;Default - No
+;mediawaitforconnect=yes
+
+;H323-ID to be used for asterisk server
+;Default - Asterisk PBX
+h323id=ObjSysAsterisk
+e164=100
+
+;CallerID to use for calls
+;Default - Same as h323id
+callerid=asterisk
+
+;Whether this asterisk server will use gatekeeper.
+;Default - DISABLE
+;gatekeeper = DISCOVER
+;gatekeeper = a.b.c.d
+gatekeeper = DISABLE
+
+;Location for H323 log file
+;Default - /var/log/asterisk/h323_log
+;logfile=/var/log/asterisk/h323_log
+
+
+;Following values apply to all users/peers/friends defined below, unless
+;overridden within their client definition
+
+;Sets default context all clients will be placed in.
+;Default - default
+context=default
+
+;Sets rtptimeout for all clients, unless overridden
+;Default - 60 seconds
+;rtptimeout=60	    ; Terminate call if 60 seconds of no RTP activity
+                    ; when we're not on hold
+
+;Type of Service
+;Default - none (lowdelay, thoughput, reliability, mincost, none)
+;tos=lowdelay
+
+;amaflags = default
+
+;The account code used by default for all clients.
+;accountcode=h3230101
+
+;The codecs to be used for all clients.Only ulaw and gsm supported as of now.
+;Default - ulaw
+; ONLY ulaw, gsm, g729 and g7231 supported as of now
+disallow=all     ;Note order of disallow/allow is important.
+allow=gsm
+allow=ulaw
+
+
+; dtmf mode to be used by default for all clients. Supports rfc2833, q931keypad
+; h245alphanumeric, h245signal.
+;Default - rfc 2833
+dtmfmode=rfc2833
+;
+; round trip delay request, default = 0,0 (not send)
+; x - count of unreplied requests before hangup, y - interval in sec between requests
+;
+;roundtrip=x,y
+
+;
+; FAX detection will cause the OOH323 channel to jump to the 'fax' extension (if it exists)
+; based one or more events being detected. The events that can be detected are an incoming
+; CNG tone or an incoming T.38 RequestMode packet
+;
+; yes - enable both detection (CNG & T.38)
+; no - disable both
+; cng - enable CNG detection (default)
+; t38 - enable T.38 request detection
+;
+faxdetect = cng
+
+; User/peer/friend definitions:
+; User config options                    Peer config options
+; ------------------                     -------------------
+; context
+; disallow                               disallow
+; allow                                  allow
+; accountcode                            accountcode
+; amaflags                               amaflags
+; dtmfmode                               dtmfmode
+; rtptimeout                             ip
+;                                        port
+;                                        h323id
+;                                        email
+;                                        url
+;                                        e164
+;                                        rtptimeout
+
+;
+
+;
+;	direct rtp between two remote peers, disabled by default
+;	can be specified globally or per peer/user section
+;
+directmedia=no
+;
+;	early direct rtp (try to establish direct rtp before asnwer)
+;	disabled by default, auto enabled by directmedia is enabled
+;	can be disabled escpecially if directmedia is enabled.
+;	can be specified globally or per peer/user section
+;
+;
+directrtpsetup=no
+
+;Define users here
+;Section header is extension
+[myuser1]
+type=user
+context=context1
+disallow=all
+allow=gsm
+allow=ulaw
+
+
+
+[mypeer1]
+type=peer
+context=context2
+ip=a.b.c.d   ; UPDATE with appropriate ip address
+port=1720    ; UPDATE with appropriate port
+e164=101
+
+
+
+[myfriend1]
+type=friend
+context=default
+ip=10.0.0.82   ; UPDATE with appropriate ip address
+port=1820    ; UPDATE with appropriate port
+disallow=all
+allow=ulaw
+e164=12345
+rtptimeout=60
+dtmfmode=rfc2833
+
+
+
+
+

asterisk/osp.conf

@@ -0,0 +1,117 @@
+;
+; 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.
+;
+; 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.
+; The default value is no.
+;
+;accelerate=no
+;
+; Enable security features.
+; If security features are disabled, Asterisk cannot validate signed tokens and
+; all certificate file name parameters are ignored.
+; The default value is no.
+;
+;securityfeatures=no
+;
+; 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.
+;
+;tokenformat=0
+;
+;[default]
+;
+; List all service points (OSP servers) for this provider.
+; Use either domain name or IP address.  Most OSP servers use port 5045.
+;
+;servicepoint=http://osptestserver.transnexus.com:5045/osp
+;
+; Define the "source" device for requesting OSP authorization.
+; This value is usually the domain name or IP address of the the Asterisk server.
+;
+;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
+; 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:
+; 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)
+; 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:
+; 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.
+;
+; 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)
+; timeout:        Timeout for response in milliseconds (default=500)
+;
+;maxconnections=20
+;retrydelay=0
+;retrylimit=2
+;timeout=500
+;
+; 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
+;                 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
+; IAX protocols.  The default protocol is set to SIP.
+;
+;defaultprotocol=SIP
+;
+; Set the work mode.
+; 0 - Direct
+; 1 - Indirect
+; Default is 0,
+;
+;workmode=0
+;
+; Set the service type.
+; 0 - Normal voice service
+; 1 - Ported number query service
+; Default is 0,
+;
+;servicetype=0

asterisk/oss.conf

@@ -0,0 +1,153 @@
+;
+; Automatically generated from ../channels/chan_oss.c
+;
+
+[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.
+
+    ; jbtargetextra = 40          ; This option only affects the jb when 'jbimpl = adaptive' is set.
+                                  ; The option represents the number of milliseconds by which the new
+                                  ; jitter buffer will pad its size. the default is 40, so without
+                                  ; modification, the new jitter buffer will set its size to the jitter
+                                  ; value plus 40 milliseconds. increasing this value may help if your
+                                  ; network normally has low jitter, but occasionally has spikes.
+
+    ; jblog = no                  ; Enables jitterbuffer frame logging. Defaults to "no".
+    ;-----------------------------------------------------------------------------------
+
+; below is an entry for a second console channel
+; [card1]
+    ; device = /dev/dsp1	; alternate device
+
+; Below are the settings to support video. You can include them
+; in your general configuration as [general](+,video)
+; The parameters are all available through the CLI as "console name value"
+; 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.
+
+; 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.
+; These entries can also be contained in the comment field of the image,
+; which is a lot more convenient to manage.
+; E.g. for jpeg you can write them with wrjpgcom (part of libjpeg).
+; The format to define keys is
+;	region = <event> <shape> x0 y0 x1 y1 h
+; where <event> is the event to be generated (a digit, pickup, hangup,...)
+; <shape> is the shape of the region (currently 'rect' and 'circle' are
+; supported, the latter is really an ellipse),  x0 y0 x1 y1 are the
+; coordinates of the base of the rectangle or main diameter of the ellipse,
+; (they can be rotated) while h is the height of the rectangle or the other
+; 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
+
+; 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
+
+; to add video support, uncomment this and remember to install
+; the keypad and keypad_font files to the right place
+; [general](+,my_video,skin2)
+

asterisk/phone.conf

@@ -0,0 +1,51 @@
+;
+; Linux Telephony Interface
+;
+; Configuration file
+;
+[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
+; 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
+; "sig".
+;
+mode=immediate
+;mode=dialtone
+;mode=fxo
+;mode=sig
+;
+; You can decide which format to use by default, "g723.1", "g729", or "slinear".
+; Note that g729 is only supported for Sigma Designs boards.
+; XXX Be careful, sometimes the card causes kernel panics when running
+; in signed linear mode for some reason... XXX
+;
+format=slinear
+;format=g723.1
+;format=g729
+;
+; And set the echo cancellation to "off", "low", "medium", and "high".
+; This is not supported on all phones.
+;
+echocancel=medium
+;
+; You can optionally use VAD/CNG silence suppression
+;
+;silencesupression=yes
+;
+; List all devices we can use.  Contexts may also be specified
+;
+;context=local
+;
+; You can set txgain and rxgain for each device in the same way as context.
+; If you want to change default gain value (1.0 =~ 100%) for device, simple
+; add txgain or rxgain line before device line. But remember, if you change
+; volume all cards listed below will be affected by these values. You can
+; use float values (1.0, 0.5, 2.0) or percentage values (100%, 150%, 50%).
+;
+;txgain=100%
+;rxgain=1.0
+;device => /dev/phone0

asterisk/phoneprov.conf

@@ -0,0 +1,137 @@
+[general]
+; The default behavior of res_phoneprov will be to set the SERVER template variable to
+; the IP address that the phone uses to contact the provisioning server and the
+; SERVER_PORT variable to the bindport setting in sip.conf.  Unless you have a very
+; unusual setup, you should not need to set serveraddr, serveriface, or serverport.
+
+;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.
+;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
+; templates to have access to.  Example:
+
+;[example]
+;mime_type => application/octet-stream
+;static_file => example/firmware
+;static_file => example/default.cfg,text/xml
+;${TOUPPER(${MAC})}.cfg => templates/example-mac.cfg
+;setvar => DB_CIDNAME=${ODBC_CID_NAME_LOOKUP(${USERNAME})}
+
+; Dynamically generated files have a filename registered with variable substitution
+; with variables obtained while reading users.conf.
+
+; Built in variables and the options in users.conf that they come from
+;   MAC (macaddress)
+;   USERNAME (username)
+;   DISPLAY_NAME (fullname)
+;   SECRET (secret)
+;   LABEL (label)
+;   CALLERID (cid_number)
+;   VOCIEMAIL_EXTEN (vmexten)
+;   EXTENSION_LENGTH (localextenlength)
+;   LINE
+;   LINEKEYS
+
+; Built-in variables and the options in phoneprov.conf that they come from
+;   SERVER (server)
+;   SERVER_PORT (serverport)
+
+
+; Built-in variables for managing timezones and daylight savings time.
+;   TZOFFSET
+;   DST_ENABLE
+;   DST_START_MONTH
+;   DST_START_MDAY
+;   DST_START_HOUR
+;   DST_END_MONTH
+;   DST_END_MDAY
+;   DST_END_HOUR
+;   TIMEZONE
+
+[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
+mime_type => text/xml ; Default mime type to use if one isn't specified or the
+                      ; 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 => 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 => 2345-12200-004.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 => 2345-12365-001.bootrom.ld,application/octet-stream
+static_file => 2345-12365-001.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 => 2345-12600-001.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
+static_file => 3111-15600-001.bootrom.ld,application/octet-stream
+static_file => 3111-15600-001.sip.ld,application/octet-stream
+static_file => 3111-40000-001.bootrom.ld,application/octet-stream
+static_file => 3111-40000-001.sip.ld,application/octet-stream
+static_file => SoundPointIPWelcome.wav,application/octet-stream
+static_file => SoundPointIPLocalization/Japanese_Japan/SoundPointIP-dictionary.xml
+static_file => SoundPointIPLocalization/Norwegian_Norway/SoundPointIP-dictionary.xml
+static_file => SoundPointIPLocalization/Spanish_Spain/SoundPointIP-dictionary.xml
+static_file => SoundPointIPLocalization/Portuguese_Portugal/SoundPointIP-dictionary.xml
+static_file => SoundPointIPLocalization/English_United_Kingdom/SoundPointIP-dictionary.xml
+static_file => SoundPointIPLocalization/English_United_States/SoundPointIP-dictionary.xml
+static_file => SoundPointIPLocalization/Russian_Russia/SoundPointIP-dictionary.xml
+static_file => SoundPointIPLocalization/Italian_Italy/SoundPointIP-dictionary.xml
+static_file => SoundPointIPLocalization/Chinese_China/SoundPointIP-dictionary.xml
+static_file => SoundPointIPLocalization/Swedish_Sweden/SoundPointIP-dictionary.xml
+static_file => SoundPointIPLocalization/English_Canada/SoundPointIP-dictionary.xml
+static_file => SoundPointIPLocalization/German_Germany/SoundPointIP-dictionary.xml
+static_file => SoundPointIPLocalization/French_France/SoundPointIP-dictionary.xml
+static_file => SoundPointIPLocalization/Danish_Denmark/SoundPointIP-dictionary.xml
+static_file => SoundPointIPLocalization/Dutch_Netherlands/SoundPointIP-dictionary.xml
+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/${MAC} => polycom.xml                 ; Dynamic Filename => template file
+${MAC}-directory.xml => 000000000000-directory.xml
+setvar => CUSTOM_CONFIG=/var/lib/asterisk/phoneprov/configs/custom.cfg  ; Custom variable
+
+;snom 300, 320, 360, 370, 820, 821, 870 support
+snom-${MAC}.xml => snom-mac.xml
+

asterisk/queuerules.conf

@@ -0,0 +1,20 @@
+; 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.
+;
+; 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 caller's stay in the queue, these will not be reflected for that caller.
+;
+; The syntax for these rules is
+; penaltychange => <number of seconds into the call>,<absolute or relative change to QUEUE_MAX_PENALTY>[,absolute or relative change to QUEUE_MIN_PENALTY]
+;
+; Example:
+; [myrule]
+; penaltychange => 30,+3   ; 30 seconds into the call increase the QUEUE_MAX_PENALTY by 3, no change to QUEUE_MIN_PENALTY
+; penaltychange => 60,10,5 ; 60 seconds into the call increase the QUEUE_MAX_PENALTY to 10 and increase the QUEUE_MIN_PENALTY to 5
+; penaltychange => 75,,7   ; 75 seconds into the call keep the QUEUE_MAX_PENALTY the same and increase the QUEUE_MIN_PENALTY to 7
+

asterisk/queues.conf

@@ -0,0 +1,580 @@
+[general]
+;
+; Global settings for call queues
+;
+; Persistent Members
+;    Store each dynamic member in each queue in the astdb so that
+;    when asterisk is restarted, each member will be automatically
+;    read into their recorded queues. Default is 'no'.
+;
+persistentmembers = yes
+;
+; AutoFill Behavior
+;    The old behavior of the queue (autofill=no) is to have 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
+;    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
+;    waits while this happens. The new behavior, enabled by setting
+;    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
+;    do not specify or comment out this option, it will default to yes.
+;
+;autofill = no
+;
+; 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
+;    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
+;    configuration block. If you do not specify or comment out this option,
+;    it will default to the old 'Monitor' behavior to keep backward
+;    compatibility.
+;
+monitor-type = MixMonitor
+;
+; 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.
+;
+; updatecdr = no
+
+;
+; Note that a timeout to fail out of a queue may be passed as part of
+; an application call from extensions.conf:
+; Queue(queuename,[options],[optionalurl],[announceoverride],[timeout])
+; example: Queue(dave,t,,,45)
+
+; shared_lastcall will make the lastcall and calls received be the same in
+; members logged in more than one queue.  This is useful to make the queue
+; respect the wrapuptime of another queue for a shared member.
+; The default value is no.
+;
+;shared_lastcall=no
+;
+; Negative_penalty_invalid will treat members with a negative penalty as logged off
+;
+;negative_penalty_invalid = no
+;
+; log_membername_as_agent will cause app_queue to log the membername rather than
+; the interface for the ADDMEMBER and REMOVEMEMBER events when a state_interface
+; is set.  The default value (no) maintains backward compatibility.
+;
+;log_membername_as_agent = no
+;
+;[markq]
+;
+; A sample call queue
+;
+; Musicclass sets which music applies for this particular call queue.
+; The only class which can override this one is if the MOH class is set
+; directly on the channel using Set(CHANNEL(musicclass)=whatever) in the
+; dialplan.
+;
+;musicclass = default
+;
+; An announcement may be specified which is played for the member as
+; soon as they answer a call, typically to indicate to them which queue
+; this call should be answered as, so that agents or members who are
+; listening to more than one queue can differentiated how they should
+; engage the customer
+;
+;announce = queue-markq
+;
+; A strategy may be specified.  Valid strategies include:
+;
+; ringall - ring all available channels until one answers (default)
+; leastrecent - ring interface which was least recently hung up by this queue
+; fewestcalls - ring the one with fewest completed calls from this queue
+; random - ring random interface
+; rrmemory - round robin with memory, remember where we left off last ring pass
+; rrordered - same as rrmemory, except the queue member order from config file 
+;             is preserved
+; linear - rings interfaces in the order specified in this configuration file.
+;          If you use dynamic members, the members will be rung in the order in
+;          which they were added
+; wrandom - rings random interface, but uses the member's penalty as a weight
+;           when calculating their metric. So a member with penalty 0 will have
+;           a metric somewhere between 0 and 1000, and a member with penalty 1 will
+;			have a metric between 0 and 2000, and a member with penalty 2 will have
+;           a metric between 0 and 3000. Please note, if using this strategy, the member
+;           penalty is not the same as when using other queue strategies. It is ONLY used
+;           as a weight for calculating metric.
+;
+;strategy = ringall
+;
+; Second settings for service level (default 0)
+; Used for service level statistics (calls answered within service level time
+; frame)
+;servicelevel = 60
+;
+; A context may be specified, in which if the user types a SINGLE
+; digit extension while they are in the queue, they will be taken out
+; of the queue and sent to that extension in this context.
+;
+;context = qoutcon
+;
+; A limit can be set to disregard penalty settings when the queue has
+; too few members.  No penalty will be weighed in if there are only X
+; or fewer queue members. (default 0)
+;
+;penaltymemberslimit = 5
+;
+;----------------------QUEUE TIMING OPTIONS------------------------------------
+; A Queue has two different "timeout" values associated with it. One is the
+; timeout parameter configured in queues.conf. This timeout specifies the
+; amount of time to try ringing a member's phone before considering the
+; member to be unavailable. The other timeout value is the second argument
+; to the Queue() application. This timeout represents the absolute amount
+; of time to allow a caller to stay in the queue before the caller is
+; removed from the queue. In certain situations, these two timeout values
+; may clash. For instance, if the timeout in queues.conf is set to 5 seconds,
+; the retry value in queues.conf is set to 4, and the second argument to Queue()
+; is 10, then the following may occur:
+;
+; A caller places a call to a queue.
+; The queue selects a member and attempts to ring that member.
+; 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
+; being in the queue for 10 seconds? This is configurable with the timeoutpriority
+; option. By setting the timeoutpriority to "conf" then you are saying that you would
+; 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
+; Queue application is more important. In the scenario above, timeoutpriority=app
+; would result in the second member's phone ringing for 1 second.
+;
+; There are a few exceptions to the priority rules. For instance, if timeoutpriority=appp
+; 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
+; when calling queue members.
+;
+; In timeoutpriority=conf mode however timeout specified in config file will take higher
+; priority than timeout in application arguments, so if config file has timeout 0, each
+; queue member will be called indefineately and application timeout will be checked only
+; after this call attempt. This is useful for having queue members with custom timeouts
+; specified within Dial application of Local channel, and allows handling NO ANSWER which
+; would otherwise be interrupted by queue destroying child channel on timeout.
+;
+; The default value for timeoutpriority is "app" since this was how previous versions of
+; Asterisk behaved.
+;
+;timeout = 15
+;retry = 5
+;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
+; more than one queue.
+;
+;weight=0
+;
+; After a successful call, how long to wait before sending a potentially
+; free member another call (default is 0, or no delay)
+;
+;wrapuptime=15
+;
+; Autofill will follow queue strategy but push multiple calls through
+; at same time until there are no more waiting callers or no more
+; available members. The per-queue setting of autofill allows you
+; to override the default setting on an individual queue level.
+;
+;autofill=yes
+;
+; Autopause will pause a queue member if they fail to answer a call
+; no: Member will not be paused
+; yes: Member will be paused only in the queue where the timeout took place
+; all: Memeber will be paused in all queues he/she is a member
+;autopause=yes
+;
+; Autopausedelay delay autopause for autopausedelay seconds from the
+; last call if a member has not taken a call the delay has no effect.
+;autopausedelay=60
+;
+; Autopausebusy controls whether or not a queue member is set as paused
+; automatically upon the member device reporting busy. The autopausedelay
+; option applies. Defaults to 'no'.
+;autopausebusy=no
+;
+; Autopauseunavail controls whether or not a queue member is set as paused
+; automatically upon the member device reporting congestion. The autopausedely
+; option applies. Defaults to 'no'.
+;autopauseunavail=no
+;
+; Maximum number of people waiting in the queue (0 for unlimited)
+;
+;maxlen = 0
+;
+; Note: for below queue channel options (setinterfacevar, setqueueentryvar,
+; setqueuevar) if the caller channel is a local channel and optimizations
+; is enabled then after optimization has occurred only the queue member
+; channel will contain the variables.
+;
+; If set to yes, just prior to the caller being bridged with a queue member
+; the following variables will be set on the caller and queue member channels:
+; 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
+; 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
+; the following variables will be set on the caller and queue member channels:
+; QEHOLDTIME callers hold time
+; QEORIGINALPOS original position of the caller in the queue
+;
+;setqueueentryvar=no
+;
+; If set to yes, the following variables will be set
+; just prior to the caller being bridged with a queue member (set on the
+; caller and queue member channels) 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;
+; QUEUECALLS number of calls currently in the queue
+; QUEUEHOLDTIME current average hold time
+; QUEUECOMPLETED number of completed calls for the queue
+; QUEUEABANDONED number of abandoned calls
+; QUEUESRVLEVEL queue service level
+; QUEUESRVLEVELPERF current service level performance
+;
+;setqueuevar=no
+
+; if set, run this macro when connected to the queue member
+; you can override this macro by setting the macro option on
+; the queue application
+;
+;membermacro=macro_name[,arg1[,...][,argN]]
+
+; if set, run this gosub when connected to the queue member
+; you can override this gosub by setting the gosub option on
+; the queue application
+;
+;membergosub=gosub_context_name[,arg1[,...][,argN]]
+
+; 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
+;
+; The absolute minimum time between the start of each
+; queue position and/or estimated holdtime announcement
+; This is useful for avoiding constant announcements
+; when the caller's queue position is changing frequently
+; (see announce-frequency)
+;
+;min-announce-frequency = 15
+;
+; How often to make any periodic announcement (see periodic-announce)
+;
+;periodic-announce-frequency=60
+;
+; Should the periodic announcements be played in a random order? Default is no.
+;
+;random-periodic-announce=no
+;
+; If set to yes, the periodic announcment frequency will be timed from the end
+; of each announcment rather than from the start of each announcment.  This
+; defaults to off.
+;
+;relative-periodic-announce=yes
+;
+; Should we include estimated hold time in position announcements?
+; Either yes, no, or only once.
+; Hold time will be announced as the estimated time.
+;
+;announce-holdtime = yes|no|once
+;
+; Queue position announce?
+; Valid values are "yes," "no," "limit," or "more." If set to "no," then the caller's position will
+; never be announced. If "yes," then the caller's position in the queue will be announced
+; to the caller. If set to "more," then if the number of callers is more than the number
+; specified by the announce-position-limit option, then the caller will hear that there
+; are more than that many callers waiting (i.e. if a caller number 6 is in a queue with the
+; announce-position-limit set to 5, then that caller will hear that there are more than 5
+; callers waiting). If set to "limit," then only callers within the limit specified by announce-position-limit
+; will have their position announced.
+;
+;announce-position = yes
+;
+; If enabled, play announcements to the first user waiting in the Queue. This may mean
+; that announcements are played when an agent attempts to connect to the waiting user,
+; which may delay the time before the agent and the user can communicate. Disabled by
+; default.
+;
+; announce-to-first-user = no
+;
+; If you have specified "limit" or "more" for the announce-position option, then the following
+; value is what is used to determine what announcement to play to waiting callers. If you have
+; set the announce-position option to anything else, then this will have no bearing on queue operation
+;
+;announce-position-limit = 5
+;
+; What's the rounding time for the seconds?
+; If this is non-zero, then we announce the seconds as well as the minutes
+; rounded to this value.
+; Valid values are 0, 5, 10, 15, 20, and 30.
+;
+; announce-round-seconds = 10
+;
+; Use these sound files in making position/holdtime announcements.  The
+; defaults are as listed below -- change only if you need to.
+;
+; Keep in mind that you may also prevent a sound from being played if you
+; explicitly set a sound to be an empty string. For example, if you want to
+; prevent the queue from playing queue-thankyou, you may set the sound using
+; the following line:
+;
+; queue-thankyou=
+;
+			;	("You are now first in line.")
+;queue-youarenext = queue-youarenext
+			;	("There are")
+;queue-thereare	= queue-thereare
+			;	("calls waiting.")
+;queue-callswaiting = queue-callswaiting
+			;	("The current est. holdtime is")
+;queue-holdtime = queue-holdtime
+			;	("minute.")
+;queue-minute = queue-minute
+			;	("minutes.")
+;queue-minutes = queue-minutes
+			;	("seconds.")
+;queue-seconds = queue-seconds
+			;	("Thank you for your patience.")
+;queue-thankyou = queue-thankyou
+			;       ("Hold time")
+;queue-reporthold = queue-reporthold
+			;       ("All reps busy / wait for next")
+;periodic-announce = queue-periodic-announce
+;
+; 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
+;
+; The announcements will be played in the order in which they are defined. After
+; playing the last announcement, the announcements begin again from the beginning.
+;
+; Calls may be recorded using Asterisk's monitor/MixMonitor resource
+; This can be enabled from within the Queue application, starting recording
+; when the call is actually picked up; thus, only successful calls are
+; recorded, and you are not recording while people are listening to MOH.
+; To enable monitoring, simply specify "monitor-format";  it will be disabled
+; otherwise.
+;
+; You can specify the monitor filename with by calling
+;    Set(MONITOR_FILENAME=foo)
+; Otherwise it will use MONITOR_FILENAME=${UNIQUEID}
+;
+; Pick any one valid extension for monitor format recording. If you leave
+; monitor-format commented out, it will not record calls.
+;
+; monitor-format = gsm|wav|wav49
+;
+; 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
+;    the concept of "joining/mixing" the in/out files now goes away
+;    when this is enabled. If you do not specify or comment out this option,
+;    it will default to the old 'Monitor' behavior to keep backward
+;    compatibility.
+;
+; monitor-type = MixMonitor
+;
+; ----------------------- TYPE MIXMONITOR OPTIONS -----------------------------
+;
+;
+; You can specify the options supplied to MixMonitor by calling (from the dialplan)
+;   Set(MONITOR_OPTIONS=av(<x>)V(<x>)W(<x>))
+; 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
+; recording by calling (from the dialplan)
+;
+;   Set(MONITOR_EXEC=mv /var/spool/asterisk/monitor/^{MONITOR_FILENAME} /tmp/^{MONITOR_FILENAME})
+;
+; or
+;
+;   Set(MONITOR_EXEC=mv /var/spool/asterisk/monitor/^{MIXMONITOR_FILENAME} /tmp/^{MIXMONITOR_FILENAME})
+;
+; If you choose to use the latter, you will not be able to switch the monitor-type back to Monitor
+; without changing this in the dialplan.
+;
+;
+; The command specified within the contents of MONITOR_EXEC will be executed when
+; the recording is over. Any strings matching ^{X} will be unescaped to ${X} and
+; all variables will be evaluated just prior to recording being started.
+;
+; The contents of MONITOR_FILENAME will also be unescaped from ^{X} to ${X} and
+; all variables will be evaluated just prior to recording being started.
+;
+; ---------------------- Queue Empty Options ----------------------------------
+;
+; Asterisk has provided the "joinempty" and "leavewhenempty" options for a while
+; with tenuous definitions of what they actually mean. The "joinempty" option controls
+; 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
+; 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
+; ringing: a member is not considered available if his phone is currently ringing
+; unavailable: This applies mainly to Agent channels. If the agent is a member of the queue
+;              but has not logged in, then do not consider the member to be available
+; invalid: Do not consider a member to be available if he has an "invalid" device state.
+;          This generally is caused by an error condition in the member's channel driver.
+; unknown: Do not consider a member to be available if we are unable to determine the member's
+;          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
+;
+; A caller will not be able to enter a queue if at least one member cannot be found
+; who is not paused, on the phone, or who has an invalid device state.
+;
+;leavewhenempty = inuse,ringing
+;
+; 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
+; options also accept the strings "yes" "no" "strict" and "loose". The following
+; serves as a translation for these values:
+;
+; yes - (empty) for joinempty; penalty,paused,invalid for leavewhenempty
+; no - penalty,paused,invalid for joinempty; (empty) for leavewhenempty
+; strict - penalty,paused,invalid,unavailable
+; loose - penalty,invalid
+;
+
+; If this is set to yes, the following manager events will be generated:
+; AgentCalled, AgentDump, AgentConnect, AgentComplete; setting this to
+; vars also sends all channel variables with the event.
+; (may generate some extra manager events, but probably ones you want)
+;
+; eventwhencalled = yes|no|vars
+;
+; If this is set to yes, the following manager events will be generated:
+; QueueMemberStatus
+; (may generate a WHOLE LOT of extra manager events)
+; The default value is yes and this can not be set globally.
+;
+; eventmemberstatus = no
+;
+; If you wish to report the caller's hold time to the member before they are
+; connected to the caller, set this to yes.
+;
+; reportholdtime = no
+;
+; If you want the queue to avoid sending calls to members whose devices are
+; known to be 'in use' (via the channel driver supporting that device state)
+; uncomment this option. This can be controlled on a per member basis by
+; setting 'ringinuse' on that member. This can be done in the member definition,
+; in the 'ringinuse' field on a realtime member, via the QUEUE_MEMBER dialplan
+; function, or with CLI/AMI. By default, the per member value will be the same
+; as the queue's ringinuse value if it isn't set on the member deliberately.
+; (Note: only the SIP channel driver currently is able to report 'in use'.)
+; ringinuse = no
+;
+; If you wish to have a delay before the member is connected to the caller (or
+; before the member hears any announcement messages), set this to the number of
+; seconds to delay.
+;
+; memberdelay = 0
+;
+; If timeoutrestart is set to yes, then the timeout for an agent to answer is
+; reset if a BUSY or CONGESTION is received.  This can be useful if agents
+; are able to cancel a call with reject or similar.
+;
+; timeoutrestart = no
+;
+; 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
+;
+; defaultrule = myrule
+;
+; Each member of this call queue is listed on a separate line in
+; the form technology/dialstring.  "member" means a normal member of a
+; queue.  An optional penalty may be specified after a comma, such that
+; entries with higher penalties are considered last.  An optional member
+; name may also be specified after a second comma, which is used in log
+; messages as a "friendly name".  Multiple interfaces may share a single
+; member name. An optional state interface may be specified after a third
+; comma. This interface will be the one for which app_queue receives device
+; state notifications, even though the first interface specified is the one
+; that is actually called.
+;
+; A hint can also be used in place of the state interface using the format
+; hint:<extension>@<context>. If no context is specified then 'default' will
+; be used.
+;
+; It is important to ensure that channel drivers used for members are loaded
+; before app_queue.so itself or they may be marked invalid until reload. This
+; can be accomplished by explicitly listing them in modules.conf before
+; app_queue.so.  Additionally, if you use Local channels as queue members, you
+; must also preload pbx_config.so and chan_local.so (or pbx_ael.so, pbx_lua.so,
+; or pbx_realtime.so, depending on how your dialplan is configured).
+;
+; syntax: member => interface,[,penalty][,membername][,state_interface][,ringinuse]
+;
+;member => DAHDI/1
+;member => DAHDI/2,10
+;member => DAHDI/3,10,Bob Johnson
+;member => Agent/1001
+;member => Agent/1002
+;member => Local/1000@default,0,John Smith,SIP/1000
+;member => Local/2000@default,0,Lorem Ipsum,SIP/2000,no
+
+;
+; Note that using agent groups is probably not what you want.  Strategies do
+; not propagate down to the Agent system so if you want round robin, least
+; recent, etc, you should list all the agents in this file individually and not
+; use agent groups.
+;
+;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
+

asterisk/res_config_mysql.conf

@@ -0,0 +1,43 @@
+;
+; Sample configuration for res_config_mysql.c
+;
+; The value of dbhost may be either a hostname or an IP address.
+; If dbhost is commented out or the string "localhost", a connection
+; to the local host is assumed and dbsock is used instead of TCP/IP
+; to connect to the server.  If no dbcharset is specified, the connection
+; is made with no extra charset configurations sent to MySQL, leaving all 
+; configured MySQL charset options and defaults untouched.
+;
+; Multiple database contexts may be configured, with the caveat that
+; all context names should be unique and must not contain the slash ('/')
+; character.  If you wish to separate reads from writes in your database
+; configuration, you specify the database (NOT HERE, in other files)
+; separated by a slash, read database first.  If your database
+; specification does not contain a slash, the implication is that reads
+; and writes should be performed to the same database.
+;
+; For example, in extconfig.conf, you could specify a line like:
+;    sippeers => mysql,readhost.asterisk/writehost.asterisk,sippeers
+; and then define the contexts [readhost.asterisk] and [writehost.asterisk]
+; below.
+;
+; The requirements parameter is available only in Asterisk 1.6.1 and
+; later and must be present in all contexts.  It specifies the behavior
+; when a column name is required by the system.  The default behavior is
+; "warn" and simply sends a warning to the logger that the column does
+; not exist (or is of the wrong type or precision).  The other two
+; possibilities are "createclose", which adds the column with the right
+; type and length, and "createchar", which adds the column as a char
+; type, with the appropriate length to accept the data.  Note that with
+; the MySQL driver, both "createclose" and "createchar" will, on occasion,
+; widen a table column width to meet the requirements specified.
+;
+[general]
+;dbhost = 127.0.0.1
+;dbname = asterisk
+;dbuser = myuser
+;dbpass = mypass
+;dbport = 3306
+;dbsock = /tmp/mysql.sock
+;dbcharset = latin1
+;requirements=warn ; or createclose or createchar

asterisk/res_config_sqlite.conf

@@ -0,0 +1,11 @@
+[general]
+
+; The database file.
+dbfile => /var/lib/asterisk/sqlite.db
+
+; Both config_table and cdr_table are optional. If config_table is omitted,
+; you must specify it in extconfig.conf. If it is both provided here and in
+; extconfig.conf, the value given here is used. If cdr_table is omitted, CDR
+; support is simply disabled.
+config_table => ast_config
+cdr_table => ast_cdr

asterisk/res_config_sqlite3.conf

@@ -0,0 +1,31 @@
+; Define a realtime database name to use in extconfig.conf
+;
+;[asterisk]
+;dbfile => /var/lib/asterisk/realtime.sqlite3
+;
+; debug - Turn on debugging information
+;debug=yes
+;
+; requirements - At startup, each realtime family will make requirements
+;   on the backend.  There are several strategies for handling requirements:
+;     warn        - Warn if the required column does not exist.
+;     createclose - Create columns as close to the requirements as possible.
+;     createchar  - Create char columns only
+;
+;requirements=warn
+;
+; batch - SQLite 3 write performance can be greatly improved by wrapping
+;   multiple writes in transactions. This option specifies the duration in
+;   milliseconds of auto-generated transactions. Any changes made during an
+;   unfinished transaction will be immediately available to the same database
+;   connection, but any external connections could see a delay up to the value
+;   of this setting. It is also possible that if asterisk crashes, any changes
+;   made during this time could be lost. Due to the nearly 100x performance
+;   benefit, the default is 100 ms. Set to 0 to disable batching.
+;
+;   PLEASE NOTE: If you need to write to the database from another connection
+;   you will need to set batch=0 as the transactions will cause the database
+;   to lock for writing.
+;
+;batch=1000
+

asterisk/res_corosync.conf

@@ -0,0 +1,31 @@
+;
+; Sample configuration file for res_corosync.
+;
+; This module allows events to be shared amongst a local cluster of
+; Asterisk servers.  Specifically, the types of events that may be
+; shared include:
+;
+;   - Device State (for shared presence information)
+;
+;   - Message Waiting Indication, or MWI (to allow Voicemail to live on
+;     a server that is different from where the phones are registered)
+;
+; For more information about Corosync, see: http://www.corosync.org/
+;
+
+[general]
+
+;
+;  Publish Message Waiting Indication (MWI) events from this server to the
+;  cluster.
+;publish_event = mwi
+;
+;  Subscribe to MWI events from the cluster.
+;subscribe_event = mwi
+;
+;  Publish Device State (presence) events from this server to the cluster.
+;publish_event = device_state
+;
+;  Subscribe to Device State (presence) events from the cluster.
+;subscribe_event = device_state
+;

asterisk/res_curl.conf

@@ -0,0 +1,8 @@
+;
+; CURLOPT global settings (mostly set for realtime)
+;
+[globals]
+;proxy=myproxy.example.com
+proxytype=http
+proxyport=8001
+;proxyuserpwd=asterisk:asteriskrocks

asterisk/res_fax.conf

@@ -0,0 +1,32 @@
+; Generic Fax Application configuration 
+
+[general]
+; Maximum Transmission Rate
+; Possible values are { 2400 | 4800 | 7200 | 9600 | 12000 | 14400 }
+; Set this value to the maximum desired transfer rate.  Default: 14400
+;maxrate=14400
+
+; Minimum Transmission Rate
+; Possible values are { 2400 | 4800 | 7200 | 9600 | 12000 | 14400 }
+; Set this value to the minimum desired transfer rate.  Default: 4800
+;minrate=4800
+
+; Send Progress/Status events to manager session
+; Manager events with 'call' class permissions will receive events indicating the
+; steps to initiate a fax session.  Fax completion events are always sent to manager
+; sessions with 'call' class permissions, regardless of the value of this option.
+; Default: no
+statusevents=yes
+
+; modem capabilities
+; Possible values are { v17 | v27 | v29 }
+; Set this value to modify the default modem options.  Default: v17,v27,v29
+;modems=v17,v27,v29
+
+; Enable/disable T.30 ECM (error correction mode) by default.
+; Default: Enabled
+;ecm=yes
+
+; T.38 Negotiation Timeout in milliseconds
+; Default: 5000
+t38timeout=5000

asterisk/res_ldap.conf

@@ -0,0 +1,198 @@
+;
+; Configuration file for res_config_ldap
+;
+
+; Realtime configuration
+; ----------------------
+; In order to use this module, you start 
+; in extconfig.conf with a configuration like this:
+;
+; 	sippeers = ldap,"dc=myDomain,dc=myDomainExt",sip
+; 	extensions = ldap,"dc=myDomain,dc=myDomainExt",extensions
+; 	sip.conf = ldap,"dc=myDomain,dc=myDomainExt",config
+;
+; In the case of LDAP the last keyword in each line above specifies
+; a section in this file.
+;
+; LDAP schema and ldif files can be located in contrib/scripts.
+
+; TLS support
+; -----------
+; Note that you can configure an ldaps: url here to get TLS support.
+; Detailed configuration of certificates and supported CAs is done in your 
+; ldap.conf file for OpenLDAP clients on your system.
+; This requires that you have OpenLDAP libraries compiled with TLS support
+
+; *********************************************************************************
+; NOTE: res_ldap.conf should be chmod 600 because it contains the plain-text LDAP
+;       password to an account with WRITE access to the asterisk configuration. 
+; *********************************************************************************
+
+[_general]
+;
+; Specify one of either host and port OR url.  URL is preferred, as you can
+; use more options.
+;host=192.168.1.1                    ; LDAP host
+;port=389
+;url=ldap://ldap3.mydomain.com:3890
+;protocol=3                          ; Version of the LDAP protocol to use; default is 3.
+;basedn=dc=example,dc=tld            ; Base DN
+;user=cn=asterisk,dc=example,dc=tld  ; Bind DN
+;pass=MyPassword                     ; Bind password
+
+; Configuration Table
+[config]
+;
+; additionalFilter - This specifies an additional set of criteria to be used
+; when querying the LDAP server.
+;
+additionalFilter=(objectClass=AstConfig)
+;
+; Attributes mapping (asterisk variable name = ldap attribute name)
+; When Asterisk requests the variable by the name of the value on the left,
+; this module will look up the attribute listed on the right.
+;
+filename = AstConfigFilename
+category = AstConfigCategory
+variable_name = AstConfigVariableName
+variable_value = AstConfigVariableValue
+cat_metric = AstConfigCategoryMetric
+commented = AstConfigCommented
+
+;
+; Extensions Table
+;
+[extensions]
+context  =  AstExtensionContext
+exten  =  AstExtensionExten
+priority = AstExtensionPriority
+app = AstExtensionApplication
+appdata = AstExtensionApplicationData
+additionalFilter=(objectClass=AstExtension)
+
+;
+; Sip Users Table
+;
+[sip]
+name = cn       ; We use the "cn" as the default value for name on the line above
+                ; because objectClass=AsteriskSIPUser does not include a uid as an allowed field
+                ; If your entry combines other objectClasses and uid is available, you may
+                ; prefer to change the line to be name = uid, especially if your LDAP entries
+                ; contain spaces in the cn field.
+                ; You may also find it appropriate to use something completely different.
+                ; This is possible by changing the line above to name = AstAccountName (or whatever you
+                ; prefer).
+                ;
+amaflags = AstAccountAMAFlags
+callgroup = AstAccountCallGroup
+callerid = AstAccountCallerID
+directmedia = AstAccountDirectMedia
+context = AstAccountContext
+dtmfmode = AstAccountDTMFMode
+fromuser = AstAccountFromUser
+fromdomain = AstAccountFromDomain
+fullcontact = AstAccountFullContact
+fullcontact = gecos
+host = AstAccountHost
+insecure = AstAccountInsecure
+mailbox = AstAccountMailbox
+md5secret = AstAccountRealmedPassword           ; Must be an MD5 hash. Field value can start with
+                                                ; {md5} but it is not required.
+                                                ; Generate the password via the md5sum command, e.g.
+                                                ; echo "my_password" | md5sum
+nat = AstAccountNAT
+deny = AstAccountDeny
+permit = AstAccountPermit
+pickupgroup = AstAccountPickupGroup
+port = AstAccountPort
+qualify = AstAccountQualify
+restrictcid = AstAccountRestrictCID
+rtptimeout = AstAccountRTPTimeout
+rtpholdtimeout = AstAccountRTPHoldTimeout
+type = AstAccountType
+disallow = AstAccountDisallowedCodec
+allow = AstAccountAllowedCodec
+MusicOnHold = AstAccountMusicOnHold
+regseconds = AstAccountExpirationTimestamp
+regcontext = AstAccountRegistrationContext
+regexten = AstAccountRegistrationExten
+CanCallForward = AstAccountCanCallForward
+ipaddr = AstAccountIPAddress
+defaultuser = AstAccountDefaultUser
+regserver = AstAccountRegistrationServer
+lastms = AstAccountLastQualifyMilliseconds
+additionalFilter=(objectClass=AsteriskSIPUser)
+
+;
+; IAX Users Table
+;
+[iax]
+amaflags = AstAccountAMAFlags
+callerid = AstAccountCallerID
+context = AstAccountContext
+fullcontact = AstAccountFullContact
+fullcontact = gecos
+host = AstAccountHost
+mailbox = AstAccountMailbox
+md5secret = AstAccountRealmedPassword           ; Must be an MD5 hash. Field value can start with
+                                                ; {md5} but it is not required.
+                                                ; Generate the password via the md5sum command, e.g.
+                                                ; echo "my_password" | md5sum
+deny = AstAccountDeny
+permit = AstAccountPermit
+port = AstAccountPort
+qualify = AstAccountQualify
+type = AstAccountType
+disallow = AstAccountDisallowedCodec
+allow = AstAccountAllowedCodec
+regseconds = AstAccountExpirationTimestamp
+regcontext = AstAccountRegistrationContext
+regexten = AstAccountRegistrationExten
+notransfer = AstAccountNoTransfer
+lastms = AstAccountLastQualifyMilliseconds
+additionalFilter=(objectClass=AstAccountIAX)
+
+;
+; A Test Family
+;
+[testfamily]
+MyUSERID = uid
+additionalFilter=(objectClass=*)
+
+[accounts]
+amaflags = AstAccountAMAFlags
+callgroup = AstAccountCallGroup
+callerid = AstAccountCallerID
+directmedia = AstAccountDirectMedia
+context = AstAccountContext
+dtmfmode = AstAccountDTMFMode
+fromuser = AstAccountFromUser
+fromdomain = AstAccountFromDomain
+fullcontact = AstAccountFullContact
+fullcontact = gecos
+host = AstAccountHost
+insecure = AstAccountInsecure
+mailbox = AstAccountMailbox
+md5secret = AstAccountRealmedPassword           ; Must be an MD5 hash. Field value can start with
+                                                ; {md5} but it is not required.
+                                                ; Generate the password via the md5sum command, e.g.
+                                                ; echo "my_password" | md5sum
+nat = AstAccountNAT
+deny = AstAccountDeny
+permit = AstAccountPermit
+pickupgroup = AstAccountPickupGroup
+port = AstAccountPort
+qualify = AstAccountQualify
+restrictcid = AstAccountRestrictCID
+rtptimeout = AstAccountRTPTimeout
+rtpholdtimeout = AstAccountRTPHoldTimeout
+type = AstAccountType
+disallow = AstAccountDisallowedCodec
+allow = AstAccountAllowedCodec
+MusicOnHold = AstAccountMusicOnHold
+regseconds = AstAccountExpirationTimestamp
+regcontext = AstAccountRegistrationContext
+regexten = AstAccountRegistrationExten
+CanCallForward = AstAccountCanCallForward
+additionalFilter=(objectClass=AstAccount)
+

asterisk/res_odbc.conf

@@ -0,0 +1,131 @@
+;;; 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,
+; so you can't have different values for different connections.
+[ENV]
+;INFORMIXSERVER => my_special_database
+;INFORMIXDIR => /opt/informix
+;ORACLE_HOME => /home/oracle
+
+; All other sections are arbitrary names for database connections.
+
+;
+; The context name is what will be used in other configuration files, such
+; as extconfig.conf and func_odbc.conf, to reference this connection.
+[asterisk]
+;
+; Permit disabling sections without needing to comment them out.
+; If not specified, it is assumed the section is enabled.
+enabled => no
+;
+; This value should match an entry in /etc/odbc.ini
+; (or /usr/local/etc/odbc.ini, on FreeBSD and similar systems).
+dsn => asterisk
+;
+; Username for connecting to the database.  The user defaults to the context
+; name if unspecified.
+;username => myuser
+;
+; Password for authenticating the user to the database.  The default
+; password is blank.
+;password => mypass
+;
+; Build a connection at startup?
+pre-connect => yes
+;
+; What should we execute to ensure that our connection is still alive?  The
+; statement should return a non-zero value in the first field of its first
+; record.  The default is "select 1".
+;sanitysql => select 1
+;
+; On some databases, the connection times out and a reconnection will be
+; necessary.  This setting configures the amount of time a connection
+; may sit idle (in seconds) before a reconnection will be attempted.
+;idlecheck => 3600
+;
+; Should we use a single connection for all queries?  Most databases will
+; allow sharing the connection, though Sybase and MS SQL Server will not.
+;share_connections => yes
+;
+; If we aren't sharing connections, what is the maximum number of connections
+; that we should attempt?
+;limit => 5
+;
+; When the channel is destroyed, should any uncommitted open transactions
+; automatically be committed?
+;forcecommit => no
+;
+; How should we perceive data in other transactions within the database?
+; Possible values are read_uncommitted, read_committed, repeatable_read,
+; and serializable.  The default is read_committed.
+;isolation => repeatable_read
+;
+; Is the backslash a native escape character?  The default is yes, but for
+; MS SQL Server, the answer is no.
+;backslash_is_escape => yes
+;
+; When enabled (default behavior), empty column values are stored as empty strings
+; during realtime updates. Disabling this option causes empty column values to be
+; stored as NULLs for non-text columns.
+; Disable it for PostgreSQL backend in order to avoid errors caused by updating
+; integer columns with an empty string instead of NULL (sippeers, sipregs, ..).
+; NOTE: This option will be removed in asterisk 13. At that point, it will always
+; behave as if it was set to 'no'.
+;allow_empty_string_in_nontext => yes
+;
+; How long (in seconds) should we attempt to connect before considering the
+; connection dead?  The default is 10 seconds, but you may wish to reduce it,
+; to increase responsiveness.
+;connect_timeout => 10
+;
+; When a connection fails, how long (in seconds) should we cache that
+; information before we attempt another connection?  This increases
+; responsiveness, when a database resource is not working.
+;negative_connection_cache => 300
+
+[mysql2]
+enabled => no
+dsn => MySQL-asterisk
+username => myuser
+password => mypass
+pre-connect => yes
+
+; Certain servers, such as MS SQL Server and Sybase use the TDS protocol, which
+; limits the number of active queries per connection to 1.  By telling res_odbc
+; not to share connections, Asterisk can be made to work with these servers.
+[sqlserver]
+enabled => no
+dsn => mickeysoft
+share_connections => no
+limit => 5
+username => oscar
+password => thegrouch
+pre-connect => yes
+sanitysql => select count(*) from systables
+; forcecommit => no            ; Default to committing uncommitted transactions?
+                               ; Note:  this is NOT the autocommit flag; this
+                               ; determines the end result of transactions which
+                               ; are not explicitly committed or rolled back.  By
+                               ; default, such transactions are rolled back if the
+                               ; call ends without an explicit commit.
+; 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.
+;
+; Many databases have a default of '\' to escape special characters.  MS SQL
+; Server does not.
+backslash_is_escape => no
+
+;
+; If you are having problems with concurrency, please read this note from the
+; mailing lists, regarding UnixODBC:
+;
+; http://lists.digium.com/pipermail/asterisk-dev/2009-February/036539.html
+;
+; In summary, try setting "Threading=2" in the relevant section within your
+; odbcinst.ini.
+;

asterisk/res_pgsql.conf

@@ -0,0 +1,29 @@
+;
+; Sample configuration for res_config_pgsql
+;
+; The value of dbhost may be either a hostname or an IP address.
+; If dbhost is commented out or the string "localhost", a connection
+; to the local host is assumed and dbsock is used instead of TCP/IP
+; to connect to the server.
+;
+[general]
+dbhost=127.0.0.1
+dbport=5432
+dbname=asterisk
+dbuser=asterisk
+dbpass=password
+;
+; dbsock is specified as the directory where the socket file may be found. The
+; actual socket is constructed as a combination of dbsock and dbport.  For
+; example, the values of '/tmp' and '5432', respectively, will specify a socket
+; file of '/tmp/.s.PGSQL.5432'.
+; 
+;dbsock=/tmp
+;
+; requirements - At startup, each realtime family will make requirements
+; on the backend.  There are several strategies for handling requirements:
+; warn        - Warn if the required column does not exist.
+; createclose - Create columns as close to the requirements as possible.
+; createchar  - Create char columns only
+;
+requirements=warn

asterisk/res_pktccops.conf

@@ -0,0 +1,32 @@
+;; Sample res_pktccops.conf
+;
+;[general]
+;gateinfoperiod => 60 ; default 60s
+;gatetimeout = 150 ; default 150
+;t1 => 250 ; default 250s
+;t7 => 200 ; default 200s
+;t8 => 300 ; default 300s
+;keepalive => 60 ; default 60s
+;
+;[teszt]
+;host => 192.168.0.24
+;pool => 10.0.1.0 10.0.1.255
+;pool => 10.0.3.0 10.0.3.255
+;pool => 10.0.7.0 10.0.8.255
+;pool => 10.0.10.0 10.0.11.255
+;
+;[general]
+;gateinfoperiod => 60 ; default 60s
+;gatetimeout = 150 ; default 150
+;t1 => 250 ; default 250s
+;t7 => 200 ; default 200s
+;t8 => 300 ; default 300s
+;keepalive => 60 ; default 60s
+;
+;[test]
+;host => 192.168.0.24
+;pool => 10.0.1.0 10.0.1.255
+;pool => 10.0.3.0 10.0.3.255
+;pool => 10.0.7.0 10.0.8.255
+;pool => 10.0.10.0 10.0.11.255
+;

asterisk/res_snmp.conf

@@ -0,0 +1,22 @@
+;
+; Configuration file for res_snmp
+;---------------------------------
+;
+; Res_snmp can run as a subagent or standalone SNMP agent. The standalone snmp
+; agent is based on net-snmp and will read a configuration file called
+; asterisk.conf in the net-snmp configuration file path, starting with
+; /etc/snmp on many systems.
+;
+; If you use the subagent model, you need to enable agentx in snmpd.conf
+; Note that you can only run one Asterisk on the system in this case.
+;
+; See https://wiki.asterisk.org/wiki/display/AST/Simple+Network+Management+Protocol+(SNMP)+Support 
+; to get more information about
+; snmp support in Asterisk
+
+[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)
+;subagent = yes
+; SNMP must be explicitly enabled to be active
+;enabled = yes

asterisk/res_stun_monitor.conf

@@ -0,0 +1,27 @@
+;
+; Configuration file for the res_stun_monitor module
+;
+; The res_stun_monitor module sends STUN requests to a configured STUN server
+; periodically.  If the monitor detects a change in the external IP address or port
+; provided by the STUN server an event is sent out internally within Asterisk
+; to alert all listeners to that event of the change.
+
+; The current default listeners for the network change event include chan_sip
+; and chan_iax.  Both of these channel drivers by default react to this event
+; by renewing all outbound registrations.  This allows the endpoints Asterisk
+; is registering with to become aware of the address change and know the new
+; location.
+;
+[general]
+;
+; ---- STUN Server configuration ---
+;  Setting the 'stunaddr' option to a valid address enables the STUN monitor.
+;
+;stunaddr = mystunserver.com    ; Address of the STUN server to query.
+                                ; Valid form:
+                                ;   [(hostname | IP-address) [':' port]]
+                                ; The port defaults to the standard STUN port (3478).
+                                ; Set to an empty value to disable STUN monitoring.
+                                ;   Default is disabled.
+;stunrefresh = 30               ; Number of seconds between STUN refreshes.
+                                ;   Default is 30.

asterisk/rtp.conf

@@ -0,0 +1,60 @@
+;
+; RTP Configuration
+;
+[general]
+;
+; RTP start and RTP end configure start and end addresses
+;
+; Defaults are rtpstart=5000 and rtpend=31000
+;
+rtpstart=10000
+rtpend=20000
+;
+; Whether to enable or disable UDP checksums on RTP traffic
+;
+;rtpchecksums=no
+;
+; The amount of time a DTMF digit with no 'end' marker should be
+; allowed to continue (in 'samples', 1/8000 of a second)
+;
+;dtmftimeout=3000
+; 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
+; enabled by default.
+; strictrtp=yes
+;
+; Number of packets containing consecutive sequence values needed
+; to change the RTP source socket address. This option only comes
+; into play while using strictrtp=yes. Consider changing this value
+; if rtp packets are dropped from one or both ends after a call is
+; connected. This option is set to 4 by default.
+; probation=8
+;
+; Whether to enable or disable ICE support. This option is disabled by default.
+; icesupport=true
+;
+; Hostname or address for the STUN server used when determining the external
+; IP address and port an RTP session can be reached at. The port number is
+; optional. If omitted the default value of 3478 will be used. This option is
+; disabled by default.
+;
+; e.g. stundaddr=mystun.server.com:3478
+;
+; stunaddr=
+;
+; Hostname or address for the TURN server to be used as a relay. The port
+; number is optional. If omitted the default value of 3478 will be used.
+; This option is disabled by default.
+;
+; e.g. turnaddr=myturn.server.com:34780
+;
+; turnaddr=
+;
+; Username used to authenticate with TURN relay server.
+; turnusername=
+;
+; Password used to authenticate with TURN relay server.
+; turnpassword=

asterisk/say.conf

@@ -0,0 +1,383 @@
+;
+; language configuration
+;
+
+[general]
+mode=old	; method for playing numbers and dates
+		; old - using asterisk core function
+		; new - using this configuration file
+
+; The new language routines produce strings of the form
+;	prefix:[format:]data
+; that are matched against the rules in this file to produce
+; an output.
+;
+; The data is generally the string to be spelled (either a number,
+; an ascii string or a date/time in the format specified below).
+; It is available, in the right hand side of a rule, as variable ${SAY}.
+;
+; The format is optional and normally used only for date/time.
+; The prefix is used to select the pronunciation - standard
+; prefixes are
+;	num		used for numbers
+;	enum		used for enumerations
+;	date 		for dates
+;	time 		for times
+;	datetime	for dates and times
+;	char		for character strings
+;	phonetic	for phonetic strings
+;	digit		for digit strings
+;
+; but others can be used at will.
+;
+; Processing occurs as follows:
+; If the format is empty, or there is no format, the entire
+; string is matched against one of the pattern on the left hand side.
+; On the first match, the various comma-separated components on the right
+; hand side are pronounced, as follows:
+; + a component starting with a prefix: (i.e. with a ':' in it)
+;   is re-processed according to these rules;
+; + a component without a ':' in it is considered a filename and
+;   the corresponding file is played.
+;
+; If the format is non-empty, the format is split into its components
+; (individual characters, or filenames in single quotes), and then
+; filenames are played, whereas single characters are used to
+; generate a new string format:pat:data to be processed.
+;
+; DATES/AND TIMES assume that the date info is available in
+; the form	 YYYYMMDDHHmm.ss-dow-doy
+; with 4 digits for the year, 2 for month, day, hour, minutes, seconds,
+; one digit for the day-of-week, and 3 digits for the day-of-year.
+;
+; Example:
+;     datetime::200604172030.00-4-102
+; (typical format for a date) is first matched against the line
+;	datetime::. => date:AdBY 'digits/at' IMp:${SAY}
+; which is normally present with the default format for dates.
+; In turn, the format string "AdBY 'digits/at' IMp" results in
+; the sequence
+;	date:A:200604172030.00-4-102
+;	date:d:200604172030.00-4-102
+;	date:B:200604172030.00-4-102
+;	date:Y:200604172030.00-4-102
+;	digits/at
+;	date:I:200604172030.00-4-102
+;	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.
+
+; We start with the basic rules that might be more-or-less
+; 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:${SAY:1}
+
+[date-base](!)		; base rules for dates and times
+    ; the 'SAY' variable contains YYYYMMDDHHmm.ss-dow-doy
+    ; these rule map the strftime attributes.
+    _date:Y:. => num:${SAY:0:4}	; year, 19xx
+    _date:[Bbh]:. => 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:[H]:. => digits/oh, num:${SAY:8:2}			; hour (oh one, oh two, ..., oh nine, ten, eleven, ..., twenty-three)
+    _date:[I]:. => num:$[${SAY:8:2} % 12]	; hour 0-12
+    _date:[M]:. => num:${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
+    _date:[Ii]:. => num:$[${SAY:8:2} % 12]			; hour 0-12
+    _date:[k]:. => num:${SAY:8:2}				; hour (one, two. three, ...,twenty three
+    _date:[m]:. => num:${SAY:4:2}				; month number
+    _date:[Q]:. => date:dby ;incompleat				; "today", "yesterday" or ABdY
+    _date:[q]:. => date:dby ;incompleat				; "" (for today), "yesterday", weekday, or ABdY
+    _date:[R]:. => date:HM${SAY}				; 24 hour time, including minute
+    _date:[T]:. => date:HMS${SAY}				; 24 hour, minure, 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:X00 => num:${SAY:0:1}, digits/hundred
+    _[n]um:XXX => num:${SAY:0:1}, digits/hundred, num:${SAY:1}
+
+    _[n]um:X000 => num:${SAY:0:1}, digits/thousand
+    _[n]um:XXXX => num:${SAY:0:1}, digits/thousand, num:${SAY:1}
+    _[n]um:XX000 => num:${SAY:0:2}, digits/thousand
+    _[n]um:XXXXX => num:${SAY:0:2}, digits/thousand, num:${SAY:2}
+    _[n]um:XXX000 => num:${SAY:0:3}, digits/thousand
+    _[n]um:XXXXXX => num:${SAY:0:3}, digits/thousand, num:${SAY:3}
+
+    _[n]um:X000000 => num:${SAY:0:1}, digits/million
+    _[n]um:XXXXXXX => num:${SAY:0:1}, digits/million, num:${SAY:1}
+    _[n]um:XX000000 => num:${SAY:0:2}, digits/million
+    _[n]um:XXXXXXXX => num:${SAY:0:2}, digits/million, num:${SAY:2}
+    _[n]um:XXX000000 => num:${SAY:0:3}, digits/million
+    _[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/million, num:${SAY:3}
+
+    _[n]um:X000000000 => num:${SAY:0:1}, digits/billion
+    _[n]um:XXXXXXXXXX => num:${SAY:0:1}, digits/billion, num:${SAY:1}
+    _[n]um:XX000000000 => num:${SAY:0:2}, digits/billion
+    _[n]um:XXXXXXXXXXX => num:${SAY:0:2}, digits/billion, num:${SAY:2}
+    _[n]um:XXX000000000 => num:${SAY:0:3}, digits/billion
+    _[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-${SAY:1}
+    _e[n]um:[1-9]00 => num:${SAY:0:1}, digits/h-hundred
+    _e[n]um:[1-9]XX => num:${SAY:0:1}, digits/h-hundred, enum:${SAY:1}
+
+[en_GB](date-base,digit-base,en-base)
+    _[n]um:XXX => num:${SAY:0:1}, digits/hundred, vm-and, num:${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:100 => digits/hundred
+    _[n]um:1XX => digits/hundred, num:${SAY:1}
+    _[n]um:[2-9]00 => num:${SAY:0:1}, digits/hundred
+    _[n]um:[2-9]XX => num:${SAY:0:1}, digits/hundred, num:${SAY:1}
+
+    _[n]um:1000 => digits/thousand
+    _[n]um:1XXX => digits/thousand, num:${SAY:1}
+    _[n]um:[2-9]000 => num:${SAY:0:1}, digits/thousands
+    _[n]um:[2-9]XXX => num:${SAY:0:1}, digits/thousands, num:${SAY:1}
+    _[n]um:XX000 => num:${SAY:0:2}, digits/thousands
+    _[n]um:XXXXX => num:${SAY:0:2}, digits/thousands, num:${SAY:2}
+    _[n]um:XXX000 => num:${SAY:0:3}, digits/thousands
+    _[n]um:XXXXXX => num:${SAY:0:3}, digits/thousands, num:${SAY:3}
+
+    _[n]um:1000000 => num:${SAY:0:1}, digits/million
+    _[n]um:1XXXXXX => num:${SAY:0:1}, digits/million, num:${SAY:1}
+    _[n]um:[2-9]000000 => num:${SAY:0:1}, digits/millions
+    _[n]um:[2-9]XXXXXX => num:${SAY:0:1}, digits/millions, num:${SAY:1}
+    _[n]um:XX000000 => num:${SAY:0:2}, digits/millions
+    _[n]um:XXXXXXXX => num:${SAY:0:2}, digits/millions, num:${SAY:2}
+    _[n]um:XXX000000 => num:${SAY:0:3}, digits/millions
+    _[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/millions, num:${SAY:3}
+
+    _[n]um:X000000000 => num:${SAY:0:1}, digits/billion
+    _[n]um:XXXXXXXXXX => num:${SAY:0:1}, digits/billion, num:${SAY:1}
+    _[n]um:XX000000000 => num:${SAY:0:2}, digits/billion
+    _[n]um:XXXXXXXXXXX => num:${SAY:0:2}, digits/billion, num:${SAY:2}
+    _[n]um:XXX000000000 => num:${SAY:0:3}, digits/billion
+    _[n]um:XXXXXXXXXXXX => num:${SAY:0:3}, digits/billion, 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:${SAY}
+
+[de](date-base,digit-base)
+    _[n]um:0. => num:${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/${SAY:0:1}0
+    _[n]um:100 => digits/ein, digits/hundred
+    _[n]um:1XX => digits/ein, digits/hundred, num:${SAY:1}
+    _[n]um:[2-9]00 => digits/${SAY:0:1}, digits/hundred
+    _[n]um:[2-9]XX => digits/${SAY:0:1}, digits/hundred, num:${SAY:1}
+
+    _[n]um:1000 => digits/ein, digits/thousand
+    _[n]um:1XXX => digits/ein, digits/thousand, num:${SAY:1}
+    _[n]um:[2-9]000 => digits/${SAY:0:1}, digits/thousand
+    _[n]um:[2-9]XXX => digits/${SAY:0:1}, digits/thousand, num:${SAY:1}
+    _[n]um:XX000 => num:${SAY:0:2}, digits/thousand
+    _[n]um:XXXXX => num:${SAY:0:2}, digits/thousand, num:${SAY:2}
+    _[n]um:XXX000 => num:${SAY:0:3}, digits/thousand
+    _[n]um:XXXXXX => num:${SAY:0:3}, digits/thousand, num:${SAY:1}
+
+    _[n]um:1000000 => digits/eine, digits/million
+    _[n]um:1XXXXXX => digits/eine, digits/million, num:${SAY:1}
+    _[n]um:[2-9]000000 => digits/${SAY:0:1}, digits/millions
+    _[n]um:[2-9]XXXXXX => digits/${SAY:0:1}, digits/millions, num:${SAY:1}
+    _[n]um:XX000000 => num:${SAY:0:2}, digits/millions
+    _[n]um:XXXXXXXX => num:${SAY:0:2}, digits/millions, num:${SAY:2}
+    _[n]um:XXX000000 => num:${SAY:0:3}, digits/millions
+    _[n]um:XXXXXXXXX => num:${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:${SAY:1}
+
+    _[n]um:X000 => num:${SAY:0:1}, digits/thousand
+    _[n]um:XXXX => num:${SAY:0:1}, digits/thousand, num:${SAY:1}
+    _[n]um:XX000 => num:${SAY:0:2}, digits/thousand
+    _[n]um:XXXXX => num:${SAY:0:2}, digits/thousand, num:${SAY:2}
+    _[n]um:XXX000 => num:${SAY:0:3}, digits/thousand
+    _[n]um:XXXXXX => num:${SAY:0:3}, digits/thousand, num:${SAY:3}
+
+    _[n]um:X000000 => num:${SAY:0:1}, digits/million
+    _[n]um:XXXXXXX => num:${SAY:0:1}, digits/million, num:${SAY:1}
+    _[n]um:XX000000 => num:${SAY:0:2}, digits/million
+    _[n]um:XXXXXXXX => num:${SAY:0:2}, digits/million, num:${SAY:2}
+    _[n]um:XXX000000 => num:${SAY:0:3}, digits/million
+    _[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/million, num:${SAY:3}
+
+    _[n]um:X000000000 => num:${SAY:0:1}, digits/billion
+    _[n]um:XXXXXXXXXX => num:${SAY:0:1}, digits/billion, num:${SAY:1}
+    _[n]um:XX000000000 => num:${SAY:0:2}, digits/billion
+    _[n]um:XXXXXXXXXXX => num:${SAY:0:2}, digits/billion, num:${SAY:2}
+    _[n]um:XXX000000000 => num:${SAY:0:3}, digits/billion
+    _[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':${SAY}
+
+[fr](date-base,digit-base)
+    _[n]um:0. => num:${SAY:1}
+    _[n]um:X => digits/${SAY}
+    _[n]um:1X => digits/${SAY}
+    _[n]um:[2-9]0 =>  digits/${SAY}
+    _[n]um:[2-6]1 => digits/${SAY:0:1}0, vm-and, digits/${SAY:1}
+    _[n]um:71 => digits/60, vm-and, num:1${SAY:1}
+    _[n]um:7X => digits/60, num:1${SAY:1}
+    _[n]um:9X => digits/80, num:1${SAY:1}
+    _[n]um:[2-9][1-9] =>  digits/${SAY:0:1}0, num:${SAY:1}
+    _[n]um:100 => digits/hundred
+    _[n]um:1XX => digits/hundred, num:${SAY:1}
+    _[n]um:[2-9]00 => num:${SAY:0:1}, digits/hundred
+    _[n]um:[2-9]XX => num:${SAY:0:1}, digits/hundred, num:${SAY:1}
+    
+    _[n]um:1000 => digits/thousand
+    _[n]um:1XXX => digits/thousand, num:${SAY:1}
+    _[n]um:[2-9]000 => num:${SAY:0:1}, digits/thousand
+    _[n]um:[2-9]XXX => num:${SAY:0:1}, digits/thousand, num:${SAY:1}
+    _[n]um:XX000 => num:${SAY:0:2}, digits/thousand
+    _[n]um:XXXXX => num:${SAY:0:2}, digits/thousand, num:${SAY:2}
+    _[n]um:XXX000 => num:${SAY:0:3}, digits/thousand
+    _[n]um:XXXXXX => num:${SAY:0:3}, digits/thousand, num:${SAY:3}
+    
+    _[n]um:1000000 => num:${SAY:0:1}, digits/million
+    _[n]um:1XXXXXX => num:${SAY:0:1}, digits/million, num:${SAY:1}
+    _[n]um:[2-9]000000 => num:${SAY:0:1}, digits/million
+    _[n]um:[2-9]XXXXXX => num:${SAY:0:1}, digits/million, num:${SAY:1}
+    _[n]um:XX000000 => num:${SAY:0:2}, digits/million
+    _[n]um:XXXXXXXX => num:${SAY:0:2}, digits/million, num:${SAY:2}
+    _[n]um:XXX000000 => num:${SAY:0:3}, digits/million
+    _[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/million, num:${SAY:3}
+    
+    _datetime::. => date:AdBY 'digits/at' H 'hours' M 'perc':${SAY}  
+    _date::. => date:AdBY:${SAY}
+    _time::. => date:H 'hours' M 'perc':${SAY}
+    
+    _pho[n]e:XXXX => num:${SAY:0:2}, num:${SAY:2:2}
+    _pho[n]e:0[1-9]XXXXXXXX => num:${SAY:0:1}, num:${SAY:1:1}, num:${SAY:2:2}, num:${SAY:4:2}, num:${SAY:6:2}, num:${SAY:8:2}
+    _pho[n]e:. => digit:${SAY}
+
+[es](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[1-9] => digits/20-and, digits/${SAY:1}
+    _[n]um:[3-9][1-9] =>  digits/${SAY:0:1}0, vm-and, digits/${SAY:1}
+    _[n]um:100 => digits/hundred
+    _[n]um:1XX => digits/100-and, num:${SAY:1}
+    _[n]um:[2-9]00 => digits/${SAY}
+    _[n]um:[2-9]XX => digits/${SAY:0:1}00, num:${SAY:1}
+
+    _[n]um:1000 => digits/thousand
+    _[n]um:1XXX => digits/thousand, num:${SAY:1}
+    _[n]um:[2-9]000 => num:${SAY:0:1}, digits/thousand
+    _[n]um:[2-9]XXX => num:${SAY:0:1}, digits/thousand, num:${SAY:1}
+    _[n]um:XX000 => num:${SAY:0:2}, digits/thousand
+    _[n]um:XXXXX => num:${SAY:0:2}, digits/thousand, num:${SAY:2}
+    _[n]um:XXX000 => num:${SAY:0:3}, digits/thousand
+    _[n]um:XXXXXX => num:${SAY:0:3}, digits/thousand, num:${SAY:3}
+
+    _[n]um:1000000 => digits/1M, digits/million
+    _[n]um:1XXXXXX => digits/1M, digits/million, num:${SAY:1}
+    _[n]um:[2-9]000000 => num:${SAY:0:1}, digits/millions
+    _[n]um:[2-9]XXXXXX => num:${SAY:0:1}, digits/millions, num:${SAY:1}
+    _[n]um:XX000000 => num:${SAY:0:2}, digits/millions
+    _[n]um:XXXXXXXX => num:${SAY:0:2}, digits/millions, num:${SAY:2}
+    _[n]um:XXX000000 => num:${SAY:0:3}, digits/millions
+    _[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/millions, num:${SAY:3}
+    _[n]um:XXXX000000 => num:${SAY:0:4}, digits/millions
+    _[n]um:XXXXXXXXXX => num:${SAY:0:4}, digits/millions, num:${SAY:4}
+
+    _datetime::. => date:Ad 'letters/d' B 'letters/d' Y 'digits/at' HMp:${SAY}
+    _date::. => date:Ad 'letters/d' B 'letters/d' Y:${SAY}
+    _time::. => date:HMp:${SAY}
+
+[da](date-base,digit-base)
+    _[n]um:0. => num:${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/${SAY:0:1}0
+    _[n]um:100 => digits/1N, digits/hundred
+    _[n]um:1XX => digits/1N, digits/hundred, num:${SAY:1}
+    _[n]um:[2-9]00 => digits/${SAY:0:1}, digits/hundred
+    _[n]um:[2-9]XX => digits/${SAY:0:1}, digits/hundred, num:${SAY:1}
+
+    _[n]um:1000 => digits/1N, digits/thousand
+    _[n]um:1XXX => digits/1N, digits/thousand, num:${SAY:1}
+    _[n]um:[2-9]000 => digits/${SAY:0:1}, digits/thousand
+    _[n]um:[2-9]XXX => digits/${SAY:0:1}, digits/thousand, num:${SAY:1}
+    _[n]um:XX000 => num:${SAY:0:2}, digits/thousand
+    _[n]um:XXXXX => num:${SAY:0:2}, digits/thousand, num:${SAY:2}
+    _[n]um:XXX000 => num:${SAY:0:3}, digits/thousand
+    _[n]um:XXXXXX => num:${SAY:0:3}, digits/thousand, num:${SAY:3}
+
+    _[n]um:X000000 => digits/${SAY:0:1}, digits/million
+    _[n]um:XXXXXXX => digits/${SAY:0:1}, digits/million, num:${SAY:1}
+    _[n]um:XX000000 => num:${SAY:0:2}, digits/millions
+    _[n]um:XXXXXXXX => num:${SAY:0:2}, digits/millions, num:${SAY:2}
+    _[n]um:XXX000000 => num:${SAY:0:3}, digits/millions
+    _[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/millions, num:${SAY:3}
+
+    _[n]um:X000000000 => num:${SAY:0:1}, digits/billion
+    _[n]um:XXXXXXXXXX => num:${SAY:0:1}, digits/billion, num:${SAY:1}
+    _[n]um:XX000000000 => num:${SAY:0:2}, digits/billion
+    _[n]um:XXXXXXXXXXX => num:${SAY:0:2}, digits/billion, num:${SAY:2}
+    _[n]um:XXX000000000 => num:${SAY:0:3}, digits/billion
+    _[n]um:XXXXXXXXXXXX => num:${SAY:0:3}, digits/billion, num:${SAY:3}
+
+    _datetime::. => date:AdBY 'digits/at' kM:${SAY}
+    _date::. => date:AdBY:${SAY}
+    _time::. => date:HM:${SAY}
+
+    ; enumeration
+    _e[n]um:X => digits/h-${SAY}
+    _e[n]um:1X => digits/h-${SAY}
+    _e[n]um:[2-9]0 => digits/h-${SAY}
+    _e[n]um:[2-9][1-9] => digits/${SAY:1}-and, digits/h-${SAY:0:1}0
+    _e[n]um:100 => digits/1N, digits/h-hundred
+    _e[n]um:1XX => digits/1N, digits/h-hundred, enum:${SAY:1}
+    _e[n]um:[2-9]00 => num:${SAY:0:1}, digits/h-hundred
+    _e[n]um:[2-9]XX => num:${SAY:0:1}, digits/h-hundred, enum:${SAY:1}

asterisk/sip_notify.conf

@@ -0,0 +1,57 @@
+; rfc3842
+; put empty "Content=>" at the end to have CRLF after last body line
+
+[clear-mwi]
+Event=>message-summary
+Content-type=>application/simple-message-summary
+Content=>Messages-Waiting: no
+Content=>Message-Account: sip:asterisk@127.0.0.1
+Content=>Voice-Message: 0/0 (0/0)
+Content=>
+
+; Aastra
+
+[aastra-check-cfg]
+Event=>check-sync
+
+[aastra-xml]
+Event=>aastra-xml
+
+; Digium
+
+[digium-check-cfg]
+Event=>check-sync
+
+; Linksys
+
+[linksys-cold-restart]
+Event=>reboot_now
+
+[linksys-warm-restart]
+Event=>restart_now
+
+; Polycom
+
+[polycom-check-cfg]
+Event=>check-sync
+
+; Sipura
+
+[sipura-check-cfg]
+Event=>resync
+
+[sipura-get-report]
+Event=>report
+
+; snom
+
+[snom-check-cfg]
+Event=>check-sync\;reboot=false
+
+[snom-reboot]
+Event=>check-sync\;reboot=true
+
+; Cisco
+
+[cisco-check-cfg]
+Event=>check-sync

asterisk/skinny.conf

@@ -0,0 +1,191 @@
+;
+; Skinny Configuration for Asterisk
+;
+[general]
+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.
+keepalive=120
+
+;authtimeout = 30       ; authtimeout specifies the maximum number of seconds a
+			; client has to authenticate.  If the client does not
+			; authenticate beofre this timeout expires, the client
+                        ; will be disconnected.  (default: 30 seconds)
+
+;authlimit = 50         ; authlimit specifies the maximum number of
+			; unauthenticated sessions that will be allowed to
+                        ; connect at any given time. (default: 50)
+
+;vmexten=8500		; Systemwide voicemailmain pilot number
+			; 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
+; 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
+; separated by '&'.  Patterns may be used in regexten.
+;
+;regcontext=skinnyregistrations
+
+;allow=all		; see https://wiki.asterisk.org/wiki/display/AST/RTP+Packetization
+			; for framing options
+;disallow=
+
+; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
+;tos=cs3		; Sets TOS for signaling packets.
+;tos_audio=ef		; Sets TOS for RTP audio packets.
+;tos_video=af41		; Sets TOS for RTP video packets.
+;cos=3			; Sets 802.1p priority for signaling packets.
+;cos_audio=5		; Sets 802.1p priority for RTP audio packets.
+;cos_video=4		; Sets 802.1p priority for RTP video packets.
+
+;------------------------------ 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.
+
+;jbforce = no                ; Forces the use of a jitterbuffer on the receive side of a skinny
+                             ; 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.
+
+;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.
+
+;jblog = no                  ; Enables jitterbuffer frame logging. Defaults to "no".
+;-----------------------------------------------------------------------------------
+
+[lines]
+;----------------------------------- LINES SECTION --------------------------------
+; Options set under [lines] apply to all lines unless explicitly set for a particular
+; device. The options that can be set under lines are specified in GENERAL LINE OPTIONS.
+; These options can also be set for each individual device as well as those under SPECIFIC
+; LINE OPTIONS.
+;
+; Each label below [lines] in [] is a new line with the specific options specified below
+; it. Config stops reading new lines when one of the following is found: [general], [devices]
+; or the end of skinny.conf.
+;
+; Where options are common to both lines and devices, the results typically take that of
+; the least permission. ie if a no is set for either line or device, the call will not be
+; able to use that permission
+;-------------------------------- GENERAL LINE OPTIONS -----------------------------
+;earlyrtp=1                  ; whether audio signalling should be provided by asterisk
+;                            ; (earlyrtp=1) or device generated (earlyrtp=0). default=yes
+;transfer=1                  ; whether the device is allowed to transfer. default=yes
+;context=default             ; context to use for this line.
+;------------------------------- SPECIFIC LINE OPTIONS -----------------------------
+;setvar=        	     ; allows for the setting of chanvars.
+;-----------------------------------------------------------------------------------
+
+;[100]
+;nat=yes
+;callerid="Customer Support" <810-234-1212>
+;mailbox=100
+;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
+;[110]
+;callerid="John Chambers" <408-526-4000>
+;context=did
+;regexten=110
+;linelabel="John"
+;mailbox=110
+
+;[120]
+;Nothing set, so all the defaults are used
+
+;[500]
+;nat=yes
+;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.
+;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.
+;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.
+
+
+[devices]
+;---------------------------------- DEVICES SECTION -------------------------------
+; Options set under [devices] apply to all devices unless explicitly set for a particular
+; device. The options that can be set under devices are specified in GENERAL DEVICE OPTIONS.
+; These options can also be set for each individual device as well as those under SPECIFIC
+; DEVICE OPTIONS.
+;
+; Each label below [devices] in [] is a new device with the specific options specified below
+; it. Config stop reading new devices when one of the following is found: [general], [lines]
+; or the end of skinny.conf.
+;
+; Where options are common to both lines and devices, the results typically take that of
+; the least permission. ie if a no is set for either line or device, the call will not be
+; able to use that permission
+;------------------------------- GENERAL DEVICE OPTIONS ----------------------------
+;earlyrtp=1                  ; whether audio signalling should be provided by asterisk
+;                            ; (earlyrtp=1) or device generated (earlyrtp=0). default=yes
+;transfer=1                  ; whether the device is allowed to transfer. default=yes
+;------------------------------ SPECIFIC DEVICE OPTIONS ----------------------------
+;device="SEPxxxxxxxxxxxx     ; id of the device. Must be set.
+;version=P002G204	     ; firmware version to be loaded. If this version is different
+;                            ; to the one on the device, the device will try to load this
+;                            ; version from the tftp server. Set to device firmware version.
+;-----------------------------------------------------------------------------------
+
+; Typical config for 12SP+
+;[florian]
+;device=SEP00D0BA847E6B
+;version=P002G204	; Thanks critch
+;context=did
+;directmedia=yes	; Allow media to go directly between two RTP endpoints.
+;line=120		; Dial(Skinny/120@florian)
+
+; Typical config for a 7910
+;[duba]			; Device name
+;device=SEP0007EB463101	; Official identifier
+;version=P002F202	; Firmware version identifier
+;host=192.168.1.144
+;permit=192.168.0/24	; Optional, used for authentication
+;line=500
+
+
+; Typical config for a 7940 with dual 7914s
+;[support]
+;device=SEP0007EB463121
+;line=100
+;line=110
+;speeddial => 111,Jack Smith         ; Adds a speeddial button to a device.
+;speeddial => 112@hints,Bob Peterson ; When a context is specified, the speeddial watches a dialplan hint.
+;addon => 7914
+;addon => 7914
+

asterisk/sla.conf

@@ -0,0 +1,151 @@
+;
+; Configuration for Shared Line Appearances (SLA).
+;
+; See http://wiki.asterisk.org or doc/AST.pdf for more information.
+;
+
+; ---- General Options ----------------
+[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.
+; -------------------------------------
+
+
+; ********************************
+; **** Configuration Ordering ****
+; ********************************
+
+; Note that SLA configuration processing assumes that *all* trunk declarations are
+; listed in the configuration file before any stations.
+
+; ********************************
+; ********************************
+
+
+; ---- Trunk Declarations -------------
+;
+;[line1]                    ; Provide a name for this trunk.
+
+;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 
+                            ; https://wiki.asterisk.org/wiki/display/AST/SLA+Trunk+Configuration
+
+;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.
+
+;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".
+
+;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.
+
+;[line2]
+;type=trunk
+;device=DAHDI/4
+;autocontext=line2
+
+;[line3]
+;type=trunk
+;device=DAHDI/3
+;autocontext=line3
+
+;[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 https://wiki.asterisk.org/wiki/display/AST/SLA+Trunk+Configuration
+;autocontext=line4
+; --------------------------------------
+
+
+; ---- Station Declarations ------------
+
+;[station1]                 ; Define a name for this station.
+
+;type=station               ; This line indicates that this entry is a station.
+
+;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.
+
+;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.
+
+;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.
+
+
+;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.
+;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.
+
+;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.
+
+
+;[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.
+;type=station
+;autocontext=sla_stations
+;trunk=line1
+;trunk=line2
+;trunk=line3
+;trunk=line4
+
+;[station2](station)        ; Define a station that uses the configuration from the template "station".
+;device=SIP/station2
+;
+;[station3](station)
+;device=SIP/station3
+;
+;[station4](station)
+;device=SIP/station4
+;
+;[station5](station)
+;device=SIP/station5
+; --------------------------------------
+

asterisk/smdi.conf

@@ -0,0 +1,75 @@
+; Asterisk SMDI configuration
+
+[interfaces]
+; Specify serial ports to listen for SMDI messages on below.  These will be
+; referenced later in chan_dahdi.conf.  If you do not specify any interfaces
+; then SMDI will be disabled.  Interfaces can have several different attributes
+; associated with them.
+
+; Set the number of stop bits to use per character here.  The default is no,
+; in which case one stop bit will be used.
+
+;twostopbits = no
+
+; Character size or bit length is the size of each character sent across the
+; link.  Character size can be 7 or 8.  The default is 7.
+
+;charsize = 7
+
+; If you need parity checking enabled you can turn it on here.  Acceptable
+; values are even, odd, and none.  The default is even.
+
+;paritybit = even
+
+; The baudrate to use for this port.  Acceptable values are 1200, 2400, 4800,
+; and 9600.  The default is 9600.
+
+;baudrate = 1200
+
+; Often the numbering scheme for a set of mailboxes or extensions will not be 7
+; or 10 digits (as SMDI requires).  Use the msdstrip option to strip unused
+; digits from the start of numbers.
+
+;msdstrip = 0
+
+; Occasionally Asterisk and the SMDI switch may become out of sync.  If this
+; happens, Asterisk will appear one or several calls behind as it processes
+; voicemail requests.  To prevent this from happening, adjust the msgexpirytime.
+; This will make Asterisk discard old SMDI messages that have not yet been
+; processed.  The default expiry time is 30000 milliseconds.
+
+;msgexpirytime = 30000
+
+;smdiport => /dev/ttyS0
+
+
+[mailboxes]
+; This section configures parameters related to MWI handling for the SMDI link.
+
+; This option configures the polling interval used to check to see if the
+; mailboxes have any new messages.  This option is specified in seconds.
+; The default value is 10 seconds.
+;
+;pollinginterval=10
+
+; Every other entry in this section of the configuration file is interpreted as
+; a mapping between the mailbox ID on the SMDI link, and the local Asterisk
+; mailbox name.  In many cases, they are the same thing, but they still must be
+; listed here so that this module knows which mailboxes it needs to pay
+; attention to.
+;
+; Syntax:
+;   <SMDI mailbox ID>=<Asterisk Mailbox Name>[@Asterisk Voicemail Context]
+;
+; If no Asterisk voicemail context is specified, "default" will be assumed.
+;
+; Before specifying mailboxes, you must specify an SMDI interface.  All mailbox
+; definitions that follow will correspond to that SMDI interface.  If you specify
+; another interface, then all definitions following that will correspond to the
+; new interface.
+;
+;smdiport=/dev/ttyS0
+;2565551234=1234@vmcontext1
+;2565555678=5678@vmcontext2
+;smdiport=/dev/ttyS1
+;2565559999=9999

asterisk/telcordia-1.adsi

@@ -0,0 +1,83 @@
+;
+; Asterisk default ADSI script
+;
+;
+; Begin with the preamble requirements
+;
+DESCRIPTION "Telcordia Demo"		; Name of vendor
+VERSION 0x02				; Version of stuff
+;SECURITY "_AST"				; Security code
+SECURITY 0x0000				; Security code
+FDN 0x0000000f				; Descriptor number
+
+;
+; Predefined strings
+;
+DISPLAY "talkingto" IS "Talking To" "$Call1p" WRAP
+DISPLAY "titles" IS "20th Century IQ Svc"
+DISPLAY "newcall" IS "New Call From" "$Call1p" WRAP
+DISPLAY "ringing" IS "Ringing"
+
+;
+; Begin state definitions
+;
+STATE "callup"		; Call is currently up
+STATE "inactive"	; No active call
+
+;
+; Begin soft key definitions
+;
+KEY "CB_OH" IS "Block" OR "Call Block"
+	OFFHOOK
+	VOICEMODE
+	WAITDIALTONE
+	SENDDTMF "*60"
+	SUBSCRIPT "offHook"
+ENDKEY
+
+KEY "CB" IS "Block" OR "Call Block"
+	SENDDTMF "*60"
+ENDKEY
+
+;
+; Begin main subroutine
+;
+
+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
+ENDSUB
+
+SUB "offHook" IS
+	IFEVENT FARRING THEN
+		CLEAR
+		SHOWDISPLAY "ringing" AT 1
+	ENDIF
+	IFEVENT FARANSWER THEN
+		CLEAR
+		SHOWDISPLAY "talkingto" AT 1
+		GOTO "stableCall"
+	ENDIF
+ENDSUB
+
+SUB "stableCall" IS
+
+ENDSUB

asterisk/udptl.conf

@@ -0,0 +1,26 @@
+;
+; UDPTL Configuration (UDPTL is one of the transports for T.38)
+;
+[general]
+;
+; UDPTL start and UDPTL end configure start and end addresses
+;
+udptlstart=4000
+udptlend=4999
+;
+; Whether to enable or disable UDP checksums on UDPTL traffic
+;
+;udptlchecksums=no
+;
+; The number of error correction entries in a UDPTL packet
+;
+udptlfecentries = 3
+;
+; The span over which parity is calculated for FEC in a UDPTL packet
+;
+udptlfecspan = 3
+;
+; Some VoIP providers will only accept an offer with an even-numbered
+; UDPTL port. Set this option so that Asterisk will only attempt to use
+; even-numbered ports when negotiating T.38. Default is no.
+use_even_ports = no

asterisk/unistim.conf

@@ -0,0 +1,86 @@
+;
+; chan_unistim configuration file.
+;
+
+[general]
+port=5000                    ; UDP port
+;
+; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
+;tos=cs3                ; Sets TOS for signaling packets.
+;tos_audio=ef           ; Sets TOS for RTP audio packets.
+;cos=3                  ; Sets 802.1p priority for signaling packets.
+;cos_audio=5            ; Sets 802.1p priority for RTP audio packets.
+;
+;debug=yes              ; Enable debug (default no)
+;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.
+;mohsuggest=default
+;------------------------------ 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.
+
+; jbforce = no                ; Forces the use of a jitterbuffer on the receive side of a SIP
+                              ; 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.
+
+; 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.
+
+; jblog = no                  ; Enables jitterbuffer frame logging. Defaults to "no".
+;-----------------------------------------------------------------------------------
+
+
+;[black]                     ; name of the device
+;device=000ae4012345         ; mac address of the phone
+;rtp_port=10000              ; RTP port used by the phone, default = 10000. RTCP = rtp_port+1
+;rtp_method=0                ; If you don't have sound, you can try 1, 2 or 3, default = 0
+                             ; value 3 works on newer i2004, 1120E and 1140E
+;status_method=0             ; If you don't see status text, try 1, default = 0
+                             ; value 1 works on 1120E and 1140E
+;titledefault=Asterisk       ; default = "TimeZone (your time zone)". 12 characters max
+;height=3                    ; default = 3, the number of display lines the device can show
+                             ; For example on a Nortel I2001 or I2002, set this to 1
+;maintext0="you can insert"  ; default = "Welcome", 24 characters max
+;maintext1="a custom text"   ; default = the name of the device, 24 characters max
+;maintext2="(main page)"     ; default = the public IP of the phone, 24 characters max
+;dateformat=1                ; 0 = month/day, 1 (default) = day/month
+;timeformat=1                ; 0 = 0:00am ; 1 (default) = 0h00, 2 = 0:00
+;contrast=8                  ; define the contrast of the LCD. From 0 to 15. Default = 8
+;country=us                  ; country (ccTLD) for dial tone frequency. See README, default = us
+;ringvolume=2                ; ring volume : 0,1,2,3, can be overrided by Dial(), default = 2
+;ringstyle=3                 ; ring style : 0 to 7, can be overrided by Dial(), default = 3
+;cwvolume=2                  ; ring volume : 0,1,2,3, default = 0
+;cwstyle=3                   ; ring style : 0 to 7, default = 2
+;sharpdial=1                 ; dial number by pressing #, default = 0
+;interdigit_timer=4000       ; timer for automatic dial after several digits of number entered (in ms, 0 is off)
+;callhistory=1               ; 0 = disable, 1 = enable call history, default = 1
+;callerid="Customer Support" <555-234-5678>
+;context=default             ; context, default="default"
+;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
+;line => 100                 ; Any number of lines can be defined in any order with bookmarks
+;line => 200                 ; After line defined it placed in next available slot
+;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
+
+;[violet]
+;device=006038abcdef
+;line => 102

asterisk/users.conf

@@ -0,0 +1,114 @@
+;
+; 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
+; 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
+; 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.
+;
+
+[general]
+;
+; Full name of a user
+;
+fullname = New User
+;
+; Starting point of allocation of extensions
+;
+userbase = 6000
+;
+; Create voicemail mailbox and use use macro-stdexten
+;
+hasvoicemail = yes
+;
+; Set voicemail mailbox 6000 password to 1234
+;
+vmsecret = 1234
+;
+; Create SIP Peer
+;
+hassip = yes
+;
+; Create IAX friend
+;
+hasiax = yes
+;
+; Create H.323 friend
+;
+;hash323 = yes
+;
+; Create manager entry
+;
+hasmanager = no
+;
+; Set permissions for manager entry (see manager.conf.sample for documentation)
+; (defaults to *all* permissions)
+;managerread = system,call,log,verbose,command,agent,user,config
+;managerwrite = system,call,log,verbose,command,agent,user,config
+;
+;
+; MAC Address for res_phoneprov
+;
+;macaddress = 112233445566
+;
+; Auto provision the phone with res_phoneprov
+;
+;autoprov = yes
+;
+; Line Keys for hardphone
+;
+;LINEKEYS = 1
+;
+; Line number for hardphone
+;
+;linenumber = 1
+;
+; Local Caller ID number used with res_phoneprov and Asterisk GUI
+;
+;cid_number = 6000
+;
+; Remaining options are not specific to users.conf entries but are general.
+;
+callwaiting = yes
+threewaycalling = yes
+callwaitingcallerid = yes
+transfer = yes
+canpark = yes
+cancallforward = yes
+callreturn = yes
+callgroup = 1
+pickupgroup = 1
+;nat = no
+
+;[6000]
+;fullname = Joe User
+;description = Courtesy Phone In Lobby    ; Used to provide a description of the
+                                          ; peer in console output
+;email = joe@foo.bar
+;secret = 1234
+;dahdichan = 1
+;hasvoicemail = yes
+;vmsecret = 1234
+;hassip = yes
+;hasiax = no
+;hash323 = no
+;hasmanager = no
+;callwaiting = no
+;context = international
+;
+; Some administrators choose alphanumeric extensions, but still want their
+; users to be reachable by traditional numeric extensions, specified by the
+; alternateexts entry.
+;
+;alternateexts = 7057,3249
+;macaddress = 112233445566
+;autoprov = yes
+;LINEKEYS = 1
+;linenumber = 1
+;cid_number = 6000
+