configs: Move sample config files into a subdirectory of configs

This moves all samples configs from configs/ to configs/samples. This allows
for additional sets of sample configuration files to be added in the future.

Review: https://reviewboard.asterisk.org/r/3804/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418870 65c4cc65-6c06-0410-ace0-fbb531ad65f3

107 files changed, 16768 insertions, 0 deletions

diff --git a/configs/samples/acl.conf.sample b/configs/samples/acl.conf.sample
new file mode 100644
index 000000000..b052606ce
--- /dev/null
+++ b/configs/samples/acl.conf.sample
@@ -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
diff --git a/configs/samples/adsi.conf.sample b/configs/samples/adsi.conf.sample
new file mode 100644
index 000000000..0f36f80da
--- /dev/null
+++ b/configs/samples/adsi.conf.sample
@@ -0,0 +1,8 @@
+;
+; Sample ADSI Configuration file
+;
+[intro]
+alignment = center
+greeting => Welcome to the
+greeting => Asterisk
+greeting => Open Source PBX
diff --git a/configs/samples/agents.conf.sample b/configs/samples/agents.conf.sample
new file mode 100644
index 000000000..0cf0c4c90
--- /dev/null
+++ b/configs/samples/agents.conf.sample
@@ -0,0 +1,70 @@
+;
+; Agent pool configuration
+;
+
+[general]
+; The general section of this config is not currently used, but reserved
+; for future use.
+
+;[agent-id]
+; Define ackcall to require the agent to give a DTMF acknowledgement
+; when the agent receives a call.
+; The channel variable AGENTACKCALL overrides on agent login.
+; Default is "no".
+;ackcall=no
+;
+; Set what DTMF key sequence the agent should use to acknowledge a call.
+; The channel variable AGENTACCEPTDTMF overrides on agent login.
+; This option is ignored unless ackcall is enabled.
+; Default is "#".
+;acceptdtmf=##
+;
+; Set how many seconds a call for the agent has to wait for the agent to
+; acknowledge the call before the agent is automatically logged off.  If
+; set to zero then the call will wait forever for the agent to acknowledge.
+; The channel variable AGENTAUTOLOGOFF overrides on agent login.
+; This option is ignored unless ackcall is enabled.
+; Default is 0.
+;autologoff=15
+;
+; Set the minimum amount of time after disconnecting a call before
+; the agent can receive a new call in milliseconds.
+; The channel variable AGENTWRAPUPTIME overrides on agent login.
+; Default is 0.
+;wrapuptime=5000
+;
+; Set the musiconhold class for the agent.
+; Default is "default".
+;musiconhold=default
+;
+; Enable recording calls the agent takes automatically by invoking the
+; DTMF automixmon feature when the agent connects to a caller.
+; See features.conf.sample for information about the automixmon feature.
+; Default is "no".
+;recordagentcalls=yes
+;
+; The sound file played to alert the agent when a call is present.
+; Default is "beep".
+;custom_beep=beep
+;
+; A friendly name for the agent used in log messages.
+; Default is "".
+;fullname=Mark Spencer
+;
+; --------------------------------------------------
+;
+; This section contains example agent definitions:
+;
+; Define a template called my-agents:
+;[my-agents](!)
+;autologoff=15
+;ackcall=yes
+;acceptdtmf=##
+;
+; Define agent 1001 using the my-agents template:
+;[1001](my-agents)
+;fullname=Mark Spencer
+;
+; Define agent 1002 using the my-agents template:
+;[1002](my-agents)
+;fullname=Will Meadows
diff --git a/configs/samples/alarmreceiver.conf.sample b/configs/samples/alarmreceiver.conf.sample
new file mode 100644
index 000000000..e4815a9eb
--- /dev/null
+++ b/configs/samples/alarmreceiver.conf.sample
@@ -0,0 +1,91 @@
+;
+; 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
+
+;
+; Wait for the connection to settle post-answer. Adjustable from 500 msec. to 10000 msec.
+; The default is 1250 msec.
+;
+
+answait = 1250
+
+; When logging individual events it may be desirable to skip grouping of metadata
+
+;no_group_meta = yes
+
+;
+; The loudness of the ACK and Kissoff tones is adjustable from 100 to 8192.
+; The default is 8192. This shouldn't need to be messed with, but is included
+; just in case there are problems with signal levels.
+;
+
+loudness = 8192
+
+;
+; The db-family setting allows the user to capture statistics on the number of
+; calls, and the errors the alarm receiver sees. The default is for no
+; db-family name to be defined and the database logging to be turned off.
+;
+
+;db-family = yourfamily:
+
+;
+; End of alarmreceiver.conf
+;
diff --git a/configs/samples/alsa.conf.sample b/configs/samples/alsa.conf.sample
new file mode 100644
index 000000000..ced5b4485
--- /dev/null
+++ b/configs/samples/alsa.conf.sample
@@ -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".
+;-----------------------------------------------------------------------------------
+
diff --git a/configs/samples/amd.conf.sample b/configs/samples/amd.conf.sample
new file mode 100644
index 000000000..ce4808a0c
--- /dev/null
+++ b/configs/samples/amd.conf.sample
@@ -0,0 +1,18 @@
+;
+; Answering Machine Detection Configuration
+;
+
+[general]
+initial_silence = 2500		; Maximum silence duration before the greeting.
+				; If exceeded then MACHINE.
+greeting = 1500			; Maximum length of a greeting. If exceeded then MACHINE.
+after_greeting_silence = 800	; Silence after detecting a greeting.
+				; If exceeded then HUMAN
+total_analysis_time = 5000	; Maximum time allowed for the algorithm to decide
+				; on a HUMAN or MACHINE
+min_word_length = 100		; Minimum duration of Voice to considered as a word
+between_words_silence = 50	; Minimum duration of silence after a word to consider
+				; the audio what follows as a new word
+maximum_number_of_words = 3	; Maximum number of words in the greeting.
+				; If exceeded then MACHINE
+silence_threshold = 256
diff --git a/configs/samples/app_mysql.conf.sample b/configs/samples/app_mysql.conf.sample
new file mode 100644
index 000000000..fafd4f7a2
--- /dev/null
+++ b/configs/samples/app_mysql.conf.sample
@@ -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
diff --git a/configs/samples/app_skel.conf.sample b/configs/samples/app_skel.conf.sample
new file mode 100644
index 000000000..ada8461e3
--- /dev/null
+++ b/configs/samples/app_skel.conf.sample
@@ -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
diff --git a/configs/samples/ari.conf.sample b/configs/samples/ari.conf.sample
new file mode 100644
index 000000000..59f9a44e5
--- /dev/null
+++ b/configs/samples/ari.conf.sample
@@ -0,0 +1,31 @@
+[general]
+enabled = yes       ; When set to no, ARI support is disabled.
+;pretty = no        ; When set to yes, responses from ARI are
+;                   ; formatted to be human readable.
+;allowed_origins =  ; Comma separated list of allowed origins, for
+;                   ; Cross-Origin Resource Sharing. May be set to * to
+;                   ; allow all origins.
+;auth_realm =       ; Realm to use for authentication. Defaults to Asterisk
+;                   ; REST Interface.
+;
+; Default write timeout to set on websockets. This value may need to be adjusted
+; for connections where Asterisk must write a substantial amount of data and the
+; receiving clients are slow to process the received information. Value is in
+; milliseconds; default is 100 ms.
+;websocket_write_timeout = 100
+
+;[username]
+;type = user        ; Specifies user configuration
+;read_only = no     ; When set to yes, user is only authorized for
+;                   ; read-only requests.
+;
+;password =         ; Crypted or plaintext password (see password_format).
+;
+; password_format may be set to plain (the default) or crypt. When set to crypt,
+; crypt(3) is used to validate the password. A crypted password can be generated
+; using mkpasswd -m sha-512.
+;
+; When set to plain, the password is in plaintext.
+;
+;password_format = plain
+
diff --git a/configs/samples/asterisk.adsi b/configs/samples/asterisk.adsi
new file mode 100644
index 000000000..a58952589
--- /dev/null
+++ b/configs/samples/asterisk.adsi
@@ -0,0 +1,159 @@
+;
+; Asterisk default ADSI script
+;
+;
+; Begin with the preamble requirements
+;
+DESCRIPTION "Asterisk PBX"		; Name of vendor
+VERSION 0x00						; Version of stuff
+;SECURITY "_AST"					; Security code
+SECURITY 0X9BDBF7AC				; Security code
+FDN 0x0000000F						; Descriptor number
+
+;
+; Flags
+;
+FLAG "nocallwaiting"
+
+;
+; Predefined strings
+;
+DISPLAY "titles" IS "** Asterisk PBX **"
+DISPLAY "talkingto" IS "Call active." JUSTIFY LEFT
+DISPLAY "callname" IS "$Call1p" JUSTIFY LEFT
+DISPLAY "callnum" IS "$Call1s" JUSTIFY LEFT
+DISPLAY "incoming" IS "Incoming call!" JUSTIFY LEFT
+DISPLAY "ringing" IS "Calling... " JUSTIFY LEFT
+DISPLAY "callended" IS "Call ended." JUSTIFY LEFT
+DISPLAY "missedcall" IS "Missed call." JUSTIFY LEFT
+DISPLAY "busy" IS "Busy." JUSTIFY LEFT
+DISPLAY "reorder" IS "Reorder." JUSTIFY LEFT
+DISPLAY "cwdisabled" IS "Callwait disabled"
+DISPLAY "empty" IS "asdf"
+
+;
+; Begin soft key definitions
+;
+KEY "callfwd" IS "CallFwd" OR "Call Forward"
+	OFFHOOK
+	VOICEMODE
+	WAITDIALTONE
+	SENDDTMF "*60"
+	GOTO "offHook"
+ENDKEY
+
+KEY "vmail_OH" IS "VMail" OR "Voicemail"
+	OFFHOOK
+	VOICEMODE
+	WAITDIALTONE
+	SENDDTMF "8500"
+ENDKEY
+
+KEY "vmail" IS "VMail" OR "Voicemail"
+	SENDDTMF "8500"
+ENDKEY
+
+KEY "backspace" IS "BackSpc" OR "Backspace"
+	BACKSPACE
+ENDKEY
+
+KEY "cwdisable" IS "CWDsble" OR "Disable Call Wait"
+	SENDDTMF "*70"
+	SETFLAG "nocallwaiting"
+	SHOWDISPLAY "cwdisabled" AT 4
+	TIMERCLEAR
+	TIMERSTART 1
+ENDKEY
+
+KEY "cidblock" IS "CIDBlk" OR "Block Callerid"
+	SENDDTMF "*67"
+	SETFLAG "nocallwaiting"
+ENDKEY
+
+;
+; Begin main subroutine
+;
+
+SUB "main" IS
+	IFEVENT NEARANSWER THEN
+		CLEAR
+		SHOWDISPLAY "titles" AT 1 NOUPDATE
+		SHOWDISPLAY "talkingto" AT 2 NOUPDATE
+		SHOWDISPLAY "callname" AT 3
+		SHOWDISPLAY "callnum" AT 4
+		GOTO "stableCall"
+	ENDIF
+	IFEVENT OFFHOOK THEN
+		CLEAR
+		CLEARFLAG "nocallwaiting"
+		CLEARDISPLAY
+		SHOWDISPLAY "titles" AT 1
+		SHOWKEYS "vmail"
+		SHOWKEYS "cidblock"
+		SHOWKEYS "cwdisable" UNLESS "nocallwaiting"
+		GOTO "offHook"
+	ENDIF
+	IFEVENT IDLE THEN
+		CLEAR
+		SHOWDISPLAY "titles" AT 1
+		SHOWKEYS "vmail_OH"
+	ENDIF
+	IFEVENT CALLERID THEN
+		CLEAR
+;		SHOWDISPLAY "titles" AT 1 NOUPDATE
+;		SHOWDISPLAY "incoming" AT 2 NOUPDATE
+		SHOWDISPLAY "callname" AT 3 NOUPDATE
+		SHOWDISPLAY "callnum" AT 4
+	ENDIF
+	IFEVENT RING THEN
+		CLEAR
+		SHOWDISPLAY "titles" AT 1 NOUPDATE
+		SHOWDISPLAY "incoming" AT 2
+	ENDIF
+	IFEVENT ENDOFRING THEN
+		SHOWDISPLAY "missedcall" AT 2
+		CLEAR
+		SHOWDISPLAY "titles" AT 1
+		SHOWKEYS "vmail_OH"
+	ENDIF
+	IFEVENT TIMER THEN
+		CLEAR
+		SHOWDISPLAY "empty" AT 4
+	ENDIF
+ENDSUB
+
+SUB "offHook" IS
+	IFEVENT FARRING THEN
+		CLEAR
+		SHOWDISPLAY "titles" AT 1 NOUPDATE
+		SHOWDISPLAY "ringing" AT 2 NOUPDATE
+		SHOWDISPLAY "callname" at 3 NOUPDATE
+		SHOWDISPLAY "callnum" at 4
+	ENDIF
+	IFEVENT FARANSWER THEN
+		CLEAR
+		SHOWDISPLAY "talkingto" AT 2
+		GOTO "stableCall"
+	ENDIF
+	IFEVENT BUSY THEN
+		CLEAR
+		SHOWDISPLAY "titles" AT 1 NOUPDATE
+		SHOWDISPLAY "busy" AT 2 NOUPDATE
+		SHOWDISPLAY "callname" at 3 NOUPDATE
+		SHOWDISPLAY "callnum" at 4
+	ENDIF
+	IFEVENT REORDER THEN
+		CLEAR
+		SHOWDISPLAY "titles" AT 1 NOUPDATE
+		SHOWDISPLAY "reorder" AT 2 NOUPDATE
+		SHOWDISPLAY "callname" at 3 NOUPDATE
+		SHOWDISPLAY "callnum" at 4
+	ENDIF
+ENDSUB
+
+SUB "stableCall" IS
+	IFEVENT REORDER THEN
+		SHOWDISPLAY "callended" AT 2
+	ENDIF
+ENDSUB
+
diff --git a/configs/samples/asterisk.conf.sample b/configs/samples/asterisk.conf.sample
new file mode 100644
index 000000000..e4883ec97
--- /dev/null
+++ b/configs/samples/asterisk.conf.sample
@@ -0,0 +1,97 @@
+[directories](!)
+astetcdir => /etc/asterisk
+astmoddir => /usr/lib/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 no
+
+; Changing the following lines may compromise your security.
+;[files]
+;astctlpermissions = 0660
+;astctlowner = root
+;astctlgroup = apache
+;astctl = asterisk.ctl
diff --git a/configs/samples/calendar.conf.sample b/configs/samples/calendar.conf.sample
new file mode 100644
index 000000000..82b8702f0
--- /dev/null
+++ b/configs/samples/calendar.conf.sample
@@ -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
diff --git a/configs/samples/ccss.conf.sample b/configs/samples/ccss.conf.sample
new file mode 100644
index 000000000..21b0b0668
--- /dev/null
+++ b/configs/samples/ccss.conf.sample
@@ -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=
diff --git a/configs/samples/cdr.conf.sample b/configs/samples/cdr.conf.sample
new file mode 100644
index 000000000..458e19ab4
--- /dev/null
+++ b/configs/samples/cdr.conf.sample
@@ -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
diff --git a/configs/samples/cdr_adaptive_odbc.conf.sample b/configs/samples/cdr_adaptive_odbc.conf.sample
new file mode 100644
index 000000000..f3c806e75
--- /dev/null
+++ b/configs/samples/cdr_adaptive_odbc.conf.sample
@@ -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
+
diff --git a/configs/samples/cdr_custom.conf.sample b/configs/samples/cdr_custom.conf.sample
new file mode 100644
index 000000000..bec9e3377
--- /dev/null
+++ b/configs/samples/cdr_custom.conf.sample
@@ -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)})}
diff --git a/configs/samples/cdr_manager.conf.sample b/configs/samples/cdr_manager.conf.sample
new file mode 100644
index 000000000..aa596b718
--- /dev/null
+++ b/configs/samples/cdr_manager.conf.sample
@@ -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
diff --git a/configs/samples/cdr_mysql.conf.sample b/configs/samples/cdr_mysql.conf.sample
new file mode 100644
index 000000000..04b70493b
--- /dev/null
+++ b/configs/samples/cdr_mysql.conf.sample
@@ -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>
+
diff --git a/configs/samples/cdr_odbc.conf.sample b/configs/samples/cdr_odbc.conf.sample
new file mode 100644
index 000000000..93bd6fff8
--- /dev/null
+++ b/configs/samples/cdr_odbc.conf.sample
@@ -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
diff --git a/configs/samples/cdr_pgsql.conf.sample b/configs/samples/cdr_pgsql.conf.sample
new file mode 100644
index 000000000..c5a989f31
--- /dev/null
+++ b/configs/samples/cdr_pgsql.conf.sample
@@ -0,0 +1,16 @@
+; 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
+;appname=asterisk    ; Postgres application_name support (optional). Whitespace not allowed.
+;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
diff --git a/configs/samples/cdr_sqlite3_custom.conf.sample b/configs/samples/cdr_sqlite3_custom.conf.sample
new file mode 100644
index 000000000..0d5dc091c
--- /dev/null
+++ b/configs/samples/cdr_sqlite3_custom.conf.sample
@@ -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)}'
diff --git a/configs/samples/cdr_syslog.conf.sample b/configs/samples/cdr_syslog.conf.sample
new file mode 100644
index 000000000..3a619be9f
--- /dev/null
+++ b/configs/samples/cdr_syslog.conf.sample
@@ -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)}"
diff --git a/configs/samples/cdr_tds.conf.sample b/configs/samples/cdr_tds.conf.sample
new file mode 100644
index 000000000..309599e5f
--- /dev/null
+++ b/configs/samples/cdr_tds.conf.sample
@@ -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
+
diff --git a/configs/samples/cel.conf.sample b/configs/samples/cel.conf.sample
new file mode 100644
index 000000000..d92e2745d
--- /dev/null
+++ b/configs/samples/cel.conf.sample
@@ -0,0 +1,116 @@
+;
+; 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
+;  BRIDGE_ENTER       -- The time a channel was connected into a conference room
+;  BRIDGE_EXIT        -- The time a channel was removed from a conference room
+;  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
+;  BLINDTRANSFER    -- When a blind transfer is initiated
+;  ATTENDEDTRANSFER -- When an attended transfer is initiated
+;  PICKUP           -- This channel picked up the specified channel
+;  FORWARD          -- This channel is being forwarded somewhere else
+;  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
+;  LOCAL_OPTIMIZE   -- A local channel pair is optimizing away.
+;
+; Default value: none
+;                (Track no events)
+
+events=APP_START,CHAN_START,CHAN_END,ANSWER,HANGUP,BRIDGE_ENTER,BRIDGE_EXIT
+
+; 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
+;
diff --git a/configs/samples/cel_custom.conf.sample b/configs/samples/cel_custom.conf.sample
new file mode 100644
index 000000000..126248aca
--- /dev/null
+++ b/configs/samples/cel_custom.conf.sample
@@ -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})}
diff --git a/configs/samples/cel_odbc.conf.sample b/configs/samples/cel_odbc.conf.sample
new file mode 100644
index 000000000..0c0b83fda
--- /dev/null
+++ b/configs/samples/cel_odbc.conf.sample
@@ -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
diff --git a/configs/samples/cel_pgsql.conf.sample b/configs/samples/cel_pgsql.conf.sample
new file mode 100644
index 000000000..2298b2ddc
--- /dev/null
+++ b/configs/samples/cel_pgsql.conf.sample
@@ -0,0 +1,68 @@
+;
+; 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
+;appname=asterisk   ; Postgres application_name support (optional). Whitespace not allowed.
\ No newline at end of file
diff --git a/configs/samples/cel_sqlite3_custom.conf.sample b/configs/samples/cel_sqlite3_custom.conf.sample
new file mode 100644
index 000000000..2d9a24fc7
--- /dev/null
+++ b/configs/samples/cel_sqlite3_custom.conf.sample
@@ -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}'
diff --git a/configs/samples/cel_tds.conf.sample b/configs/samples/cel_tds.conf.sample
new file mode 100644
index 000000000..399093b47
--- /dev/null
+++ b/configs/samples/cel_tds.conf.sample
@@ -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
diff --git a/configs/samples/chan_dahdi.conf.sample b/configs/samples/chan_dahdi.conf.sample
new file mode 100644
index 000000000..13691fcdf
--- /dev/null
+++ b/configs/samples/chan_dahdi.conf.sample
@@ -0,0 +1,1695 @@
+;
+; DAHDI Telephony Configuration file
+;
+; You need to restart Asterisk to re-configure the DAHDI channel
+; CLI> module reload chan_dahdi.so
+;      will reload the configuration file, but not all configuration options
+;      are re-configured during a reload (signalling, as well as PRI and
+;      SS7-related settings cannot be changed on a reload).
+;
+; This file documents many configuration variables.  Normally unless you know
+; what a variable means or that it should be changed, there's no reason to
+; un-comment those lines.
+;
+; Examples below that are commented out (those lines that begin with a ';' but
+; no space afterwards) typically show a value that is not the default value,
+; but would make sense under certain circumstances. The default values are
+; usually sane. Thus you should typically not touch them unless you know what
+; they mean or you know you should change them.
+
+[trunkgroups]
+;
+; Trunk groups are used for NFAS connections.
+;
+; Group: Defines a trunk group.
+;        trunkgroup => <trunkgroup>,<dchannel>[,<backup1>...]
+;
+;        trunkgroup  is the numerical trunk group to create
+;        dchannel    is the DAHDI channel which will have the
+;                    d-channel for the trunk.
+;        backup1     is an optional list of backup d-channels.
+;
+;trunkgroup => 1,24,48
+;trunkgroup => 1,24
+;
+; Spanmap: Associates a span with a trunk group
+;        spanmap => <dahdispan>,<trunkgroup>[,<logicalspan>]
+;
+;        dahdispan   is the DAHDI span number to associate
+;        trunkgroup  is the trunkgroup (specified above) for the mapping
+;        logicalspan is the logical span number within the trunk group to use.
+;                    if unspecified, no logical span number is used.
+;
+;spanmap => 1,1,1
+;spanmap => 2,1,2
+;spanmap => 3,1,3
+;spanmap => 4,1,4
+
+[channels]
+;
+; Default language
+;
+;language=en
+;
+; Context for incoming calls. Defaults to 'default'
+;
+context=public
+;
+; Switchtype:  Only used for PRI.
+;
+; national:    National ISDN 2 (default)
+; dms100:      Nortel DMS100
+; 4ess:        AT&T 4ESS
+; 5ess:        Lucent 5ESS
+; euroisdn:    EuroISDN (common in Europe)
+; ni1:         Old National ISDN 1
+; qsig:        Q.SIG
+;
+;switchtype=euroisdn
+;
+; MSNs for ISDN spans.  Asterisk will listen for the listed numbers on
+; incoming calls and ignore any calls not listed.
+; Here you can give a comma separated list of numbers or dialplan extension
+; patterns.  An empty list disables MSN matching to allow any incoming call.
+; Only set on PTMP CPE side of ISDN span if needed.
+; The default is an empty list.
+;msn=
+;
+; Some switches (AT&T especially) require network specific facility IE.
+; Supported values are currently 'none', 'sdn', 'megacom', 'tollfreemegacom', 'accunet'
+;
+; nsf cannot be changed on a reload.
+;
+;nsf=none
+;
+;service_message_support=yes
+; Enable service message support for channel. Must be set after switchtype.
+;
+; Dialing options for ISDN (i.e., Dial(DAHDI/g1/exten/options)):
+; R      Reverse Charge Indication
+;          Indicate to the called party that the call will be reverse charged.
+; K(n)   Keypad digits n
+;          Send out the specified digits as keypad digits.
+;
+; PRI Dialplan: The ISDN-level Type Of Number (TON) or numbering plan, used for
+; the dialed number.  Leaving this as 'unknown' (the default) works for most
+; cases.  In some very unusual circumstances, you may need to set this to
+; 'dynamic' or 'redundant'.
+;
+; unknown:        Unknown
+; private:        Private ISDN
+; local:          Local ISDN
+; national:       National ISDN
+; international:  International ISDN
+; dynamic:        Dynamically selects the appropriate dialplan using the
+;                 prefix settings.
+; redundant:      Same as dynamic, except that the underlying number is not
+;                 changed (not common)
+;
+; pridialplan cannot be changed on reload.
+;pridialplan=unknown
+;
+; PRI Local Dialplan:  Only RARELY used for PRI (sets the calling number's
+; numbering plan).  In North America, the typical use is sending the 10 digit
+; callerID number and setting the prilocaldialplan to 'national' (the default).
+; Only VERY rarely will you need to change this.
+;
+; unknown:        Unknown
+; private:        Private ISDN
+; local:          Local ISDN
+; national:       National ISDN
+; international:  International ISDN
+; from_channel:   Use the CALLERID(ton) value from the channel.
+; dynamic:        Dynamically selects the appropriate dialplan using the
+;                 prefix settings.
+; redundant:      Same as dynamic, except that the underlying number is not
+;                 changed (not common)
+;
+; prilocaldialplan cannot be changed on reload.
+;prilocaldialplan=national
+;
+; PRI Connected Line Dialplan:  Sets the connected party number's numbering plan.
+;
+; unknown:        Unknown
+; private:        Private ISDN
+; local:          Local ISDN
+; national:       National ISDN
+; international:  International ISDN
+; from_channel:   Use the CONNECTEDLINE(ton) value from the channel.
+; dynamic:        Dynamically selects the appropriate dialplan using the
+;                 prefix settings.
+; redundant:      Same as dynamic, except that the underlying number is not
+;                 changed (not common)
+;
+; pricpndialplan cannot be changed on reload.
+;pricpndialplan=from_channel
+;
+; pridialplan may be also set at dialtime, by prefixing the dialed number with
+; one of the following letters:
+; U - Unknown
+; I - International
+; N - National
+; L - Local (Net Specific)
+; S - Subscriber
+; V - Abbreviated
+; R - Reserved (should probably never be used but is included for completeness)
+;
+; Additionally, you may also set the following NPI bits (also by prefixing the
+; dialed string with one of the following letters):
+; u - Unknown
+; e - E.163/E.164 (ISDN/telephony)
+; x - X.121 (Data)
+; f - F.69 (Telex)
+; n - National
+; p - Private
+; r - Reserved (should probably never be used but is included for completeness)
+;
+; You may also set the prilocaldialplan in the same way, but by prefixing the
+; Caller*ID Number rather than the dialed number.
+
+; Please note that telcos which require this kind of additional manipulation
+; of the TON/NPI are *rare*.  Most telco PRIs will work fine simply by
+; setting pridialplan to unknown or dynamic.
+;
+;
+; PRI caller ID prefixes based on the given TON/NPI (dialplan)
+; This is especially needed for EuroISDN E1-PRIs
+;
+; None of the prefix settings can be changed on reload.
+;
+; sample 1 for Germany
+;internationalprefix = 00
+;nationalprefix = 0
+;localprefix = 0711
+;privateprefix = 07115678
+;unknownprefix =
+;
+; sample 2 for Germany
+;internationalprefix = +
+;nationalprefix = +49
+;localprefix = +49711
+;privateprefix = +497115678
+;unknownprefix =
+;
+; PRI resetinterval: sets the time in seconds between restart of unused
+; B channels; defaults to 'never'.
+;
+;resetinterval = 3600
+;
+; Assume inband audio may be present when a SETUP ACK message is received.
+; Q.931 Section 5.1.3 says that in scenarios with overlap dialing, when a
+; dialtone is sent from the network side, progress indicator 8 "Inband info
+; now available" MAY be sent to the CPE if no digits were received with
+; the SETUP.  It is thus implied that the ie is mandatory if digits came
+; with the SETUP and dialtone is needed.
+; This option should be enabled, when the network sends dialtone and you
+; want to hear it, but the network doesn't send the progress indicator when
+; needed.
+;
+; NOTE: For Q.SIG setups this option should be enabled when outgoing overlap
+; dialing is also enabled because Q.SIG does not send the progress indicator
+; with the SETUP ACK.
+; Default no.
+;
+;inband_on_setup_ack=yes
+;
+; Assume inband audio may be present when a PROCEEDING message is received.
+; Q.931 Section 5.1.2 says the network cannot assume that the CPE side has
+; attached to the B channel at this time without explicitly sending the
+; progress indicator ie informing the CPE side to attach to the B channel
+; for audio.  However, some non-compliant ISDN switches send a PROCEEDING
+; without the progress indicator ie indicating inband audio is available and
+; assume that the CPE device has connected the media path for listening to
+; ringback and other messages.
+; Default no.
+;
+;inband_on_proceeding=yes
+;
+; Overlap dialing mode (sending overlap digits)
+; Cannot be changed on a reload.
+;
+; incoming: incoming direction only
+; outgoing: outgoing direction only
+; no: neither direction
+; yes or both: both directions
+;
+;overlapdial=yes
+
+; Send/receive ISDN display IE options.  The display options are a comma separated
+; list of the following options:
+;
+; block:        Do not pass display text data.
+;               Q.SIG: Default for send/receive.
+;               ETSI CPE: Default for send.
+; name_initial: Use display text in SETUP/CONNECT messages as the party name.
+;               Default for all other modes.
+; name_update:  Use display text in other messages (NOTIFY/FACILITY) for COLP name
+;               update.
+; name:         Combined name_initial and name_update options.
+; text:         Pass any unused display text data as an arbitrary display message
+;               during a call.  Sent text goes out in an INFORMATION message.
+;
+; * Default is an empty string for legacy behavior.
+; * The name options are not recommended for Q.SIG since Q.SIG already
+;   supports names.
+; * The send block is the only recommended setting for CPE mode since Q.931 uses
+;   the display IE only in the network to user direction.
+;
+; display_send and display_receive cannot be changed on reload.
+;
+;display_send=
+;display_receive=
+
+; Allow sending an ISDN Malicious Caller ID (MCID) request on this span.
+; Default disabled
+;
+;mcid_send=yes
+
+; Send ISDN date/time IE in CONNECT message option.  Only valid on NT spans.
+;
+; no:           Do not send date/time IE in CONNECT message.
+; date:         Send date only.
+; date_hh       Send date and hour.
+; date_hhmm     Send date, hour, and minute.
+; date_hhmmss   Send date, hour, minute, and second.
+;
+; Default is an empty string which lets libpri pick the default
+; date/time IE send policy.
+;
+;datetime_send=
+
+; Send ISDN conected line information.
+;
+; block:   Do not send any connected line information.
+; connect: Send connected line information on initial connect.
+; update:  Same as connect but also send any updates during a call.
+;          Updates happen if the call is transferred. (Default)
+;
+;colp_send=update
+
+; Allow inband audio (progress) when a call is DISCONNECTed by the far end of a PRI
+;
+;inbanddisconnect=yes
+;
+; Allow a held call to be transferred to the active call on disconnect.
+; This is useful on BRI PTMP NT lines where an ISDN phone can simulate the
+; transfer feature of an analog phone.
+; The default is no.
+;hold_disconnect_transfer=yes
+
+; BRI PTMP layer 1 presence.
+; You should normally not need to set this option.
+; You may need to set this option if your telco brings layer 1 down when
+; the line is idle.
+; required:      Layer 1 presence required for outgoing calls. (default)
+; ignore:        Ignore alarms from DAHDI about this span.
+;                (Layer 1 and 2 will be brought back up for an outgoing call.)
+;                NOTE:  You will not be able to detect physical line problems
+;                until an outgoing call is attempted and fails.
+;
+;layer1_presence=ignore
+
+; BRI PTMP layer 2 persistence.
+; You should normally not need to set this option.
+; You may need to set this option if your telco brings layer 1 down when
+; the line is idle.
+; <blank>:       Use libpri default.
+; keep_up:       Bring layer 2 back up if peer takes it down.
+; leave_down:    Leave layer 2 down if peer takes it down. (Libpri default)
+;                (Layer 2 will be brought back up for an outgoing call.)
+;
+;layer2_persistence=leave_down
+
+; PRI Out of band indications.
+; Enable this to report Busy and Congestion on a PRI using out-of-band
+; notification. Inband indication, as used by Asterisk doesn't seem to work
+; with all telcos.
+;
+; outofband:      Signal Busy/Congestion out of band with RELEASE/DISCONNECT
+; inband:         Signal Busy/Congestion using in-band tones (default)
+;
+; priindication cannot be changed on a reload.
+;
+;priindication = outofband
+;
+; If you need to override the existing channels selection routine and force all
+; PRI channels to be marked as exclusively selected, set this to yes.
+;
+; priexclusive cannot be changed on a reload.
+;
+;priexclusive = yes
+;
+;
+; If you need to use the logical channel mapping with your Q.SIG PRI instead
+; of the physical mapping you must use the qsigchannelmapping option.
+;
+; logical:  Use the logical channel mapping
+; physical: Use physical channel mapping (default)
+;
+;qsigchannelmapping=logical
+;
+; If you wish to ignore remote hold indications (and use MOH that is supplied over
+; the B channel) enable this option.
+;
+;discardremoteholdretrieval=yes
+;
+; ISDN Timers
+; All of the ISDN timers and counters that are used are configurable.  Specify
+; the timer name, and its value (in ms for timers).
+; K:    Layer 2 max number of outstanding unacknowledged I frames (default 7)
+; N200: Layer 2 max number of retransmissions of a frame (default 3)
+; T200: Layer 2 max time before retransmission of a frame (default 1000 ms)
+; T203: Layer 2 max time without frames being exchanged (default 10000 ms)
+; T305: Wait for DISCONNECT acknowledge (default 30000 ms)
+; T308: Wait for RELEASE acknowledge (default 4000 ms)
+; T309: Maintain active calls on Layer 2 disconnection (default 6000 ms)
+;       EuroISDN: 6000 to 12000 ms, according to (N200 + 1) x T200 + 2s
+;       May vary in other ISDN standards (Q.931 1993 : 90000 ms)
+; T313: Wait for CONNECT acknowledge, CPE side only (default 3000 ms)
+;
+; T-RESPONSE:   Maximum time to wait for a typical APDU response. (default 4000 ms)
+;               This is an implementation timer when the standard does not specify one.
+; T-ACTIVATE:   Request supervision timeout. (default 10000 ms)
+; T-RETENTION:  Maximum  time to wait for user A to activate call-completion. (default 30000 ms)
+;               Used by ETSI PTP, ETSI PTMP, and Q.SIG as the cc_offer_timer.
+; T-CCBS1:      T-STATUS timer equivalent for CC user A status. (default 4000 ms)
+; T-CCBS2:      Maximum  time the CCBS service will be active (default 45 min in ms)
+; T-CCBS3:      Maximum  time to wait for user A to respond to user B availability. (default 20000 ms)
+; T-CCBS5:      Network B CCBS supervision timeout. (default 60 min in ms)
+; T-CCBS6:      Network A CCBS supervision timeout. (default 60 min in ms)
+; T-CCNR2:      Maximum  time the CCNR service will be active (default 180 min in ms)
+; T-CCNR5:      Network B CCNR supervision timeout. (default 195 min in ms)
+; T-CCNR6:      Network A CCNR supervision timeout. (default 195 min in ms)
+; CC-T1:        Q.SIG CC request supervision timeout. (default 30000 ms)
+; CCBS-T2:      Q.SIG CCBS supervision timeout. (default 60 min in ms)
+; CCNR-T2:      Q.SIG CCNR supervision timeout. (default 195 min in ms)
+; CC-T3:        Q.SIG CC Maximum time to wait for user A to respond to user B availability. (default 30000 ms)
+;
+;pritimer => t200,1000
+;pritimer => t313,4000
+;
+; CC PTMP recall mode:
+; specific - Only the CC original party A can participate in the CC callback
+; global - Other compatible endpoints on the PTMP line can be party A in the CC callback
+;
+; cc_ptmp_recall_mode cannot be changed on a reload.
+;
+;cc_ptmp_recall_mode = specific
+;
+; CC Q.SIG Party A (requester) retain signaling link option
+; retain       Require that the signaling link be retained.
+; release      Request that the signaling link be released.
+; do_not_care  The responder is free to choose if the signaling link will be retained.
+;
+;cc_qsig_signaling_link_req = retain
+;
+; CC Q.SIG Party B (responder) retain signaling link option
+; retain       Prefer that the signaling link be retained.
+; release      Prefer that the signaling link be released.
+;
+;cc_qsig_signaling_link_rsp = retain
+;
+; See ccss.conf.sample for more options.  The timers described by ccss.conf.sample
+; are not used by ISDN for the native protocol since they are defined by the
+; standards and set by pritimer above.
+;
+; To enable transmission of facility-based ISDN supplementary services (such
+; as caller name from CPE over facility), enable this option.
+; Cannot be changed on a reload.
+;
+;facilityenable = yes
+;
+
+; This option enables Advice of Charge pass-through between the ISDN PRI and
+; Asterisk.  This option can be set to any combination of 's', 'd', and 'e' which
+; represent the different variants of Advice of Charge, AOC-S, AOC-D, and AOC-E.
+; Advice of Charge pass-through is currently only supported for ETSI.  Since most
+; AOC messages are sent on facility messages, the 'facilityenable' option must
+; also be enabled to fully support AOC pass-through.
+;
+;aoc_enable=s,d,e
+;
+; When this option is enabled, a hangup initiated by the ISDN PRI side of the
+; asterisk channel will result in the channel delaying its hangup in an
+; attempt to receive the final AOC-E message from its bridge.  The delay
+; period is configured as one half the T305 timer length. If the channel
+; is not bridged the hangup will occur immediatly without delay.
+;
+;aoce_delayhangup=yes
+
+; pritimer cannot be changed on a reload.
+;
+; Signalling method. The default is "auto". Valid values:
+; auto:           Use the current value from DAHDI.
+; em:             E & M
+; em_e1:          E & M E1
+; em_w:           E & M Wink
+; featd:          Feature Group D (The fake, Adtran style, DTMF)
+; featdmf:        Feature Group D (The real thing, MF (domestic, US))
+; featdmf_ta:     Feature Group D (The real thing, MF (domestic, US)) through
+;                 a Tandem Access point
+; featb:          Feature Group B (MF (domestic, US))
+; fgccama:        Feature Group C-CAMA (DP DNIS, MF ANI)
+; fgccamamf:      Feature Group C-CAMA MF (MF DNIS, MF ANI)
+; fxs_ls:         FXS (Loop Start)
+; fxs_gs:         FXS (Ground Start)
+; fxs_ks:         FXS (Kewl Start)
+; fxo_ls:         FXO (Loop Start)
+; fxo_gs:         FXO (Ground Start)
+; fxo_ks:         FXO (Kewl Start)
+; pri_cpe:        PRI signalling, CPE side
+; pri_net:        PRI signalling, Network side
+; bri_cpe:        BRI PTP signalling, CPE side
+; bri_net:        BRI PTP signalling, Network side
+; bri_cpe_ptmp:   BRI PTMP signalling, CPE side
+; bri_net_ptmp:   BRI PTMP signalling, Network side
+; sf:             SF (Inband Tone) Signalling
+; sf_w:           SF Wink
+; sf_featd:       SF Feature Group D (The fake, Adtran style, DTMF)
+; sf_featdmf:     SF Feature Group D (The real thing, MF (domestic, US))
+; sf_featb:       SF Feature Group B (MF (domestic, US))
+; e911:           E911 (MF) style signalling
+; ss7:            Signalling System 7
+; mfcr2:          MFC/R2 Signalling. To specify the country variant see 'mfcr2_variant'
+;
+; The following are used for Radio interfaces:
+; fxs_rx:         Receive audio/COR on an FXS kewlstart interface (FXO at the
+;                 channel bank)
+; fxs_tx:         Transmit audio/PTT on an FXS loopstart interface (FXO at the
+;                 channel bank)
+; fxo_rx:         Receive audio/COR on an FXO loopstart interface (FXS at the
+;                 channel bank)
+; fxo_tx:         Transmit audio/PTT on an FXO groundstart interface (FXS at
+;                 the channel bank)
+; em_rx:          Receive audio/COR on an E&M interface (1-way)
+; em_tx:          Transmit audio/PTT on an E&M interface (1-way)
+; em_txrx:        Receive audio/COR AND Transmit audio/PTT on an E&M interface
+;                 (2-way)
+; em_rxtx:        Same as em_txrx (for our dyslexic friends)
+; sf_rx:          Receive audio/COR on an SF interface (1-way)
+; sf_tx:          Transmit audio/PTT on an SF interface (1-way)
+; sf_txrx:        Receive audio/COR AND Transmit audio/PTT on an SF interface
+;                 (2-way)
+; sf_rxtx:        Same as sf_txrx (for our dyslexic friends)
+; ss7:            Signalling System 7
+;
+; signalling of a channel can not be changed on a reload.
+;
+;signalling=fxo_ls
+;
+; If you have an outbound signalling format that is different from format
+; specified above (but compatible), you can specify outbound signalling format,
+; (see below). The 'signalling' format specified will be the inbound signalling
+; format. If you only specify 'signalling', then it will be the format for
+; both inbound and outbound.
+;
+; outsignalling can only be one of:
+;   em, em_e1, em_w, sf, sf_w, sf_featd, sf_featdmf, sf_featb, featd,
+;   featdmf, featdmf_ta, e911, fgccama, fgccamamf
+;
+; outsignalling cannot be changed on a reload.
+;
+;signalling=featdmf
+;
+;outsignalling=featb
+;
+; For Feature Group D Tandem access, to set the default CIC and OZZ use these
+; parameters (Will not be updated on reload):
+;
+;defaultozz=0000
+;defaultcic=303
+;
+; A variety of timing parameters can be specified as well
+; The default values for those are "-1", which is to use the
+; compile-time defaults of the DAHDI kernel modules. The timing
+; parameters, (with the standard default from DAHDI):
+;
+;    prewink:     Pre-wink time (default 50ms)
+;    preflash:    Pre-flash time (default 50ms)
+;    wink:        Wink time (default 150ms)
+;    flash:       Flash time (default 750ms)
+;    start:       Start time (default 1500ms)
+;    rxwink:      Receiver wink time (default 300ms)
+;    rxflash:     Receiver flashtime (default 1250ms)
+;    debounce:    Debounce timing (default 600ms)
+;
+; None of them will update on a reload.
+;
+; How long generated tones (DTMF and MF) will be played on the channel
+; (in milliseconds).
+;
+; This is a global, rather than a per-channel setting. It will not be
+; updated on a reload.
+;
+;toneduration=100
+;
+; Whether or not to do distinctive ring detection on FXO lines:
+;
+;usedistinctiveringdetection=yes
+;
+; enable dring detection after caller ID for those countries like Australia
+; where the ring cadence is changed *after* the caller ID spill:
+;
+;distinctiveringaftercid=yes
+;
+; Whether or not to use caller ID:
+;
+usecallerid=yes
+;
+; Type of caller ID signalling in use
+;     bell     = bell202 as used in US (default)
+;     v23      = v23 as used in the UK
+;     v23_jp   = v23 as used in Japan
+;     dtmf     = DTMF as used in Denmark, Sweden and Netherlands
+;     smdi     = Use SMDI for caller ID.  Requires SMDI to be enabled (usesmdi).
+;
+;cidsignalling=v23
+;
+; What signals the start of caller ID
+;     ring        = a ring signals the start (default)
+;     polarity    = polarity reversal signals the start
+;     polarity_IN = polarity reversal signals the start, for India,
+;                   for dtmf dialtone detection; using DTMF.
+;     (see https://wiki.asterisk.org/wiki/display/AST/Caller+ID+in+India)
+;     dtmf        = causes monitor loop to look for dtmf energy on the
+;                   incoming channel to initate cid acquisition
+;
+;cidstart=polarity
+;
+; When cidstart=dtmf, the energy level on the line used to trigger dtmf cid
+; acquisition. This number is compared to the average over a packet of audio
+; of the absolute values of 16 bit signed linear samples. The default is set
+; to 256. The choice of 256 is arbitrary. The value you should select should
+; be high enough to prevent false detections while low enough to insure that
+; no dtmf spills are missed.
+;
+;dtmfcidlevel=256
+;
+; Whether or not to hide outgoing caller ID (Override with *67 or *82)
+; (If your dialplan doesn't catch it)
+;
+;hidecallerid=yes
+;
+; Enable if you need to hide just the name and not the number for legacy PBX use.
+; Only applies to PRI channels.
+;hidecalleridname=yes
+;
+; On UK analog lines, the caller hanging up determines the end of calls.  So
+; Asterisk hanging up the line may or may not end a call (DAHDI could just as
+; easily be re-attaching to a prior incoming call that was not yet hung up).
+; This option changes the hangup to wait for a dialtone on the line, before
+; marking the line as once again available for use with outgoing calls.
+; Specified in milliseconds, not set by default.
+;waitfordialtone=1000
+;
+; For analog lines, enables Asterisk to use dialtone detection per channel
+; if an incoming call was hung up before it was answered.  If dialtone is
+; detected, the call is hung up.
+; no:       Disabled. (Default)
+; yes:      Look for dialtone for 10000 ms after answer.
+; <number>: Look for dialtone for the specified number of ms after answer.
+; always:   Look for dialtone for the entire call.  Dialtone may return
+;           if the far end hangs up first.
+;
+;dialtone_detect=no
+;
+; The following option enables receiving MWI on FXO lines.  The default
+; value is no.
+; 	The mwimonitor can take the following values
+;		no - No mwimonitoring occurs. (default)
+; 		yes - The same as specifying fsk
+; 		fsk - the FXO line is monitored for MWI FSK spills
+;		fsk,rpas - the FXO line is monitored for MWI FSK spills preceded
+;			by a ring pulse alert signal.
+;		neon - The fxo line is monitored for the presence of NEON pulses
+;			indicating MWI.
+; When detected, an internal Asterisk MWI event is generated so that any other
+; part of Asterisk that cares about MWI state changes is notified, just as if
+; the state change came from app_voicemail.
+; For FSK MWI Spills, the energy level that must be seen before starting the
+; MWI detection process can be set with 'mwilevel'.
+;
+;mwimonitor=no
+;mwilevel=512
+;
+; This option is used in conjunction with mwimonitor.  This will get executed
+; when incoming MWI state changes.  The script is passed 2 arguments.  The
+; first is the corresponding configured mailbox, and the second is 1 or 0,
+; indicating if there are messages waiting or not.
+; Note: app_voicemail mailboxes are in the form of mailbox@context.
+;
+; /usr/local/bin/dahdinotify.sh 501@mailboxes 1
+;
+;mwimonitornotify=/usr/local/bin/dahdinotify.sh
+;
+; The following keyword 'mwisendtype' enables various VMWI methods on FXS lines (if supported).
+; The default is to send FSK only.
+; The following options are available;
+; 'rpas' Ring Pulse Alert Signal, alerts intelligent phones that a FSK message is about to be sent.
+; 'lrev' Line reversed to indicate messages waiting.
+; 'hvdc' 90Vdc OnHook DC voltage to indicate messages waiting.
+; 'hvac' or 'neon' 90Vac OnHook AC voltage to light Neon bulb.
+; 'nofsk' Disables FSK MWI spills from being sent out.
+; It is feasible that multiple options can be enabled.
+;mwisendtype=rpas,lrev
+;
+; Whether or not to enable call waiting on internal extensions
+; With this set to 'yes', busy extensions will hear the call-waiting
+; tone, and can use hook-flash to switch between callers. The Dial()
+; app will not return the "BUSY" result for extensions.
+;
+callwaiting=yes
+;
+; Configure the number of outstanding call waiting calls for internal ISDN
+; endpoints before bouncing the calls as busy.  This option is equivalent to
+; the callwaiting option for analog ports.
+; A call waiting call is a SETUP message with no B channel selected.
+; The default is zero to disable call waiting for ISDN endpoints.
+;max_call_waiting_calls=0
+;
+; Allow incoming ISDN call waiting calls.
+; A call waiting call is a SETUP message with no B channel selected.
+;allow_call_waiting_calls=no
+
+; Configure the ISDN span to indicate MWI for the list of mailboxes.
+; You can give a comma separated list of up to 8 mailboxes per span.
+; An empty list disables MWI.
+;
+; The default is an empty list.
+;mwi_mailboxes=vm-mailbox{,vm-mailbox}
+;  vm-mailbox = Internal voicemail mailbox identifier.
+;  Note: app_voicemail mailboxes must be in the form of mailbox@context.
+;mwi_mailboxes=501@mailboxes,502@mailboxes
+
+; Configure the ISDN mailbox number sent over the span for MWI mailboxes.
+; The position of the number in the list corresponds to the position in
+; mwi_mailboxes.  If either position in mwi_mailboxes or mwi_vm_boxes is
+; empty then that position is disabled.
+;
+; The default is an empty list.
+;mwi_vm_boxes=mailbox_number{,mailbox_number}
+;mwi_vm_boxes=501,502
+
+; Configure the ISDN span voicemail controlling numbers for MWI mailboxes.
+; What number to call for a user to retrieve voicemail messages.
+;
+; You can give a comma separated list of numbers.  The position of the number
+; corresponds to the position in mwi_mailboxes.  If a position is empty then
+; the last number is reused.
+;
+; For example:
+;  mwi_vm_numbers=700,,800,,900
+; is equivalent to:
+;  mwi_vm_numbers=700,700,800,800,900,900,900,900
+;
+; The default is no number.
+;mwi_vm_numbers=
+
+; Whether or not restrict outgoing caller ID (will be sent as ANI only, not
+; available for the user)
+; Mostly use with FXS ports
+; Does nothing.  Use hidecallerid instead.
+;
+;restrictcid=no
+;
+; Whether or not to use the caller ID presentation from the Asterisk channel
+; for outgoing calls.
+; See dialplan function CALLERID(pres) for more information.
+; Only applies to PRI and SS7 channels.
+;
+usecallingpres=yes
+;
+; Some countries (UK) have ring tones with different ring tones (ring-ring),
+; which means the caller ID needs to be set later on, and not just after
+; the first ring, as per the default (1).
+;
+;sendcalleridafter = 2
+;
+;
+; Support caller ID on Call Waiting
+;
+callwaitingcallerid=yes
+;
+; Support three-way calling
+;
+threewaycalling=yes
+;
+; For FXS ports (either direct analog or over T1/E1):
+;   Support flash-hook call transfer (requires three way calling)
+;   Also enables call parking (overrides the 'canpark' parameter)
+;
+; For digital ports using ISDN PRI protocols:
+;   Support switch-side transfer (called 2BCT, RLT or other names)
+;   This setting must be enabled on both ports involved, and the
+;   'facilityenable' setting must also be enabled to allow sending
+;   the transfer to the ISDN switch, since it sent in a FACILITY
+;   message.
+;   NOTE:  This should be disabled for NT PTMP mode.  Phones cannot
+;   have tromboned calls pushed down to them.
+;
+transfer=yes
+;
+; Allow call parking
+; ('canpark=no' is overridden by 'transfer=yes')
+;
+canpark=yes
+
+; Sets the default parking lot for call parking.
+; This is setable per channel.
+; Parkinglots are configured in features.conf
+;
+;parkinglot=plaza
+
+;
+; Support call forward variable
+;
+cancallforward=yes
+;
+; Whether or not to support Call Return (*69, if your dialplan doesn't
+; catch this first)
+;
+callreturn=yes
+;
+; Stutter dialtone support: If voicemail is received in the mailbox then
+; taking the phone off hook will cause a stutter dialtone instead of a
+; normal one.
+;
+; Note: app_voicemail mailboxes must be in the form of mailbox@context.
+;
+;mailbox=1234@context
+;
+; Enable echo cancellation
+; Use either "yes", "no", or a power of two from 32 to 256 if you wish to
+; actually set the number of taps of cancellation.
+;
+; Note that when setting the number of taps, the number 256 does not translate
+; to 256 ms of echo cancellation.  echocancel=256 means 256 / 8 = 32 ms.
+;
+; Note that if any of your DAHDI cards have hardware echo cancellers,
+; then this setting only turns them on and off; numeric settings will
+; be treated as "yes". There are no special settings required for
+; hardware echo cancellers; when present and enabled in their kernel
+; modules, they take precedence over the software echo canceller compiled
+; into DAHDI automatically.
+;
+;
+echocancel=yes
+;
+; Some DAHDI echo cancellers (software and hardware) support adjustable
+; parameters; these parameters can be supplied as additional options to
+; the 'echocancel' setting. Note that Asterisk does not attempt to
+; validate the parameters or their values, so if you supply an invalid
+; parameter you will not know the specific reason it failed without
+; checking the kernel message log for the error(s) put there by DAHDI.
+;
+;echocancel=128,param1=32,param2=0,param3=14
+;
+; Generally, it is not necessary (and in fact undesirable) to echo cancel when
+; the circuit path is entirely TDM.  You may, however, change this behavior
+; by enabling the echo canceller during pure TDM bridging below.
+;
+echocancelwhenbridged=yes
+;
+; In some cases, the echo canceller doesn't train quickly enough and there
+; is echo at the beginning of the call.  Enabling echo training will cause
+; DAHDI to briefly mute the channel, send an impulse, and use the impulse
+; response to pre-train the echo canceller so it can start out with a much
+; closer idea of the actual echo.  Value may be "yes", "no", or a number of
+; milliseconds to delay before training (default = 400)
+;
+; WARNING:  In some cases this option can make echo worse!  If you are
+; trying to debug an echo problem, it is worth checking to see if your echo
+; is better with the option set to yes or no.  Use whatever setting gives
+; the best results.
+;
+; Note that these parameters do not apply to hardware echo cancellers.
+;
+;echotraining=yes
+;echotraining=800
+;
+; If you are having trouble with DTMF detection, you can relax the DTMF
+; detection parameters.  Relaxing them may make the DTMF detector more likely
+; to have "talkoff" where DTMF is detected when it shouldn't be.
+;
+;relaxdtmf=yes
+;
+; Hardware gain settings increase/decrease the analog volume level on a channel.
+;   The values are in db (decibels) and can be adjusted in 0.1 dB increments.
+;   A positive number increases the volume level on a channel, and a negavive
+;   value decreases volume level.
+;
+;   Hardware gain settings are only possible on hardware with analog ports
+;   because the gain is done on the analog side of the analog/digital conversion.
+;
+;   When hardware gains are disabled, Asterisk will NOT touch the gain setting
+;   already configured in hardware.
+;
+;   hwrxgain: Hardware receive gain for the channel (into Asterisk).
+;             Default: disabled
+;   hwtxgain: Hardware transmit gain for the channel (out of Asterisk).
+;             Default: disabled
+;
+;hwrxgain=disabled
+;hwtxgain=disabled
+;hwrxgain=2.0
+;hwtxgain=3.0
+;
+; Software gain settings digitally increase/decrease the volume level on a channel.
+;   The values are in db (decibels).  A positive number increases the volume
+;   level on a channel, and a negavive value decreases volume level.
+;
+;   Software gains work on the digital side of the analog/digital conversion
+;   and thus can also work with T1/E1 cards.
+;
+;   rxgain: Software receive gain for the channel (into Asterisk). Default: 0.0
+;   txgain: Software transmit gain for the channel (out of Asterisk).
+;             Default: 0.0
+;
+;   cid_rxgain: Add this gain to rxgain when Asterisk expects to receive
+;               a Caller ID stream.
+;               Default: 5.0 .
+;
+;rxgain=2.0
+;txgain=3.0
+;
+; Dynamic Range Compression: You can also enable dynamic range compression
+;   on a channel.  This will digitally amplify quiet sounds while leaving louder
+;   sounds untouched.  This is useful in situations where a linear gain setting
+;   would cause clipping.  Acceptable values are in the range of 0.0 to around
+;   6.0 with higher values causing more compression to be done.
+;
+;   rxdrc: dynamic range compression for the rx channel. Default: 0.0
+;   txdrc: dynamic range compression for the tx channel. Default: 0.0
+;
+;rxdrc=1.0
+;txdrc=4.0
+;
+; Logical groups can be assigned to allow outgoing roll-over.  Groups range
+; from 0 to 63, and multiple groups can be specified. By default the
+; channel is not a member of any group.
+;
+; Note that an explicit empty value for 'group' is invalid, and will not
+; override a previous non-empty one. The same applies to callgroup and
+; pickupgroup as well.
+;
+group=1
+;
+; Ring groups (a.k.a. call groups) and pickup groups.  If a phone is ringing
+; and it is a member of a group which is one of your pickup groups, then
+; you can answer it by picking up and dialing *8#.  For simple offices, just
+; make these both the same.  Groups range from 0 to 63.
+;
+callgroup=1
+pickupgroup=1
+;
+; Named ring groups (a.k.a. named call groups) and named pickup groups.
+; If a phone is ringing and it is a member of a group which is one of your
+; named pickup groups, then you can answer it by picking up and dialing *8#.
+; For simple offices, just make these both the same.
+; The number of named groups is not limited.
+;
+;namedcallgroup=engineering,sales,netgroup,protgroup
+;namedpickupgroup=sales
+
+; Channel variables to be set for all calls from this channel
+;setvar=CHANNEL=42
+;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep   ; This channel variable will
+                                                ; cause the given audio file to
+                                                ; be played upon completion of
+                                                ; an attended transfer to the
+                                                ; target of the transfer.
+
+;
+; Specify whether the channel should be answered immediately or if the simple
+; switch should provide dialtone, read digits, etc.
+; Note: If immediate=yes the dialplan execution will always start at extension
+; 's' priority 1 regardless of the dialed number!
+;
+;immediate=yes
+;
+; Specify whether flash-hook transfers to 'busy' channels should complete or
+; return to the caller performing the transfer (default is yes).
+;
+;transfertobusy=no
+
+; Calls will have the party id user tag set to this string value.
+;
+;cid_tag=
+
+; With this set, you can automatically append the MSN of a party
+; to the cid_tag.  An '_' is used to separate the tag from the MSN.
+; Applies to ISDN spans.
+; Default is no.
+;
+; Table of what number is appended:
+;      outgoing  incoming
+; net  dialed    caller
+; cpe  caller    dialed
+;
+;append_msn_to_cid_tag=no
+
+; caller ID can be set to "asreceived" or a specific number if you want to
+; override it.  Note that "asreceived" only applies to trunk interfaces.
+; fullname sets just the
+;
+; fullname: sets just the name part.
+; cid_number: sets just the number part:
+;
+;callerid = 123456
+;
+;callerid = My Name <2564286000>
+; Which can also be written as:
+;cid_number = 2564286000
+;fullname = My Name
+;
+;callerid = asreceived
+;
+; should we use the caller ID from incoming call on DAHDI transfer?
+;
+;useincomingcalleridondahditransfer = yes
+;
+; Add a description for the channel which can be shown through the Asterisk
+; console  when executing the 'dahdi show channels' command is run.
+;
+;description=Phone located in lobby
+;
+; AMA flags affects the recording of Call Detail Records.  If specified
+; it may be 'default', 'omit', 'billing', or 'documentation'.
+;
+;amaflags=default
+;
+; Channels may be associated with an account code to ease
+; billing
+;
+;accountcode=lss0101
+;
+; ADSI (Analog Display Services Interface) can be enabled on a per-channel
+; basis if you have (or may have) ADSI compatible CPE equipment
+;
+;adsi=yes
+;
+; SMDI (Simplified Message Desk Interface) can be enabled on a per-channel
+; basis if you would like that channel to behave like an SMDI message desk.
+; The SMDI port specified should have already been defined in smdi.conf.  The
+; default port is /dev/ttyS0.
+;
+;usesmdi=yes
+;smdiport=/dev/ttyS0
+;
+; On trunk interfaces (FXS) and E&M interfaces (E&M, Wink, Feature Group D
+; etc, it can be useful to perform busy detection either in an effort to
+; detect hangup or for detecting busies.  This enables listening for
+; the beep-beep busy pattern.
+;
+;busydetect=yes
+;
+; If busydetect is enabled, it is also possible to specify how many busy tones
+; to wait for before hanging up.  The default is 3, but it might be
+; safer to set to 6 or even 8.  Mind that the higher the number, the more
+; time that will be needed to hangup a channel, but lowers the probability
+; that you will get random hangups.
+;
+;busycount=6
+;
+; If busydetect is enabled, it is also possible to specify the cadence of your
+; busy signal.  In many countries, it is 500msec on, 500msec off.  Without
+; busypattern specified, we'll accept any regular sound-silence pattern that
+; repeats <busycount> times as a busy signal.  If you specify busypattern,
+; then we'll further check the length of the sound (tone) and silence, which
+; will further reduce the chance of a false positive.
+;
+;busypattern=500,500
+;
+; NOTE: In make menuselect, you'll find further options to tweak the busy
+; detector.  If your country has a busy tone with the same length tone and
+; silence (as many countries do), consider enabling the
+; BUSYDETECT_COMPARE_TONE_AND_SILENCE option.
+;
+; To further detect which hangup tone your telco provider is sending, it is
+; useful to use the dahdi_monitor utility to record the audio that main/dsp.c
+; is receiving after the caller hangs up.
+;
+; For FXS (FXO signalled) ports
+;   switch the line polarity to signal the connected PBX that an outgoing
+;   call was answered by the remote party.
+; For FXO (FXS signalled) ports
+;   watch for a polarity reversal to mark when a outgoing call is
+;   answered by the remote party.
+;
+;answeronpolarityswitch=yes
+;
+; For FXS (FXO signalled) ports
+;   switch the line polarity to signal the connected PBX that the current
+;   call was "hung up" by the remote party
+; For FXO (FXS signalled) ports
+;   In some countries, a polarity reversal is used to signal the disconnect of a
+;   phone line.  If the hanguponpolarityswitch option is selected, the call will
+;   be considered "hung up" on a polarity reversal.
+;
+;hanguponpolarityswitch=yes
+;
+; polarityonanswerdelay: minimal time period (ms) between the answer
+;                        polarity switch and hangup polarity switch.
+;                        (default: 600ms)
+;
+; On trunk interfaces (FXS) it can be useful to attempt to follow the progress
+; of a call through RINGING, BUSY, and ANSWERING.   If turned on, call
+; progress attempts to determine answer, busy, and ringing on phone lines.
+; This feature is HIGHLY EXPERIMENTAL and can easily detect false answers,
+; so don't count on it being very accurate.
+;
+; Few zones are supported at the time of this writing, but may be selected
+; with "progzone".
+;
+; progzone also affects the pattern used for buzydetect (unless
+; busypattern is set explicitly). The possible values are:
+;   us (default)
+;   ca (alias for 'us')
+;   cr (Costa Rica)
+;   br (Brazil, alias for 'cr')
+;   uk
+;
+; This feature can also easily detect false hangups. The symptoms of this is
+; being disconnected in the middle of a call for no reason.
+;
+;callprogress=yes
+;progzone=uk
+;
+; Set the tonezone. Equivalent of the defaultzone settings in
+; /etc/dahdi/system.conf. This sets the tone zone by number.
+; Note that you'd still need to load tonezones (loadzone in
+; /etc/dahdi/system.conf).
+; The default is -1: not to set anything.
+;tonezone = 0 ; 0 is US
+;
+; FXO (FXS signalled) devices must have a timeout to determine if there was a
+; hangup before the line was answered.  This value can be tweaked to shorten
+; how long it takes before DAHDI considers a non-ringing line to have hungup.
+;
+; ringtimeout will not update on a reload.
+;
+;ringtimeout=8000
+;
+; For FXO (FXS signalled) devices, whether to use pulse dial instead of DTMF
+; Pulse digits from phones (FXS devices, FXO signalling) are always
+; detected.
+;
+;pulsedial=yes
+;
+; For fax detection, uncomment one of the following lines.  The default is *OFF*
+;
+;faxdetect=both
+;faxdetect=incoming
+;faxdetect=outgoing
+;faxdetect=no
+;
+; When 'faxdetect' is used, one could use 'faxbuffers' to configure the DAHDI
+; transmit buffer policy.  The default is *OFF*.  When this configuration
+; option is used, the faxbuffer policy will be used for the life of the call
+; after a fax tone is detected.  The faxbuffer policy is reverted after the
+; call is torn down.  The sample below will result in 6 buffers and a full
+; buffer policy.
+;
+;faxbuffers=>6,full
+;
+; Configure the default number of DAHDI buffers and the transmit policy to use.
+; This can be used to eliminate data drops when scheduling jitter prevents
+; Asterisk from writing to a DAHDI channel regularly. Most users will probably
+; want "faxbuffers" instead of "buffers".
+;
+; The policies are:
+; immediate - DAHDI will immediately start sending the data to the hardware after
+;             Asterisk writes to the channel. This is the default mode. It
+;             introduces the least amount of latency but has an increased chance for
+;             hardware under runs if Asterisk is not able to keep the DAHDI write
+;             queue from going empty.
+; half      - DAHDI will wait until half of the configured buffers are full before
+;             starting to transmit. This adds latency to the audio but reduces
+;             the chance of under runs. Essentially, this is like an in-kernel jitter
+;             buffer.
+; full      - DAHDI will not start transmitting until all buffers are full.
+;             Introduces the most amount of latency and is susceptible to over
+;             runs from the Asterisk process.
+;
+; The receive policy is never changed. DAHDI will always pass up audio as soon
+; as possible.
+;
+; The default number of buffers is 4 (from jitterbuffers) and the default policy
+; is immediate.
+;
+;buffers=4,immediate
+;
+; This option specifies what to do when the channel's bridged peer puts the
+; ISDN channel on hold.  Settable per logical ISDN span.
+; moh:          Generate music-on-hold to the remote party.
+; notify:       Send hold notification signaling to the remote party.
+;               For ETSI PTP and ETSI PTMP NT links.
+;               (The notify setting deprecates the mohinterpret=passthrough setting.)
+; hold:         Use HOLD/RETRIEVE signaling to release the B channel while on hold.
+;               For ETSI PTMP TE links.
+;
+;moh_signaling=moh
+;
+; 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.
+;
+; This option may be set globally or on a per-channel basis.
+;
+;mohinterpret=default
+;
+; This option specifies which music on hold class to suggest to the peer channel
+; when this channel places the peer on hold.  This option may be set globally,
+; or on a per-channel basis.
+;
+;mohsuggest=default
+;
+; PRI channels can have an idle extension and a minunused number.  So long as
+; at least "minunused" channels are idle, chan_dahdi will try to call "idledial"
+; on them, and then dump them into the PBX in the "idleext" extension (which
+; is of the form exten@context).  When channels are needed the "idle" calls
+; are disconnected (so long as there are at least "minidle" calls still
+; running, of course) to make more channels available.  The primary use of
+; this is to create a dynamic service, where idle channels are bundled through
+; multilink PPP, thus more efficiently utilizing combined voice/data services
+; than conventional fixed mappings/muxings.
+;
+; Those settings cannot be changed on reload.
+;
+;idledial=6999
+;idleext=6999@dialout
+;minunused=2
+;minidle=1
+;
+;
+; ignore_failed_channels: Continue even if some channels failed to configure.
+; True by default. Disable this if you can guarantee that DAHDI starts before
+; Asterisk and want to be sure chan_dahdi will not start with broken
+; configuration.
+;
+;ignore_failed_channels = false
+;
+; Configure jitter buffers in DAHDI (each one is 20ms, default is 4)
+; This is set globally, rather than per-channel.
+;
+;jitterbuffers=4
+;
+;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
+; jbenable = yes              ; Enables the use of a jitterbuffer on the receiving side of a
+                              ; DAHDI channel. Defaults to "no". An enabled jitterbuffer will
+                              ; be used only if the sending side can create and the receiving
+                              ; side can not accept jitter. The DAHDI channel can't accept jitter,
+                              ; thus an enabled jitterbuffer on the receive DAHDI side will always
+                              ; be used if the sending side can create jitter.
+
+; jbmaxsize = 200             ; Max length of the jitterbuffer in milliseconds.
+
+; jbresyncthreshold = 1000    ; Jump in the frame timestamps over which the jitterbuffer is
+                              ; resynchronized. Useful to improve the quality of the voice, with
+                              ; big jumps in/broken timestamps, usually sent from exotic devices
+                              ; and programs. Defaults to 1000.
+
+; jbimpl = fixed              ; Jitterbuffer implementation, used on the receiving side of a DAHDI
+                              ; channel. Two implementations are currently available - "fixed"
+                              ; (with size always equals to jbmax-size) and "adaptive" (with
+                              ; variable size, actually the new jb of IAX2). Defaults to fixed.
+
+; 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".
+;-----------------------------------------------------------------------------------
+;
+; You can define your own custom ring cadences here.  You can define up to 8
+; pairs.  If the silence is negative, it indicates where the caller ID spill is
+; to be placed.  Also, if you define any custom cadences, the default cadences
+; will be turned off.
+;
+; This setting is global, rather than per-channel. It will not update on
+; a reload.
+;
+; Syntax is:  cadence=ring,silence[,ring,silence[...]]
+;
+; These are the default cadences:
+;
+;cadence=125,125,2000,-4000
+;cadence=250,250,500,1000,250,250,500,-4000
+;cadence=125,125,125,125,125,-4000
+;cadence=1000,500,2500,-5000
+;
+; Each channel consists of the channel number or range.  It inherits the
+; parameters that were specified above its declaration.
+;
+;
+;callerid="Green Phone"<(256) 428-6121>
+;description=Reception Phone			; add a description for 'dahdi show channels'
+;channel => 1
+;callerid="Black Phone"<(256) 428-6122>
+;description=Courtesy Phone
+;channel => 2
+;callerid="CallerID Phone" <(630) 372-1564>
+;description=					; reset the description for following channels
+;channel => 3
+;callerid="Pac Tel Phone" <(256) 428-6124>
+;channel => 4
+;callerid="Uniden Dead" <(256) 428-6125>
+;channel => 5
+;callerid="Cortelco 2500" <(256) 428-6126>
+;channel => 6
+;callerid="Main TA 750" <(256) 428-6127>
+;channel => 44
+;
+; For example, maybe we have some other channels which start out in a
+; different context and use E & M signalling instead.
+;
+;context=remote
+;signaling=em
+;channel => 15
+;channel => 16
+
+;signalling=em_w
+;
+; All those in group 0 I'll use for outgoing calls
+;
+; Strip most significant digit (9) before sending
+;
+;stripmsd=1
+;callerid=asreceived
+;group=0
+;signalling=fxs_ls
+;channel => 45
+
+;signalling=fxo_ls
+;group=1
+;callerid="Joe Schmoe" <(256) 428-6131>
+;channel => 25
+;callerid="Megan May" <(256) 428-6132>
+;channel => 26
+;callerid="Suzy Queue" <(256) 428-6233>
+;channel => 27
+;callerid="Larry Moe" <(256) 428-6234>
+;channel => 28
+;
+; Sample PRI (CPE) config:  Specify the switchtype, the signalling as either
+; pri_cpe or pri_net for CPE or Network termination, and generally you will
+; want to create a single "group" for all channels of the PRI.
+;
+; switchtype cannot be changed on a reload.
+;
+; switchtype = national
+; signalling = pri_cpe
+; group = 2
+; channel => 1-23
+;
+; Alternatively, the number of the channel may be replaced with a relative
+; path to a device file under /dev/dahdi . The final element of that file
+; must be a number, though. The directory separator is '!', as we can't
+; use '/' in a dial string. So if we have
+;
+;   /dev/dahdi/span-name/pstn/00/1
+;   /dev/dahdi/span-name/pstn/00/2
+;   /dev/dahdi/span-name/pstn/00/3
+;   /dev/dahdi/span-name/pstn/00/4
+;
+; we could use:
+;channel => span-name!pstn!00!1-4
+;
+; or:
+;channel => span-name!pstn!00!1,2,3,4
+;
+; See also ignore_failed_channels above.
+
+;  Used for distinctive ring support for x100p.
+;  You can see the dringX patterns is to set any one of the dringXcontext fields
+;  and they will be printed on the console when an inbound call comes in.
+;
+;  dringXrange is used to change the acceptable ranges for "tone offsets".  Defaults to 10.
+;  Note: a range of 0 is NOT what you might expect - it instead forces it to the default.
+;  A range of -1 will force it to always match.
+;  Anything lower than -1 would presumably cause it to never match.
+;
+;dring1=95,0,0
+;dring1context=internal1
+;dring1range=10
+;dring2=325,95,0
+;dring2context=internal2
+;dring2range=10
+; If no pattern is matched here is where we go.
+;context=default
+;channel => 1
+
+; AMI alarm event reporting
+;reportalarms=channels
+;Possible values are:
+;channels - report each channel alarms (current behavior, default for backward compatibility)
+;spans - report an "SpanAlarm" event when the span of any configured channel is alarmed
+;all - report channel and span alarms (aggregated behavior)
+;none - do not report any alarms.
+
+; ---------------- Options for use with signalling=ss7 -----------------
+; None of them can be changed by a reload.
+;
+; Variant of SS7 signalling:
+; Options are itu and ansi
+;ss7type = itu
+
+; SS7 Called Nature of Address Indicator
+;
+; unknown:        Unknown
+; subscriber:     Subscriber
+; national:       National
+; international:  International
+; dynamic:        Dynamically selects the appropriate dialplan
+;
+;ss7_called_nai=dynamic
+;
+; SS7 Calling Nature of Address Indicator
+;
+; unknown:        Unknown
+; subscriber:     Subscriber
+; national:       National
+; international:  International
+; dynamic:        Dynamically selects the appropriate dialplan
+;
+;ss7_calling_nai=dynamic
+;
+;
+; sample 1 for Germany
+;ss7_internationalprefix = 00
+;ss7_nationalprefix = 0
+;ss7_subscriberprefix =
+;ss7_unknownprefix =
+;
+
+; This option is used to disable automatic sending of ACM when the call is started
+; in the dialplan.  If you do use this option, you will need to use the Proceeding()
+; application in the dialplan to send ACM or enable ss7_autoacm below.
+;ss7_explicitacm=yes
+
+; Use this option to automatically send ACM when the call rings or is answered and
+; has not seen proceeding yet. If you use this option, you should disable ss7_explicitacm.
+; You may still use Proceeding() to explicitly send an ACM from the dialplan.
+;ss7_autoacm=yes
+
+; Create the linkset with all CICs in hardware remotely blocked state.
+;ss7_initialhwblo=yes
+
+; This option is whether or not to trust the remote echo control indication.  This means
+; that in cases where echo control is reported by the remote end, we will trust them and
+; not enable echo cancellation on the call.
+;ss7_use_echocontrol=yes
+
+; This option is to set what our echo control indication is to the other end.  Set to
+; yes to indicate that we are using echo cancellation or no if we are not.
+;ss7_default_echocontrol=yes
+
+; All settings apply to linkset 1
+;linkset = 1
+
+; Set the Signaling Link Code (SLC) for each sigchan.
+; If you manually set any you need to manually set all.
+; Should be defined before sigchan.
+; The default SLC starts with zero and increases for each defined sigchan.
+;slc=
+
+; Point code of the linkset.  For ITU, this is the decimal number
+; format of the point code.  For ANSI, this can either be in decimal
+; number format or in the xxx-xxx-xxx format
+;pointcode = 1
+
+; Point code of node adjacent to this signalling link (Possibly the STP between you and
+; your destination).  Point code format follows the same rules as above.
+;adjpointcode = 2
+
+; Default point code that you would like to assign to outgoing messages (in case of
+; routing through STPs, or using A links).  Point code format follows the same rules
+; as above.
+;defaultdpc = 3
+
+; Begin CIC (Circuit indication codes) count with this number
+;cicbeginswith = 1
+
+; What the MTP3 network indicator bits should be set to.  Choices are
+; national, national_spare, international, international_spare
+;networkindicator=international
+
+; First signalling channel
+;sigchan = 48
+
+; Additional signalling channel for this linkset (So you can have a linkset
+; with two signalling links in it).  It seems like a silly way to do it, but
+; for linksets with multiple signalling links, you add an additional sigchan
+; line for every additional signalling link on the linkset.
+;sigchan = 96
+
+; Channels to associate with CICs on this linkset
+;channel = 25-47
+;
+
+; Set this option if you wish to send an Information Request Message (INR) request
+; if no calling party number is specified. This will attempt to tell the other end
+; to send it anyways. Should be defined after sigchan.
+;inr_if_no_calling=yes
+
+; Set this to set whether or not the originating access is (non) ISDN in the forward and
+; backward call indicators. Should be defined after sigchan
+;non_isdn_access=yes
+
+; This sets the number of binary places to shift the CIC when doing load balancing between
+; sigchans on a linkset. Should be defined after sigchan. Default 0
+;sls_shift = 0
+
+; Send custom cause_location value
+; Should be defined after sigchan. Default 1 (private local)
+;cause_location=1
+
+; SS7 timers (ISUP and MTP3) should be explicitly defined for each linkset to be used.
+; For a full list of supported timers and their default values (applicable for both ITU
+; and ANSI) see ss7.timers
+; Should be defined after sigchan
+;#include ss7.timers
+
+; For more information on setting up SS7, see the README file in libss7 or
+; https://wiki.asterisk.org/wiki/display/AST/Signaling+System+Number+7
+; ----------------- SS7 Options ----------------------------------------
+
+; ---------------- Options for use with signalling=mfcr2 --------------
+
+; MFC-R2 signaling has lots of variants from country to country and even sometimes
+; minor variants inside the same country. The only mandatory parameters here are:
+; mfcr2_variant, mfcr2_max_ani and mfcr2_max_dnis.
+; IT IS RECOMMENDED that you leave the default values (leaving it commented) for the 
+; other parameters unless you have problems or you have been instructed to change some 
+; parameter. OpenR2 library uses the mfcr2_variant parameter to try to determine the 
+; best defaults for your country, also refer to the OpenR2 package directory 
+; doc/asterisk/ where you can find sample configurations for some countries. If you 
+; want to contribute your configs for a particular country send them to the e-mail 
+; of the primary OpenR2 developer that you can find in the AUTHORS file of the OpenR2 package
+
+; MFC/R2 variant. This depends on the OpenR2 supported variants
+; A list of values can be found by executing the openr2 command r2test -l
+; some valid values are:
+; ar (Argentina)
+; br (Brazil)
+; mx (Mexico)
+; ph (Philippines)
+; itu (per ITU spec)
+; mfcr2_variant=mx
+
+; Max amount of ANI to ask for
+; mfcr2_max_ani=10
+
+; Max amount of DNIS to ask for
+; mfcr2_max_dnis=4
+
+; whether or not to get the ANI before getting DNIS.
+; some telcos require ANI first some others do not care
+; if this go wrong, change this value
+; mfcr2_get_ani_first=no
+
+; Caller Category to send
+; national_subscriber
+; national_priority_subscriber
+; international_subscriber
+; international_priority_subscriber
+; collect_call
+; usually national_subscriber works just fine
+; you can change this setting from the dialplan
+; by setting the variable MFCR2_CATEGORY
+; (remember to set _MFCR2_CATEGORY from originating channels)
+; MFCR2_CATEGORY will also be a variable available in your context
+; on incoming calls set to the value received from the far end
+; mfcr2_category=national_subscriber
+
+; Call logging is stored at the Asterisk
+; logging directory specified in asterisk.conf
+; plus mfcr2/<whatever you put here>
+; if you specify 'span1' here and asterisk.conf has
+; as logging directory /var/log/asterisk then the full
+; path to your MFC/R2 call logs will be /var/log/asterisk/mfcr2/span1
+; (the directory will be automatically created if not present already)
+; remember to set mfcr2_call_files=yes
+; mfcr2_logdir=span1
+
+; whether or not to drop call files into mfcr2_logdir
+; mfcr2_call_files=yes|no
+
+; MFC/R2 valid logging values are: all,error,warning,debug,notice,cas,mf,stack,nothing
+; error,warning,debug and notice are self-descriptive
+; 'cas' is for logging ABCD CAS tx and rx
+; 'mf' is for logging of the Multi Frequency tones
+; 'stack' is for very verbose output of the channel and context call stack, only useful
+; if you are debugging a crash or want to learn how the library works. The stack logging
+; will be only enabled if the openr2 library was compiled with -DOR2_TRACE_STACKS
+; You can mix up values, like: loglevel=error,debug,mf to log just error, debug and
+; multi frequency messages
+; 'all' is a special value to log all the activity
+; 'nothing' is a clean-up value, in case you want to not log any activity for
+; a channel or group of channels
+; BE AWARE that the level of output logged will ALSO depend on
+; the value you have in logger.conf, if you disable output in logger.conf
+; then it does not matter you specify 'all' here, nothing will be logged
+; so logger.conf has the last word on what is going to be logged
+; mfcr2_logging=all
+
+; MFC/R2 value in milliseconds for the MF timeout. Any negative value
+; means 'default', smaller values than 500ms are not recommended
+; and can cause malfunctioning. If you experience protocol error
+; due to MF timeout try incrementing this value in 500ms steps
+; mfcr2_mfback_timeout=-1
+
+; MFC/R2 value in milliseconds for the metering pulse timeout.
+; Metering pulses are sent by some telcos for some R2 variants
+; during a call presumably for billing purposes to indicate costs,
+; however this pulses use the same signal that is used to indicate
+; call hangup, therefore a timeout is sometimes required to distinguish
+; between a *real* hangup and a billing pulse that should not
+; last more than 500ms, If you experience call drops after some
+; minutes of being stablished try setting a value of some ms here,
+; values greater than 500ms are not recommended.
+; BE AWARE that choosing the proper protocol mfcr2_variant parameter
+; implicitly sets a good recommended value for this timer, use this
+; parameter only when you *really* want to override the default, otherwise
+; just comment out this value or put a -1
+; Any negative value means 'default'.
+; mfcr2_metering_pulse_timeout=-1
+
+; Brazil uses a special calling party category for collect calls (llamadas por cobrar)
+; instead of using the operator (as in Mexico). The R2 spec in Brazil says a special GB tone
+; should be used to reject collect calls. If you want to ALLOW collect calls specify 'yes',
+; if you want to BLOCK collect calls then say 'no'. Default is to block collect calls.
+; (see also 'mfcr2_double_answer')
+; mfcr2_allow_collect_calls=no
+
+; This feature is related but independent of mfcr2_allow_collect_calls
+; Some PBX's require a double-answer process to block collect calls, if
+; you ever have problems blocking collect calls using Group B signals (mfcr2_allow_collect_calls=no)
+; then you may want to try with mfcr2_double_answer=yes, this will cause that every answer signal
+; is changed by answer->clear back->answer (sort of a flash)
+; (see also 'mfcr2_allow_collect_calls')
+; mfcr2_double_answer=no
+
+; This feature allows to skip the use of Group B/II signals and go directly
+; to the accepted state for incoming calls
+; mfcr2_immediate_accept=no
+
+; You most likely dont need this feature. Default is yes.
+; When this is set to yes, all calls that are offered (incoming calls) which
+; DNIS is valid (exists in extensions.conf) and pass collect call validation
+; will be accepted with a Group B tone (either call with charge or not, depending on mfcr2_charge_calls)
+; with this set to 'no' then the call will NOT be accepted on offered, and the call will start its
+; execution in extensions.conf without being accepted until the channel is answered (either with Answer() or
+; any other application resulting in the channel being answered).
+; This can be set to 'no' if your telco or PBX needs the hangup cause to be set accurately
+; when this option is set to no you must explicitly accept the call with DAHDIAcceptR2Call
+; or implicitly through the Answer() application.
+; mfcr2_accept_on_offer=yes
+
+; Skip request of calling party category and ANI
+; you need openr2 >= 1.2.0 to use this feature
+; mfcr2_skip_category=no
+
+; WARNING: advanced users only! I really mean it
+; this parameter is commented by default because
+; YOU DON'T NEED IT UNLESS YOU REALLY GROK MFC/R2
+; READ COMMENTS on doc/r2proto.conf in openr2 package
+; for more info
+; mfcr2_advanced_protocol_file=/path/to/r2proto.conf
+
+; Brazil use a special signal to force the release of the line (hangup) from the
+; backward perspective. When mfcr2_forced_release=no, the normal clear back signal
+; will be sent on hangup, which is OK for all mfcr2 variants I know of, except for
+; Brazilian variant, where the central will leave the line up for several seconds (30, 60)
+; which sometimes is not what people really want. When mfcr2_forced_release=yes, a different
+; signal will be sent to hangup the call indicating that the line should be released immediately
+; mfcr2_forced_release=no
+
+; Whether or not report to the other end 'accept call with charge'
+; This setting has no effect with most telecos, usually is safe
+; leave the default (yes), but once in a while when interconnecting with
+; old PBXs this may be useful.
+; Concretely this affects the Group B signal used to accept calls
+; The application DAHDIAcceptR2Call can also be used to decide this
+; in the dial plan in a per-call basis instead of doing it here for all calls
+; mfcr2_charge_calls=yes
+
+; ---------------- END of options to be used with signalling=mfcr2
+
+; Configuration Sections
+; ~~~~~~~~~~~~~~~~~~~~~~
+; You can also configure channels in a separate chan_dahdi.conf section. In
+; this case the keyword 'channel' is not used. Instead the keyword
+; 'dahdichan' is used (as in users.conf) - configuration is only processed
+; in a section where the keyword dahdichan is used. It will only be
+; processed in the end of the section. Thus the following section:
+;
+;[phones]
+;echocancel = 64
+;dahdichan = 1-8
+;group = 1
+;
+; Is somewhat equivalent to the following snippet in the section
+; [channels]:
+;
+;echocancel = 64
+;group = 1
+;channel => 1-8
+;
+; When starting a new section almost all of the configuration values are
+; copied from their values at the end of the section [channels] in
+; chan_dahdi.conf and [general] in users.conf - one section's configuration
+; does not affect another one's.
+;
+; Instead of letting common configuration values "slide through" you can
+; use configuration templates to easily keep the common part in one
+; place and override where needed.
+;
+;[phones](!)
+;echocancel = yes
+;group = 0,4
+;callgroup = 3
+;pickupgroup = 3
+;threewaycalling = yes
+;transfer = yes
+;context = phones
+;faxdetect = incoming
+;
+;[phone-1](phones)
+;dahdichan = 1
+;callerid = My Name <501>
+;mailbox = 501@mailboxes
+;
+;
+;[fax](phones)
+;dahdichan = 2
+;faxdetect = no
+;context = fax
+;
+;[phone-3](phones)
+;dahdichan = 3
+;pickupgroup = 3,4
diff --git a/configs/samples/chan_mobile.conf.sample b/configs/samples/chan_mobile.conf.sample
new file mode 100644
index 000000000..c9c4d2d85
--- /dev/null
+++ b/configs/samples/chan_mobile.conf.sample
@@ -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
diff --git a/configs/samples/cli.conf.sample b/configs/samples/cli.conf.sample
new file mode 100644
index 000000000..0ddd92c99
--- /dev/null
+++ b/configs/samples/cli.conf.sample
@@ -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
diff --git a/configs/samples/cli_aliases.conf.sample b/configs/samples/cli_aliases.conf.sample
new file mode 100644
index 000000000..21353fdb4
--- /dev/null
+++ b/configs/samples/cli_aliases.conf.sample
@@ -0,0 +1,203 @@
+;
+; 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 = asterisk_1dot2	; Asterisk 1.2 style syntax
+;template = asterisk_1dot4	; 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
+pjsip reload=module reload res_pjsip.so
+
+; 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
+
+;[asterisk_1dot6](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
+
+[asterisk_1dot2_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
+
+[asterisk_1dot2](asterisk_1dot2_tpl)
+; add any additional custom commands you want below here, for example:
+;die quickly=stop now
+
+[asterisk_1dot4_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
+
+[asterisk_1dot4](asterisk_1dot4_tpl)
+; add any additional custom commands you want below here.
+
+[asterisk_11_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
+features reload=module reload features
+
+[asterisk_11](asterisk_11_tpl)
+; add any additional custom commands you want below here.
diff --git a/configs/samples/cli_permissions.conf.sample b/configs/samples/cli_permissions.conf.sample
new file mode 100644
index 000000000..4a6973f50
--- /dev/null
+++ b/configs/samples/cli_permissions.conf.sample
@@ -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
+;
+;
diff --git a/configs/samples/codecs.conf.sample b/configs/samples/codecs.conf.sample
new file mode 100644
index 000000000..9b5738577
--- /dev/null
+++ b/configs/samples/codecs.conf.sample
@@ -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
diff --git a/configs/samples/confbridge.conf.sample b/configs/samples/confbridge.conf.sample
new file mode 100644
index 000000000..860f1cb87
--- /dev/null
+++ b/configs/samples/confbridge.conf.sample
@@ -0,0 +1,373 @@
+[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.
+;announce_join_leave_review=yes ; When enabled, implies announce_join_leave, but the user
+                                ; will be prompted to review their recording before
+                                ; entering the conference. During this phase, the recording
+                                ; may be listened to, re-recorded, or accepted as is. 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
diff --git a/configs/samples/config_test.conf.sample b/configs/samples/config_test.conf.sample
new file mode 100644
index 000000000..2fff45ece
--- /dev/null
+++ b/configs/samples/config_test.conf.sample
@@ -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]
diff --git a/configs/samples/console.conf.sample b/configs/samples/console.conf.sample
new file mode 100644
index 000000000..606254eee
--- /dev/null
+++ b/configs/samples/console.conf.sample
@@ -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.
diff --git a/configs/samples/dbsep.conf.sample b/configs/samples/dbsep.conf.sample
new file mode 100644
index 000000000..9cfa23740
--- /dev/null
+++ b/configs/samples/dbsep.conf.sample
@@ -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
+
diff --git a/configs/samples/dnsmgr.conf.sample b/configs/samples/dnsmgr.conf.sample
new file mode 100644
index 000000000..e34dbcf0a
--- /dev/null
+++ b/configs/samples/dnsmgr.conf.sample
@@ -0,0 +1,5 @@
+[general]
+;enable=yes		; enable creation of managed DNS lookups
+			;   default is 'no'
+;refreshinterval=1200	; refresh managed DNS lookups every <n> seconds
+			;   default is 300 (5 minutes)
\ No newline at end of file
diff --git a/configs/samples/dsp.conf.sample b/configs/samples/dsp.conf.sample
new file mode 100644
index 000000000..08c5a5715
--- /dev/null
+++ b/configs/samples/dsp.conf.sample
@@ -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
+
diff --git a/configs/samples/dundi.conf.sample b/configs/samples/dundi.conf.sample
new file mode 100644
index 000000000..70f97d422
--- /dev/null
+++ b/configs/samples/dundi.conf.sample
@@ -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
+
+;[*]
+;
diff --git a/configs/samples/enum.conf.sample b/configs/samples/enum.conf.sample
new file mode 100644
index 000000000..39c723175
--- /dev/null
+++ b/configs/samples/enum.conf.sample
@@ -0,0 +1,22 @@
+;
+; ENUM Configuration for resolving phone numbers over DNS
+;
+; Sample config for Asterisk
+; This file is reloaded at "module reload enum" in the CLI
+;
+[general]
+;
+; The search list for domains may be customized.  Domains are searched
+; in the order they are listed here.
+;
+search => e164.arpa
+;
+; If you'd like to use the E.164.org public ENUM registry in addition
+; to the official e164.arpa one, uncomment the following line
+;
+;search => e164.org
+;
+; As there are more H323 drivers available you have to select to which
+; drive a H323 URI will map. Default is "H323".
+;
+h323driver => H323
diff --git a/configs/samples/extconfig.conf.sample b/configs/samples/extconfig.conf.sample
new file mode 100644
index 000000000..52a49cfbe
--- /dev/null
+++ b/configs/samples/extconfig.conf.sample
@@ -0,0 +1,108 @@
+;
+; 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
+;    sqlite3 ... res_config_sqlite3
+;    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)
+;ps_endpoints => odbc,asterisk
+;ps_auths => odbc,asterisk
+;ps_aors => odbc,asterisk
+;ps_domain_aliases => odbc,asterisk
+;ps_endpoint_id_ips => odbc,asterisk
+;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.
+
diff --git a/configs/samples/extensions.ael.sample b/configs/samples/extensions.ael.sample
new file mode 100644
index 000000000..495001f90
--- /dev/null
+++ b/configs/samples/extensions.ael.sample
@@ -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.
+//
+}
diff --git a/configs/samples/extensions.conf.sample b/configs/samples/extensions.conf.sample
new file mode 100644
index 000000000..df91223f0
--- /dev/null
+++ b/configs/samples/extensions.conf.sample
@@ -0,0 +1,857 @@
+; 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(${ARG1})
+exten => s,n(fail),Hangup
+
+
+[demo]
+include => stdexten
+;
+; We start with what to do when a call first comes in.
+;
+exten => s,1,Wait(1)			; Wait a second, just for fun
+exten => s,n,Answer			; Answer the line
+exten => s,n,Set(TIMEOUT(digit)=5)	; Set Digit Timeout to 5 seconds
+exten => s,n,Set(TIMEOUT(response)=10)	; Set Response Timeout to 10 seconds
+exten => s,n(restart),BackGround(demo-congrats)	; Play a congratulatory message
+exten => s,n(instruct),BackGround(demo-instruct)	; Play some instructions
+exten => s,n,WaitExten			; Wait for an extension to be dialed.
+
+exten => 2,1,BackGround(demo-moreinfo)	; Give some more information.
+exten => 2,n,Goto(s,instruct)
+
+exten => 3,1,Set(LANGUAGE()=fr)		; Set language to french
+exten => 3,n,Goto(s,restart)		; Start with the congratulations
+
+exten => 1000,1,Goto(default,s,1)
+;
+; We also create an example user, 1234, who is on the console and has
+; voicemail, etc.
+;
+exten => 1234,1,Playback(transfer,skip)		; "Please hold while..."
+					; (but skip if channel is not up)
+exten => 1234,n,Gosub(${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(${WIL}))
+						; Ditto for wil
+;exten => 6536,n,Goto(default,s,1)		; exited Voicemail
+;exten => wil,1,Goto(6236,1)
+
+;If you want to subscribe to the status of a parking space, this is
+;how you do it. Subscribe to extension 6600 in sip, and you will see
+;the status of the first parking lot with this extensions' help
+;exten => 6600,hint,park:701@parkedcalls
+;exten => 6600,1,noop
+;
+;To subscribe to the availability of a free member in the 'markq' queue.
+;Note: '_avail' is added to the QueueName
+;exten => 8501,hint,Queue:markq_avail
+;exten => 8501,1,Queue(markq)
+;
+; You can also monitor the status of a queue by providing a hint for a
+; particular queue name.
+;exten => 8502,hint,Queue:markq
+;exten => 8502,1,Queue(markq)
+;
+; Some other handy things are an extension for checking voicemail via
+; voicemailmain
+;
+;exten => 8500,1,VoicemailMain
+;exten => 8500,n,Hangup
+;
+; Or a conference room (you'll need to edit meetme.conf to enable this room)
+;
+;exten => 8600,1,Meetme(1234)
+;
+; Or playing an announcement to the called party, as soon it answers
+;
+;exten = 8700,1,Dial(${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.
diff --git a/configs/samples/extensions.lua.sample b/configs/samples/extensions.lua.sample
new file mode 100644
index 000000000..1b4428c7c
--- /dev/null
+++ b/configs/samples/extensions.lua.sample
@@ -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";
+	};
+}
+
diff --git a/configs/samples/extensions_minivm.conf.sample b/configs/samples/extensions_minivm.conf.sample
new file mode 100644
index 000000000..2f9d24637
--- /dev/null
+++ b/configs/samples/extensions_minivm.conf.sample
@@ -0,0 +1,159 @@
+; MINI-VOICEMAIL dialplan example
+; ---------------------------------------------------------------------------------------
+; ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+;
+;
+; This is an example on how to use the Mini-Voicemail system to build
+; voicemail systems.
+;
+;.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-
+; A macro to test the MINIVMACCOUNT dialplan function
+; 	Currently, accountcode and pincode is not used in the application
+; 	They where added to be used in dialplan scripting
+;
+;
+[macro-minivmfunctest]
+exten => s,1,set(account=${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))
+
diff --git a/configs/samples/features.conf.sample b/configs/samples/features.conf.sample
new file mode 100644
index 000000000..1724fc155
--- /dev/null
+++ b/configs/samples/features.conf.sample
@@ -0,0 +1,115 @@
+;
+; Sample Call Features (transfer, monitor/mixmonitor, etc) configuration
+;
+
+; Asterisk 12 Note - All parking lot configuration is now done in res_parking.conf
+
+[general]
+;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
+;pickupexten = *8               ; Configure the pickup extension. (default is *8)
+;pickupsound = beep             ; to indicate a successful pickup (default: no sound)
+;pickupfailsound = beeperr      ; to indicate that the pickup failed (default: no sound)
+;featuredigittimeout = 1000     ; Max time (ms) between digits for
+                                ; feature activation  (default is 1000 ms)
+;recordingfailsound = beeperr   ; indicates that a one-touch monitor or one-touch mixmonitor feature failed
+                                ; to be applied to the call. (default: no sound)
+;atxfernoanswertimeout = 15     ; Timeout for answer on attended transfer default is 15 seconds.
+;atxferdropcall = no            ; If someone does an attended transfer, then hangs up before the transfer
+                                ; target answers, then by default, the system will try to call back the
+                                ; person that did the transfer.  If this is set to "yes", the ringing
+                                ; transfer target is immediately transferred to the transferee.
+;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.
+
+
+; 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!
+;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!
+
+[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 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   -> ActivatedBy is no longer honored.  The feature is activated by which
+;                   channel DYNAMIC_FEATURES includes the feature is on.  Use predial
+;                   to set different values of DYNAMIC_FEATURES on the channels.
+;                   Historic values are: "caller", "callee", and "both".
+;  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 Goto,
+;   Background, WaitExten, and many more.  The exceptions to this are Gosub and
+;   Macro routines which must complete for the call to continue.
+;
+; 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].
diff --git a/configs/samples/festival.conf.sample b/configs/samples/festival.conf.sample
new file mode 100644
index 000000000..e91821719
--- /dev/null
+++ b/configs/samples/festival.conf.sample
@@ -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
+;
+;
+
diff --git a/configs/samples/followme.conf.sample b/configs/samples/followme.conf.sample
new file mode 100644
index 000000000..a233948bf
--- /dev/null
+++ b/configs/samples/followme.conf.sample
@@ -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.
+
+
diff --git a/configs/samples/func_odbc.conf.sample b/configs/samples/func_odbc.conf.sample
new file mode 100644
index 000000000..fd528d216
--- /dev/null
+++ b/configs/samples/func_odbc.conf.sample
@@ -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})}'
+
diff --git a/configs/samples/hep.conf.sample b/configs/samples/hep.conf.sample
new file mode 100644
index 000000000..40b17aa0e
--- /dev/null
+++ b/configs/samples/hep.conf.sample
@@ -0,0 +1,16 @@
+;
+; res_hep Module configuration for Asterisk
+;
+
+; All settings are currently set in the general section.
+[general]
+enabled = yes                      ; Enable/disable forwarding of packets to a
+                                   ; HEP server. Default is "yes".
+capture_address = 192.168.1.1:9061 ; The address of the HEP capture server.
+capture_password = foo             ; If specified, the authorization passsword
+                                   ; for the HEP server. If not specified, no
+                                   ; authorization password will be sent.
+capture_id = 1234                  ; A unique integer identifier for this
+                                   ; server. This ID will be embedded sent
+                                   ; with each packet from this server.
+
diff --git a/configs/samples/http.conf.sample b/configs/samples/http.conf.sample
new file mode 100644
index 000000000..44095a11e
--- /dev/null
+++ b/configs/samples/http.conf.sample
@@ -0,0 +1,96 @@
+;
+; 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
+;
+; session_keep_alive specifies the number of milliseconds to wait for
+; the next HTTP request over a persistent connection.
+;
+; Set to 0 to disable persistent HTTP connections.
+; Default: 15000
+;session_keep_alive=15000
+;
+; Whether Asterisk should serve static content from static-http
+; 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]
+;
+; NOTE: You need a valid HTTP AMI mansession_id cookie with the manager
+; config permission to POST files.
+;
+; 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/
+;
diff --git a/configs/samples/iax.conf.sample b/configs/samples/iax.conf.sample
new file mode 100644
index 000000000..e17c7dfeb
--- /dev/null
+++ b/configs/samples/iax.conf.sample
@@ -0,0 +1,672 @@
+;
+; 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) or for
+                         ; IPv6 the address needs to be in brackets then colon
+                         ; and port (e.g. bindaddr=[2001:db8::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. IPv6 addresses are accepted.
+
+;
+; 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 to the
+                                                ; target of the 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
+; Note: app_voicemail mailboxes must be in the form of mailbox@context.
+;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
+;
diff --git a/configs/samples/iaxprov.conf.sample b/configs/samples/iaxprov.conf.sample
new file mode 100644
index 000000000..668809c4d
--- /dev/null
+++ b/configs/samples/iaxprov.conf.sample
@@ -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
+
diff --git a/configs/samples/indications.conf.sample b/configs/samples/indications.conf.sample
new file mode 100644
index 000000000..7ac1ba936
--- /dev/null
+++ b/configs/samples/indications.conf.sample
@@ -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
diff --git a/configs/samples/logger.conf.sample b/configs/samples/logger.conf.sample
new file mode 100644
index 000000000..0fa7dcc8d
--- /dev/null
+++ b/configs/samples/logger.conf.sample
@@ -0,0 +1,134 @@
+;
+; 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
+;
+; When using realtime for the queue log, use GMT for the timestamp
+; instead of localtime.  The default of this option is 'no'.
+;queue_log_realtime_use_gmt = yes
+;
+; Log rotation strategy:
+; none:  Do not perform any logrotation at all.  You should make
+;        very sure to set up some external logrotate mechanism
+;        as the asterisk logs can get very large, very quickly.
+; 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
+;
diff --git a/configs/samples/manager.conf.sample b/configs/samples/manager.conf.sample
new file mode 100644
index 000000000..f31c6d397
--- /dev/null
+++ b/configs/samples/manager.conf.sample
@@ -0,0 +1,157 @@
+;
+; AMI - The Asterisk Manager Interface
+;
+; Third party application call management support and PBX event supervision
+;
+; This configuration file is read every time someone logs in
+;
+; 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: (PJ)?SIP/(james|jim|john)-
+;eventfilter=!Channel: DAHDI/
+; The eventfilter option is used to whitelist or blacklist events per user.
+; A filter consists of an (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.
+; security  - Security Events.  Read-only.
+; 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
+
diff --git a/configs/samples/meetme.conf.sample b/configs/samples/meetme.conf.sample
new file mode 100644
index 000000000..c40c4606e
--- /dev/null
+++ b/configs/samples/meetme.conf.sample
@@ -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
diff --git a/configs/samples/mgcp.conf.sample b/configs/samples/mgcp.conf.sample
new file mode 100644
index 000000000..7c725bc3d
--- /dev/null
+++ b/configs/samples/mgcp.conf.sample
@@ -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
diff --git a/configs/samples/minivm.conf.sample b/configs/samples/minivm.conf.sample
new file mode 100644
index 000000000..55a39c869
--- /dev/null
+++ b/configs/samples/minivm.conf.sample
@@ -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
diff --git a/configs/samples/misdn.conf.sample b/configs/samples/misdn.conf.sample
new file mode 100644
index 000000000..ac54dbc5a
--- /dev/null
+++ b/configs/samples/misdn.conf.sample
@@ -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
diff --git a/configs/samples/modules.conf.sample b/configs/samples/modules.conf.sample
new file mode 100644
index 000000000..38e33e1f4
--- /dev/null
+++ b/configs/samples/modules.conf.sample
@@ -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
+;
diff --git a/configs/samples/motif.conf.sample b/configs/samples/motif.conf.sample
new file mode 100644
index 000000000..9d5bc808c
--- /dev/null
+++ b/configs/samples/motif.conf.sample
@@ -0,0 +1,99 @@
+; 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
+
+; Additional options
+; callgroup
+; pickupgroup
+; language
+; musicclass
+; parkinglot
diff --git a/configs/samples/musiconhold.conf.sample b/configs/samples/musiconhold.conf.sample
new file mode 100644
index 000000000..1211c8a6d
--- /dev/null
+++ b/configs/samples/musiconhold.conf.sample
@@ -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
diff --git a/configs/samples/muted.conf.sample b/configs/samples/muted.conf.sample
new file mode 100644
index 000000000..15c7409d9
--- /dev/null
+++ b/configs/samples/muted.conf.sample
@@ -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
diff --git a/configs/samples/ooh323.conf.sample b/configs/samples/ooh323.conf.sample
new file mode 100644
index 000000000..aeeb02fab
--- /dev/null
+++ b/configs/samples/ooh323.conf.sample
@@ -0,0 +1,208 @@
+; ---------------------------------------------------------------------------------
+; ---  ******* 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 asterisk send back to caller own connected line id on incoming call as called number
+; Default - no
+aniasdni=no
+
+;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
+
+
+
+
+
diff --git a/configs/samples/osp.conf.sample b/configs/samples/osp.conf.sample
new file mode 100644
index 000000000..e72aa3259
--- /dev/null
+++ b/configs/samples/osp.conf.sample
@@ -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
diff --git a/configs/samples/oss.conf.sample b/configs/samples/oss.conf.sample
new file mode 100644
index 000000000..c3781a2a2
--- /dev/null
+++ b/configs/samples/oss.conf.sample
@@ -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)
+
diff --git a/configs/samples/phone.conf.sample b/configs/samples/phone.conf.sample
new file mode 100644
index 000000000..3d4a7c2dd
--- /dev/null
+++ b/configs/samples/phone.conf.sample
@@ -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
diff --git a/configs/samples/phoneprov.conf.sample b/configs/samples/phoneprov.conf.sample
new file mode 100644
index 000000000..17d8b1f49
--- /dev/null
+++ b/configs/samples/phoneprov.conf.sample
@@ -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
+
diff --git a/configs/samples/pjsip.conf.sample b/configs/samples/pjsip.conf.sample
new file mode 100644
index 000000000..3aa05a96b
--- /dev/null
+++ b/configs/samples/pjsip.conf.sample
@@ -0,0 +1,749 @@
+; PJSIP Configuration Samples and Quick Reference
+;
+; This file has several very basic configuration examples, to serve as a quick
+; reference to jog your memory when you need to write up a new configuration.
+; It is not intended to teach PJSIP configuration or serve as an exhaustive
+; reference of options and potential scenarios.
+;
+; This file has two main sections.
+; First, manually written examples to serve as a handy reference.
+; Second, a list of all possible PJSIP config options by section. This is
+; pulled from the XML config help. It only shows the synopsis for every item.
+; If you want to see more detail please check the documentation sources
+; mentioned at the top of this file.
+
+; Documentation
+;
+; The official documentation is at http://wiki.asterisk.org
+; You can read the XML configuration help via Asterisk command line with
+; "config show help res_pjsip", then you can drill down through the various
+; sections and their options.
+;
+
+;========!!!!!!!!!!!!!!!!!!!  SECURITY NOTICE  !!!!!!!!!!!!!!!!!!!!===========
+;
+; At a minimum please read the file "README-SERIOUSLY.bestpractices.txt",
+; located in the Asterisk source directory before starting Asterisk.
+; Otherwise you risk allowing the security of the Asterisk system to be
+; compromised. Beyond that please visit and read the security information on
+; the wiki at: https://wiki.asterisk.org/wiki/x/EwFB
+;
+; A few basics to pay attention to:
+;
+; Anonymous Calls
+;
+; By default anonymous inbound calls via PJSIP are not allowed. If you want to
+; route anonymous calls you'll need to define an endpoint named "anonymous".
+; res_pjsip_endpoint_identifier_anonymous.so handles that functionality so it
+; must be loaded. It is not recommended to accept anonymous calls.
+;
+; Access Control Lists
+;
+; See the example ACL configuration in this file. Read the configuration help
+; for the section and all of its options. Look over the samples in acl.conf
+; and documentation at https://wiki.asterisk.org/wiki/x/uA80AQ
+; If possible, restrict access to only networks and addresses you trust.
+;
+; Dialplan Contexts
+;
+; When defining configuration (such as an endpoint) that links into
+; dialplan configuration, be aware of what that dialplan does. It's easy to
+; accidentally provide access to internal or outbound dialing extensions which
+; could cost you severely. The "context=" line in endpoint configuration
+; determines which dialplan context inbound calls will enter into.
+;
+;=============================================================================
+
+; Overview of Configuration Section Types Used in the Examples
+;
+; * Transport "transport"
+;   * Configures res_pjsip transport layer interaction.
+; * Endpoint "endpoint"
+;   * Configures core SIP functionality related to SIP endpoints.
+; * Authentication "auth"
+;   * Stores inbound or outbound authentication credentials for use by trunks,
+;     endpoints, registrations.
+; * Address of Record "aor"
+;   * Stores contact information for use by endpoints.
+; * Endpoint Identification "identify"
+;   * Maps a host directly to an endpoint
+; * Access Control List "acl"
+;   * Defines a permission list or references one stored in acl.conf
+; * Registration "registration"
+;   * Contains information about an outbound SIP registration
+
+; The following sections show example configurations for various scenarios.
+; Most require a couple or more configuration types configured in concert.
+
+;=============================================================================
+
+; Naming of Configuration Sections
+;
+; Configuration section names are denoted with enclosing brackets,
+; e.g. [6001]
+; In most cases, you can name a section whatever makes sense to you. For example
+; you might name a transport [transport-udp-nat] to help you remember how that
+; section is being used. However, in some cases, ("endpoint" and "aor" types)
+; the section name has a relationship to its function.
+;
+; Depending on the modules loaded, Asterisk can match SIP requests to an
+; endpoint or aor in a few ways:
+;
+; 1) Match a section name for endpoint type sections to the username in the
+;    "From" header of inbound SIP requests.
+; 2) Match a section name for aor type sections to the username in the "To"
+;    header of inbound SIP REGISTER requests.
+; 3) With an identify type section configured, match an inbound SIP request of
+;    any type to an endpoint or aor based on the IP source address of the
+;    request.
+;
+; Note that sections can have the same name as long as their "type" options are
+; set to different values. In most cases it makes sense to have associated
+; configuration sections use the same name, as you'll see in the examples within
+; this file.
+
+;===============EXAMPLE TRANSPORTS============================================
+;
+; A few examples for potential transport options.
+;
+; For the NAT transport example, be aware that the options starting with
+; the prefix "external_" will only apply to communication with addresses
+; outside the range set with "local_net=".
+;
+; IPv6: For endpoints using IPv6, remember to set "rtp_ipv6=yes" so that the RTP
+; engine will also be able to bind to an IPv6 address.
+;
+; You can have more than one of any type of transport, as long as it doesn't
+; use the same resources (bind address, port, etc) as the others.
+
+; Basic UDP transport
+;
+;[transport-udp]
+;type=transport
+;protocol=udp    ;udp,tcp,tls,ws,wss
+;bind=0.0.0.0
+
+; UDP transport behind NAT
+;
+;[transport-udp-nat]
+;type=transport
+;protocol=udp
+;bind=0.0.0.0
+;local_net=192.0.2.0/24
+;external_media_address=203.0.113.1
+;external_signaling_address=203.0.113.1
+
+; Basic IPv6 UDP transport
+;
+;[transport-udp-ipv6]
+;type=transport
+;protocol=udp
+;bind=::
+
+; Example IPv4 TLS transport
+;
+;[transport-tls]
+;type=transport
+;protocol=tls
+;bind=0.0.0.0
+;cert_file=/path/mycert.crt
+;priv_key_file=/path/mykey.key
+;cipher=ALL
+;method=tlsv1
+
+
+;===============OUTBOUND REGISTRATION WITH OUTBOUND AUTHENTICATION============
+;
+; This is a simple registration that works with some SIP trunking providers.
+; You'll need to set up the auth example "mytrunk_auth" below to enable outbound
+; authentication. Note that we "outbound_auth=" use for outbound authentication
+; instead of "auth=", which is for inbound authentication.
+;
+; If you are registering to a server from behind NAT, be sure you assign a transport
+; that is appropriately configured with NAT related settings. See the NAT transport example.
+;
+; "contact_user=" sets the SIP contact header's user portion of the SIP URI
+; this will affect the extension reached in dialplan when the far end calls you at this
+; registration. The default is 's'.
+
+;[mytrunk]
+;type=registration
+;transport=transport-udp
+;outbound_auth=mytrunk_auth
+;server_uri=sip:sip.example.com
+;client_uri=sip:1234567890@sip.example.com
+;contact_user=1234567890
+;retry_interval=60
+;forbidden_retry_interval=600
+;expiration=3600
+
+;[mytrunk_auth]
+;type=auth
+;auth_type=userpass
+;password=1234567890
+;username=1234567890
+;realm=sip.example.com
+
+;===============ENDPOINT CONFIGURED AS A TRUNK, OUTBOUND AUTHENTICATION=======
+;
+; This is one way to configure an endpoint as a trunk. It is set up with
+; "outbound_auth=" to enable authentication when dialing out through this
+; endpoint. There is no inbound authentication set up since a provider will
+; not normally authenticate when calling you.
+;
+; The identify configuration enables IP address matching against this endpoint.
+; For calls from a trunking provider, the From user may be different every time,
+; so we want to match against IP address instead of From user.
+;
+; If you want the provider of your trunk to know where to send your calls
+; you'll need to use an outbound registration as in the example above this
+; section.
+;
+; NAT
+;
+; At a basic level configure the endpoint with a transport that is set up
+; with the appropriate NAT settings. There may be some additional settings you
+; need here based on your NAT/Firewall scenario. Look to the CLI config help
+; "config show help res_pjsip endpoint" or on the wiki for other NAT related
+; options and configuration. We've included a few below.
+;
+; AOR
+;
+; Endpoints use one or more AOR sections to store their contact details.
+; You can define multiple contact addresses in SIP URI format in multiple
+; "contact=" entries.
+;
+
+;[mytrunk]
+;type=endpoint
+;transport=transport-udp
+;context=from-external
+;disallow=all
+;allow=ulaw
+;outbound_auth=mytrunk_auth
+;aors=mytrunk
+;                   ;A few NAT relevant options that may come in handy.
+;force_rport=yes    ;It's a good idea to read the configuration help for each
+;direct_media=no    ;of these options.
+;ice_support=yes
+
+;[mytrunk]
+;type=aor
+;contact=sip:198.51.100.1:5060
+;contact=sip:198.51.100.2:5060
+
+;[mytrunk]
+;type=identify
+;endpoint=mytrunk
+;match=198.51.100.1
+;match=198.51.100.2
+
+
+;=============ENDPOINT CONFIGURED AS A TRUNK, INBOUND AUTH AND REGISTRATION===
+;
+; Here we are allowing a remote device to register to Asterisk and requiring
+; that they authenticate for registration and calls.
+; You'll note that this configuration is essentially the same as configuring
+; an endpoint for use with a SIP phone.
+
+
+;[7000]
+;type=endpoint
+;context=from-external
+;disallow=all
+;allow=ulaw
+;transport=transport-udp
+;auth=7000
+;aors=7000
+
+;[7000]
+;type=auth
+;auth_type=userpass
+;password=7000
+;username=7000
+
+;[7000]
+;type=aor
+;max_contacts=1
+
+
+;===============ENDPOINT CONFIGURED FOR USE WITH A SIP PHONE==================
+;
+; This example includes the endpoint, auth and aor configurations. It
+; requires inbound authentication and allows registration, as well as references
+; a transport that you'll need to uncomment from the previous examples.
+;
+; Uncomment one of the transport lines to choose which transport you want. If
+; not specified then the default transport chosen is the first defined transport
+; in the configuration file.
+;
+; Modify the "max_contacts=" line to change how many unique registrations to allow.
+;
+; Use the "contact=" line instead of max_contacts= if you want to statically
+; define the location of the device.
+;
+; If using the TLS enabled transport, you may want the "media_encryption=sdes"
+; option to additionally enable SRTP, though they are not mutually inclusive.
+;
+; Use the "rtp_ipv6=yes" option if you want to utilize RTP over an ipv6 transport.
+;
+; If this endpoint were remote, and it was using a transport configured for NAT
+; then you likely want to use "direct_media=no" to prevent audio issues.
+
+
+;[6001]
+;type=endpoint
+;transport=transport-udp
+;context=from-internal
+;disallow=all
+;allow=ulaw
+;allow=gsm
+;auth=6001
+;aors=6001
+;
+; A few more transports to pick from, and some related options below them.
+;
+;transport=transport-tls
+;media_encryption=sdes
+;transport=transport-udp-ipv6
+;rtp_ipv6=yes
+;transport=transport-udp-nat
+;direct_media=no
+;
+; MWI related options
+
+;aggregate_mwi=yes
+;mailboxes=6001@default,7001@default
+;mwi_from_user=6001
+;
+; Extension and Device state options
+;
+;device_state_busy_at=1
+;allow_subscribe=yes
+;sub_min_expiry=30
+
+;[6001]
+;type=auth
+;auth_type=userpass
+;password=6001
+;username=6001
+
+;[6001]
+;type=aor
+;max_contacts=1
+;contact=sip:6001@192.0.2.1:5060
+
+;===============ENDPOINT BEHIND NAT OR FIREWALL===============================
+;
+; This example assumes your transport is configured with a public IP and the
+; endpoint itself is behind NAT and maybe a firewall, rather than having
+; Asterisk behind NAT. For the sake of simplicity, we'll assume a typical
+; VOIP phone. The most important settings to configure are:
+;
+;  * direct_media, to ensure Asterisk stays in the media path
+;  * rtp_symmetric and force_rport options to help the far-end NAT/firewall
+;
+; Depending on the settings of your remote SIP device or NAT/firewall device
+; you may have to experiment with a combination of these settings.
+;
+; If both Asterisk and the remote phones are a behind NAT/firewall then you'll
+; have to make sure to use a transport with appropriate settings (as in the
+; transport-udp-nat example).
+;
+;[6002]
+;type=endpoint
+;transport=transport-udp
+;context=from-internal
+;disallow=all
+;allow=ulaw
+;auth=6002
+;aors=6002
+;direct_media=no
+;rtp_symmetric=yes
+;force_rport=yes
+;ice_support=yes   ;This is specific to clients that support NAT traversal
+                   ;for media via ICE,STUN,TURN. See the wiki at:
+                   ;https://wiki.asterisk.org/wiki/x/D4FHAQ
+                   ;for a deeper explanation of this topic.
+
+;[6002]
+;type=auth
+;auth_type=userpass
+;password=6002
+;username=6002
+
+;[6002]
+;type=aor
+;max_contacts=2
+
+
+;============EXAMPLE ACL CONFIGURATION==========================================
+;
+; The ACL or Access Control List section defines a set of permissions to permit
+; or deny access to various address or addresses. Alternatively it references an
+; ACL configuration already set in acl.conf.
+;
+; The ACL configuration is independent of individual endpoint configuration and
+; operates on all inbound SIP communication using res_pjsip.
+
+; Reference an ACL defined in acl.conf.
+;
+;[acl]
+;type=acl
+;acl=example_named_acl1
+
+; Reference a contactacl specifically.
+;
+;[acl]
+;type=acl
+;contact_acl=example_contact_acl1
+
+; Define your own ACL here in pjsip.conf and
+; permit or deny by IP address or range.
+;
+;[acl]
+;type=acl
+;deny=0.0.0.0/0.0.0.0
+;permit=209.16.236.0/24
+;deny=209.16.236.1
+
+; Restrict based on Contact Headers rather than IP.
+; Define options multiple times for various addresses or use a comma-delimited string.
+;
+;[acl]
+;type=acl
+;contact_deny=0.0.0.0/0.0.0.0
+;contact_permit=209.16.236.0/24
+;contact_permit=209.16.236.1
+;contact_permit=209.16.236.2,209.16.236.3
+
+; Restrict based on Contact Headers rather than IP and use
+; advanced syntax. Note the bang symbol used for "NOT", so we can deny
+; 209.16.236.12/32 within the permit= statement.
+;
+;[acl]
+;type=acl
+;contact_deny=0.0.0.0/0.0.0.0
+;contact_permit=209.16.236.0
+;permit=209.16.236.0/24, !209.16.236.12/32
+
+
+
+; MODULE PROVIDING BELOW SECTION(S): res_pjsip
+;==========================ENDPOINT SECTION OPTIONS=========================
+;[endpoint]
+;  SYNOPSIS: Endpoint
+;100rel=yes     ; Allow support for RFC3262 provisional ACK tags (default:
+                ; "yes")
+;aggregate_mwi=yes      ;  (default: "yes")
+;allow= ; Media Codec s to allow (default: "")
+;aors=  ; AoR s to be used with the endpoint (default: "")
+;auth=  ; Authentication Object s associated with the endpoint (default: "")
+;callerid=      ; CallerID information for the endpoint (default: "")
+;callerid_privacy=      ; Default privacy level (default: "")
+;callerid_tag=  ; Internal id_tag for the endpoint (default: "")
+;context=default        ; Dialplan context for inbound sessions (default:
+                        ; "default")
+;direct_media_glare_mitigation=none     ; Mitigation of direct media re INVITE
+                                        ; glare (default: "none")
+;direct_media_method=invite     ; Direct Media method type (default: "invite")
+;connected_line_method=invite   ; Connected line method type (default:
+                                ; "invite")
+;direct_media=yes       ; Determines whether media may flow directly between
+                        ; endpoints (default: "yes")
+;disable_direct_media_on_nat=no ; Disable direct media session refreshes when
+                                ; NAT obstructs the media session (default:
+                                ; "no")
+;disallow=      ; Media Codec s to disallow (default: "")
+;dtmf_mode=rfc4733      ; DTMF mode (default: "rfc4733")
+;media_address=         ; IP address used in SDP for media handling (default: "")
+;force_rport=yes        ; Force use of return port (default: "yes")
+;ice_support=no ; Enable the ICE mechanism to help traverse NAT (default: "no")
+;identify_by=username   ; Way s for Endpoint to be identified (default:
+                        ; "username")
+;redirect_method=user   ; How redirects received from an endpoint are handled
+                        ; (default: "user")
+;mailboxes=     ; Mailbox es to be associated with (default: "")
+;moh_suggest=default    ; Default Music On Hold class (default: "default")
+;outbound_auth= ; Authentication object used for outbound requests (default:
+                ; "")
+;outbound_proxy=        ; Proxy through which to send requests a full SIP URI
+                        ; must be provided (default: "")
+;rewrite_contact=no     ; Allow Contact header to be rewritten with the source
+                        ; IP address port (default: "no")
+;rtp_ipv6=no    ; Allow use of IPv6 for RTP traffic (default: "no")
+;rtp_symmetric=no       ; Enforce that RTP must be symmetric (default: "no")
+;send_diversion=yes     ; Send the Diversion header conveying the diversion
+                        ; information to the called user agent (default: "yes")
+;send_pai=no    ; Send the P Asserted Identity header (default: "no")
+;send_rpid=no   ; Send the Remote Party ID header (default: "no")
+;timers_min_se=90       ; Minimum session timers expiration period (default:
+                        ; "90")
+;timers=yes     ; Session timers for SIP packets (default: "yes")
+;timers_sess_expires=1800       ; Maximum session timer expiration period
+                                ; (default: "1800")
+;transport=     ; Desired transport configuration (default: "")
+;trust_id_inbound=no    ; Accept identification information received from this
+                        ; endpoint (default: "no")
+;trust_id_outbound=no   ; Send private identification details to the endpoint
+                        ; (default: "no")
+;type=  ; Must be of type endpoint (default: "")
+;use_ptime=no   ; Use Endpoint s requested packetisation interval (default:
+                ; "no")
+;use_avpf=no    ; Determines whether res_pjsip will use and enforce usage of
+                ; AVPF for this endpoint (default: "no")
+;media_encryption=no    ; Determines whether res_pjsip will use and enforce
+                        ; usage of media encryption for this endpoint (default:
+                        ; "no")
+;inband_progress=no     ; Determines whether chan_pjsip will indicate ringing
+                        ; using inband progress (default: "no")
+;call_group=    ; The numeric pickup groups for a channel (default: "")
+;pickup_group=  ; The numeric pickup groups that a channel can pickup (default:
+                ; "")
+;named_call_group=      ; The named pickup groups for a channel (default: "")
+;named_pickup_group=    ; The named pickup groups that a channel can pickup
+                        ; (default: "")
+;device_state_busy_at=0 ; The number of in use channels which will cause busy
+                        ; to be returned as device state (default: "0")
+;t38_udptl=no   ; Whether T 38 UDPTL support is enabled or not (default: "no")
+;t38_udptl_ec=none      ; T 38 UDPTL error correction method (default: "none")
+;t38_udptl_maxdatagram=0        ; T 38 UDPTL maximum datagram size (default:
+                                ; "0")
+;fax_detect=no  ; Whether CNG tone detection is enabled (default: "no")
+;t38_udptl_nat=no       ; Whether NAT support is enabled on UDPTL sessions
+                        ; (default: "no")
+;t38_udptl_ipv6=no      ; Whether IPv6 is used for UDPTL Sessions (default:
+                        ; "no")
+;tone_zone=     ; Set which country s indications to use for channels created
+                ; for this endpoint (default: "")
+;language=      ; Set the default language to use for channels created for this
+                ; endpoint (default: "")
+;one_touch_recording=no ; Determines whether one touch recording is allowed for
+                        ; this endpoint (default: "no")
+;record_on_feature=automixmon   ; The feature to enact when one touch recording
+                                ; is turned on (default: "automixmon")
+;record_off_feature=automixmon  ; The feature to enact when one touch recording
+                                ; is turned off (default: "automixmon")
+;rtp_engine=asterisk    ; Name of the RTP engine to use for channels created
+                        ; for this endpoint (default: "asterisk")
+;allow_transfer=yes     ; Determines whether SIP REFER transfers are allowed
+                        ; for this endpoint (default: "yes")
+;sdp_owner=-    ; String placed as the username portion of an SDP origin o line
+                ; (default: "-")
+;sdp_session=Asterisk   ; String used for the SDP session s line (default:
+                        ; "Asterisk")
+;tos_audio=0    ; DSCP TOS bits for audio streams (default: "0")
+;tos_video=0    ; DSCP TOS bits for video streams (default: "0")
+;cos_audio=0    ; Priority for audio streams (default: "0")
+;cos_video=0    ; Priority for video streams (default: "0")
+;allow_subscribe=yes    ; Determines if endpoint is allowed to initiate
+                        ; subscriptions with Asterisk (default: "yes")
+;sub_min_expiry=0       ; The minimum allowed expiry time for subscriptions
+                        ; initiated by the endpoint (default: "0")
+;from_user=     ; Username to use in From header for requests to this endpoint
+                ; (default: "")
+;mwi_from_user= ; Username to use in From header for unsolicited MWI NOTIFYs to
+                ; this endpoint (default: "")
+;from_domain=   ; Domain to user in From header for requests to this endpoint
+                ; (default: "")
+;dtls_verify=   ; Verify that the provided peer certificate is valid (default:
+                ; "")
+;dtls_rekey=    ; Interval at which to renegotiate the TLS session and rekey
+                ; the SRTP session (default: "")
+;dtls_cert_file=        ; Path to certificate file to present to peer (default:
+                        ; "")
+;dtls_private_key=      ; Path to private key for certificate file (default:
+                        ; "")
+;dtls_cipher=   ; Cipher to use for DTLS negotiation (default: "")
+;dtls_ca_file=  ; Path to certificate authority certificate (default: "")
+;dtls_ca_path=  ; Path to a directory containing certificate authority
+                ; certificates (default: "")
+;dtls_setup=    ; Whether we are willing to accept connections connect to the
+                ; other party or both (default: "")
+;srtp_tag_32=no ; Determines whether 32 byte tags should be used instead of 80
+                ; byte tags (default: "no")
+;set_var=       ; Variable set on a channel involving the endpoint. For multiple
+		; channel variables specify multiple 'set_var'(s)
+
+;==========================AUTH SECTION OPTIONS=========================
+;[auth]
+;  SYNOPSIS: Authentication type
+;auth_type=userpass     ; Authentication type (default: "userpass")
+;nonce_lifetime=32      ; Lifetime of a nonce associated with this
+                        ; authentication config (default: "32")
+;md5_cred=      ; MD5 Hash used for authentication (default: "")
+;password=      ; PlainText password used for authentication (default: "")
+;realm= ; SIP realm for endpoint (default: "")
+;type=  ; Must be auth (default: "")
+;username=      ; Username to use for account (default: "")
+
+
+;==========================DOMAIN_ALIAS SECTION OPTIONS=========================
+;[domain_alias]
+;  SYNOPSIS: Domain Alias
+;type=  ; Must be of type domain_alias (default: "")
+;domain=        ; Domain to be aliased (default: "")
+
+
+;==========================TRANSPORT SECTION OPTIONS=========================
+;[transport]
+;  SYNOPSIS: SIP Transport
+;async_operations=1     ; Number of simultaneous Asynchronous Operations
+                        ; (default: "1")
+;bind=  ; IP Address and optional port to bind to for this transport (default:
+        ; "")
+;ca_list_file=  ; File containing a list of certificates to read TLS ONLY
+                ; (default: "")
+;cert_file=     ; Certificate file for endpoint TLS ONLY (default: "")
+;cipher=        ; Preferred Cryptography Cipher TLS ONLY (default: "")
+;domain=        ; Domain the transport comes from (default: "")
+;external_media_address=        ; External IP address to use in RTP handling
+                                ; (default: "")
+;external_signaling_address=    ; External address for SIP signalling (default:
+                                ; "")
+;external_signaling_port=0      ; External port for SIP signalling (default:
+                                ; "0")
+;method=        ; Method of SSL transport TLS ONLY (default: "")
+;local_net=     ; Network to consider local used for NAT purposes (default: "")
+;password=      ; Password required for transport (default: "")
+;priv_key_file= ; Private key file TLS ONLY (default: "")
+;protocol=udp   ; Protocol to use for SIP traffic (default: "udp")
+;require_client_cert=   ; Require client certificate TLS ONLY (default: "")
+;type=  ; Must be of type transport (default: "")
+;verify_client= ; Require verification of client certificate TLS ONLY (default:
+                ; "")
+;verify_server= ; Require verification of server certificate TLS ONLY (default:
+                ; "")
+;tos=0  ; Enable TOS for the signalling sent over this transport (default: "0")
+;cos=0  ; Enable COS for the signalling sent over this transport (default: "0")
+;websocket_write_timeout=100    ; Default write timeout to set on websocket
+                                ; transports. This value may need to be adjusted
+                                ; for connections where Asterisk must write a
+                                ; substantial amount of data and the receiving
+                                ; clients are slow to process the received
+                                ; information. Value is in milliseconds; default
+                                ; is 100 ms.
+
+;==========================CONTACT SECTION OPTIONS=========================
+;[contact]
+;  SYNOPSIS: A way of creating an aliased name to a SIP URI
+;type=  ; Must be of type contact (default: "")
+;uri=   ; SIP URI to contact peer (default: "")
+;expiration_time=       ; Time to keep alive a contact (default: "")
+;qualify_frequency=0    ; Interval at which to qualify a contact (default: "0")
+;outbound_proxy=        ; Outbound proxy used when sending OPTIONS request
+                        ; (default: "")
+
+
+;==========================AOR SECTION OPTIONS=========================
+;[aor]
+;  SYNOPSIS: The configuration for a location of an endpoint
+;contact=       ; Permanent contacts assigned to AoR (default: "")
+;default_expiration=3600        ; Default expiration time in seconds for
+                                ; contacts that are dynamically bound to an AoR
+                                ; (default: "3600")
+;mailboxes=     ; Mailbox es to be associated with (default: "")
+;maximum_expiration=7200        ; Maximum time to keep an AoR (default: "7200")
+;max_contacts=0 ; Maximum number of contacts that can bind to an AoR (default:
+                ; "0")
+;minimum_expiration=60  ; Minimum keep alive time for an AoR (default: "60")
+;remove_existing=no     ; Determines whether new contacts replace existing ones
+                        ; (default: "no")
+;type=  ; Must be of type aor (default: "")
+;qualify_frequency=0    ; Interval at which to qualify an AoR (default: "0")
+;authenticate_qualify=no        ; Authenticates a qualify request if needed
+                                ; (default: "no")
+;outbound_proxy=        ; Outbound proxy used when sending OPTIONS request
+                        ; (default: "")
+
+
+;==========================SYSTEM SECTION OPTIONS=========================
+;[system]
+;  SYNOPSIS: Options that apply to the SIP stack as well as other system-wide settings
+;timer_t1=500   ; Set transaction timer T1 value milliseconds (default: "500")
+;timer_b=32000  ; Set transaction timer B value milliseconds (default: "32000")
+;compact_headers=no     ; Use the short forms of common SIP header names
+                        ; (default: "no")
+;threadpool_initial_size=0      ; Initial number of threads in the res_pjsip
+                                ; threadpool (default: "0")
+;threadpool_auto_increment=5    ; The amount by which the number of threads is
+                                ; incremented when necessary (default: "5")
+;threadpool_idle_timeout=60     ; Number of seconds before an idle thread
+                                ; should be disposed of (default: "60")
+;threadpool_max_size=0  ; Maximum number of threads in the res_pjsip threadpool
+                        ; A value of 0 indicates no maximum (default: "0")
+;type=  ; Must be of type system (default: "")
+
+;==========================GLOBAL SECTION OPTIONS=========================
+;[global]
+;  SYNOPSIS: Options that apply globally to all SIP communications
+;max_forwards=70        ; Value used in Max Forwards header for SIP requests
+                        ; (default: "70")
+;type=  ; Must be of type global (default: "")
+;user_agent=Asterisk PBX SVN-branch-12-r404375  ; Value used in User Agent
+                                                ; header for SIP requests and
+                                                ; Server header for SIP
+                                                ; responses (default: "Asterisk
+                                                ; PBX SVN-branch-12-r404375")
+;default_outbound_endpoint=default_outbound_endpoint    ; Endpoint to use when
+                                                        ; sending an outbound
+                                                        ; request to a URI
+                                                        ; without a specified
+                                                        ; endpoint (default: "d
+                                                        ; efault_outbound_endpo
+                                                        ; int")
+;debug=no ; Enable/Disable SIP debug logging.  Valid options include yes|no
+          ; or a host address (default: "no")
+
+
+; MODULE PROVIDING BELOW SECTION(S): res_pjsip_acl
+;==========================ACL SECTION OPTIONS=========================
+;[acl]
+;  SYNOPSIS: Access Control List
+;acl=   ; List of IP ACL section names in acl conf (default: "")
+;contact_acl=   ; List of Contact ACL section names in acl conf (default: "")
+;contact_deny=  ; List of Contact header addresses to deny (default: "")
+;contact_permit=        ; List of Contact header addresses to permit (default:
+                        ; "")
+;deny=  ; List of IP addresses to deny access from (default: "")
+;permit=        ; List of IP addresses to permit access from (default: "")
+;type=  ; Must be of type acl (default: "")
+
+
+
+
+; MODULE PROVIDING BELOW SECTION(S): res_pjsip_outbound_registration
+;==========================REGISTRATION SECTION OPTIONS=========================
+;[registration]
+;  SYNOPSIS: The configuration for outbound registration
+;auth_rejection_permanent=yes   ; Determines whether failed authentication
+                                ; challenges are treated as permanent failures
+                                ; (default: "yes")
+;client_uri=    ; Client SIP URI used when attemping outbound registration
+                ; (default: "")
+;contact_user=  ; Contact User to use in request (default: "")
+;expiration=3600        ; Expiration time for registrations in seconds
+                        ; (default: "3600")
+;max_retries=10 ; Maximum number of registration attempts (default: "10")
+;outbound_auth= ; Authentication object to be used for outbound registrations
+                ; (default: "")
+;outbound_proxy=        ; Outbound Proxy used to send registrations (default:
+                        ; "")
+;retry_interval=60      ; Interval in seconds between retries if outbound
+                        ; registration is unsuccessful (default: "60")
+;forbidden_retry_interval=0     ; Interval used when receiving a 403 Forbidden
+                                ; response (default: "0")
+;server_uri=    ; SIP URI of the server to register against (default: "")
+;transport=     ; Transport used for outbound authentication (default: "")
+;type=  ; Must be of type registration (default: "")
+
+
+
+
+; MODULE PROVIDING BELOW SECTION(S): res_pjsip_endpoint_identifier_ip
+;==========================IDENTIFY SECTION OPTIONS=========================
+;[identify]
+;  SYNOPSIS: Identifies endpoints via source IP address
+;endpoint=      ; Name of Endpoint (default: "")
+;match= ; IP addresses or networks to match against (default: "")
+;type=  ; Must be of type identify (default: "")
diff --git a/configs/samples/pjsip_notify.conf.sample b/configs/samples/pjsip_notify.conf.sample
new file mode 100644
index 000000000..8224ee1ff
--- /dev/null
+++ b/configs/samples/pjsip_notify.conf.sample
@@ -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
diff --git a/configs/samples/queuerules.conf.sample b/configs/samples/queuerules.conf.sample
new file mode 100644
index 000000000..fb2a1ba87
--- /dev/null
+++ b/configs/samples/queuerules.conf.sample
@@ -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
+
diff --git a/configs/samples/queues.conf.sample b/configs/samples/queues.conf.sample
new file mode 100644
index 000000000..bea62e83e
--- /dev/null
+++ b/configs/samples/queues.conf.sample
@@ -0,0 +1,549 @@
+[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
+;
+; If set to yes, just prior to the caller being bridged with a queue member
+; the following variables will be set
+; MEMBERINTERFACE is the interface name (eg. Agent/1234)
+; MEMBERNAME is the member name (eg. Joe Soap)
+; MEMBERCALLS is the number of calls that interface has taken,
+; MEMBERLASTCALL is the last time the member took a call.
+; MEMBERPENALTY is the penalty of the member
+; 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:
+; 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
+; 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 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 => Local/1001@agents,0,May Flowers,Agent:1001
+;member => Local/1002@agents,0,John Doe,Agent:1002
+;member => Local/1000@default,0,John Smith,SIP/1000
+;member => Local/2000@default,0,Lorem Ipsum,SIP/2000,no
diff --git a/configs/samples/res_config_mysql.conf.sample b/configs/samples/res_config_mysql.conf.sample
new file mode 100644
index 000000000..ab8ebf095
--- /dev/null
+++ b/configs/samples/res_config_mysql.conf.sample
@@ -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
diff --git a/configs/samples/res_config_sqlite.conf.sample b/configs/samples/res_config_sqlite.conf.sample
new file mode 100644
index 000000000..04e6ae2e7
--- /dev/null
+++ b/configs/samples/res_config_sqlite.conf.sample
@@ -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
diff --git a/configs/samples/res_config_sqlite3.conf.sample b/configs/samples/res_config_sqlite3.conf.sample
new file mode 100644
index 000000000..5416c809e
--- /dev/null
+++ b/configs/samples/res_config_sqlite3.conf.sample
@@ -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
+
diff --git a/configs/samples/res_corosync.conf.sample b/configs/samples/res_corosync.conf.sample
new file mode 100644
index 000000000..9a72c1ccd
--- /dev/null
+++ b/configs/samples/res_corosync.conf.sample
@@ -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
+;
diff --git a/configs/samples/res_curl.conf.sample b/configs/samples/res_curl.conf.sample
new file mode 100644
index 000000000..cc472989f
--- /dev/null
+++ b/configs/samples/res_curl.conf.sample
@@ -0,0 +1,8 @@
+;
+; CURLOPT global settings (mostly set for realtime)
+;
+[globals]
+;proxy=myproxy.example.com
+proxytype=http
+proxyport=8001
+;proxyuserpwd=asterisk:asteriskrocks
diff --git a/configs/samples/res_fax.conf.sample b/configs/samples/res_fax.conf.sample
new file mode 100644
index 000000000..dfaa4ce93
--- /dev/null
+++ b/configs/samples/res_fax.conf.sample
@@ -0,0 +1,28 @@
+; 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
diff --git a/configs/samples/res_ldap.conf.sample b/configs/samples/res_ldap.conf.sample
new file mode 100644
index 000000000..ac345cdd4
--- /dev/null
+++ b/configs/samples/res_ldap.conf.sample
@@ -0,0 +1,199 @@
+;
+; 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
+supportpath = AstAccountPathSupport
+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)
+
diff --git a/configs/samples/res_odbc.conf.sample b/configs/samples/res_odbc.conf.sample
new file mode 100644
index 000000000..66659ae42
--- /dev/null
+++ b/configs/samples/res_odbc.conf.sample
@@ -0,0 +1,121 @@
+;;; 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
+;
+; 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.
+;
diff --git a/configs/samples/res_parking.conf.sample b/configs/samples/res_parking.conf.sample
new file mode 100644
index 000000000..d24be2039
--- /dev/null
+++ b/configs/samples/res_parking.conf.sample
@@ -0,0 +1,121 @@
+[general]
+;parkeddynamic = yes            ; Enables dynamically created parkinglots. (default is no)
+
+; A parking lot named 'default' will automatically be used when no other
+; named parking lot is indicated for use by the park application or a
+; channel's parkinglot function and PARKINGLOT channel variable. This parking
+; lot is guaranteed to exist and will be created even if default is left out of
+; the configuration file.
+
+[default]                       ; Default Parking Lot
+parkext => 700                  ; What extension to dial to park. (optional; if
+                                ; specified, extensions will be created for parkext and
+                                ; the whole range of parkpos)
+
+;parkext_exclusive=yes          ; Specify that the parkext created for this parking lot
+                                ; will only access this parking lot. (default is no)
+
+parkpos => 701-720              ; What range of parking spaces to use - must be numeric
+                                ; Creates these spaces as extensions if parkext is set.
+                                ; Since this value is interpreted numerically, leading 0's
+                                ; will be ignored (so expect 00700-00720 to map to 700-720)
+
+context => parkedcalls          ; Which context parked calls and the default park
+
+;parkinghints = no              ; Add hints priorities automatically for parkpos
+                                ; extensions if parkext is set
+
+;parkingtime => 45              ; Number of seconds a call can be parked before returning
+
+;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 the next priority (usually none
+                                ;         unless you deliberately set up a catch-all second priority in the park-call
+                                ;         context) in the dialplan for extension matching the peer name (same as how
+                                ;         peer names are flattened into extensions when comebacktoorigin is 'no').
+                                ;
+                                ; '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 - dial string to call 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'.
+
+;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)
+
+;parkedcalltransfers = caller   ; Enables or disables DTMF based transfers when picking up a parked call.
+                                ; one of: callee, caller, both, no (default is no)
+
+;parkedcallreparking = caller   ; Enables or disables DTMF based parking when picking up a parked call.
+                                ; one of: callee, caller, both, no (default is no)
+
+;parkedcallhangup = caller      ; Enables or disables DTMF based hangups when picking up a parked call.
+                                ; one of: callee, caller, both, no (default is no)
+
+;findslot => next               ; Sets the method for selecting parking spaces when a call is parked
+                                ; 'next' - use the next parking space from the most recently used one.
+                                ; 'first' - use the lowest numbered parking space available
+
+;parkedmusicclass = default     ; This is the MOH class to use for the parked channel
+                                ; as long as the class is not set on the channel directly
+                                ; using Set(CHANNEL(musicclass)=whatever) in the dialplan
+
+;*** Define another parking lot
+;
+; The parkinglot used can be set with the CHANNEL(parkinglot) dialplan function or by
+; setting the 'parkinglot' configuration for a channel in its configuration file.
+;
+; Parking lots can now be any named configuration category aside from
+; 'general' which is reserved for general options. They no longer need to be
+; prefixed with 'parkinglot_'
+;
+;[edvina]
+;context => edvina_park
+;parkpos => 800-850
+;findslot => next
+;comebacktoorigin = no
+;comebackdialtime = 90
+;comebackcontext = edvinapark-timeout
+;parkedmusicclass = edvina
+;
+; Since edvina doesn't define parkext, extensions won't automatically be
+; created for parking to it or for retrieving calls from it. These can be
+; created manually in the dial plan by using the Park and ParkedCall
+; applications.
diff --git a/configs/samples/res_pgsql.conf.sample b/configs/samples/res_pgsql.conf.sample
new file mode 100644
index 000000000..cc7442fd0
--- /dev/null
+++ b/configs/samples/res_pgsql.conf.sample
@@ -0,0 +1,30 @@
+;
+; 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
+;dbappname=asterisk    ; Postgres application_name support (optional). Whitespace not allowed.
+;
+; 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
diff --git a/configs/samples/res_pktccops.conf.sample b/configs/samples/res_pktccops.conf.sample
new file mode 100644
index 000000000..769508db6
--- /dev/null
+++ b/configs/samples/res_pktccops.conf.sample
@@ -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
+;
diff --git a/configs/samples/res_snmp.conf.sample b/configs/samples/res_snmp.conf.sample
new file mode 100644
index 000000000..a6e40c8e2
--- /dev/null
+++ b/configs/samples/res_snmp.conf.sample
@@ -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
diff --git a/configs/samples/res_stun_monitor.conf.sample b/configs/samples/res_stun_monitor.conf.sample
new file mode 100644
index 000000000..12d32a4cd
--- /dev/null
+++ b/configs/samples/res_stun_monitor.conf.sample
@@ -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.
diff --git a/configs/samples/rtp.conf.sample b/configs/samples/rtp.conf.sample
new file mode 100644
index 000000000..c22acaa9f
--- /dev/null
+++ b/configs/samples/rtp.conf.sample
@@ -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=
diff --git a/configs/samples/say.conf.sample b/configs/samples/say.conf.sample
new file mode 100644
index 000000000..6285dbe1e
--- /dev/null
+++ b/configs/samples/say.conf.sample
@@ -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}
diff --git a/configs/samples/sip.conf.sample b/configs/samples/sip.conf.sample
new file mode 100644
index 000000000..0d81a6b63
--- /dev/null
+++ b/configs/samples/sip.conf.sample
@@ -0,0 +1,1571 @@
+;
+; SIP Configuration example for Asterisk
+;
+; Note: Please read the security documentation for Asterisk in order to
+; 	understand the risks of installing Asterisk with the sample
+;	configuration. If your Asterisk is installed on a public
+;	IP address connected to the Internet, you will want to learn
+;	about the various security settings BEFORE you start
+;	Asterisk.
+;
+;	Especially note the following settings:
+;		- allowguest (default enabled)
+;		- permit/deny/acl - IP address filters
+;		- contactpermit/contactdeny/contactacl - IP address filters for registrations
+;		- context - Which set of services you offer various users
+;
+; SIP dial strings
+;-----------------------------------------------------------
+; In the dialplan (extensions.conf) you can use several
+; syntaxes for dialing SIP devices.
+;        SIP/devicename
+;        SIP/username@domain   (SIP uri)
+;        SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]
+;        SIP/devicename/extension
+;        SIP/devicename/extension/IPorHost
+;        SIP/username@domain//IPorHost
+;
+;
+; Devicename
+;        devicename is defined as a peer in a section below.
+;
+; username@domain
+;        Call any SIP user on the Internet
+;        (Don't forget to enable DNS SRV records if you want to use this)
+;
+; devicename/extension
+;        If you define a SIP proxy as a peer below, you may call
+;        SIP/proxyhostname/user or SIP/user@proxyhostname
+;        where the proxyhostname is defined in a section below
+;        This syntax also works with ATA's with FXO ports
+;
+; SIP/username[:password[:md5secret[:authname]]]@host[:port]
+;        This form allows you to specify password or md5secret and authname
+;        without altering any authentication data in config.
+;        Examples:
+;
+;        SIP/*98@mysipproxy
+;        SIP/sales:topsecret::account02@domain.com:5062
+;        SIP/12345678::bc53f0ba8ceb1ded2b70e05c3f91de4f:myname@192.168.0.1
+;
+; IPorHost
+;        The next server for this call regardless of domain/peer
+;
+; All of these dial strings specify the SIP request URI.
+; In addition, you can specify a specific To: header by adding an
+; exclamation mark after the dial string, like
+;
+;         SIP/sales@mysipproxy!sales@edvina.net
+;
+; A new feature for 1.8 allows one to specify a host or IP address to use
+; when routing the call. This is typically used in tandem with func_srv if
+; multiple methods of reaching the same domain exist. The host or IP address
+; is specified after the third slash in the dialstring. Examples:
+;
+; SIP/devicename/extension/IPorHost
+; SIP/username@domain//IPorHost
+;
+; CLI Commands
+; -------------------------------------------------------------
+; Useful CLI commands to check peers/users:
+;   sip show peers               Show all SIP peers (including friends)
+;   sip show registry            Show status of hosts we register with
+;
+;   sip set debug on             Show all SIP messages
+;
+;   sip reload                   Reload configuration file
+;   sip show settings            Show the current channel configuration
+;
+;------- Naming devices ------------------------------------------------------
+;
+; When naming devices, make sure you understand how Asterisk matches calls
+; that come in.
+;	1. Asterisk checks the SIP From: address username and matches against
+;	   names of devices with type=user
+;	   The name is the text between square brackets [name]
+;	2. Asterisk checks the From: addres and matches the list of devices
+;	   with a type=peer
+;	3. Asterisk checks the IP address (and port number) that the INVITE
+;	   was sent from and matches against any devices with type=peer
+;
+; Don't mix extensions with the names of the devices. Devices need a unique
+; name. The device name is *not* used as phone numbers. Phone numbers are
+; anything you declare as an extension in the dialplan (extensions.conf).
+;
+; When setting up trunks, make sure there's no risk that any From: username
+; (caller ID) will match any of your device names, because then Asterisk
+; might match the wrong device.
+;
+; Note: The parameter "username" is not the username and in most cases is
+;       not needed at all. Check below. In later releases, it's renamed
+;       to "defaultuser" which is a better name, since it is used in
+;       combination with the "defaultip" setting.
+;-----------------------------------------------------------------------------
+
+; ** Old configuration options **
+; The "call-limit" configuation option is considered old is replaced
+; by new functionality. To enable callcounters, you use the new 
+; "callcounter" setting (for extension states in queue and subscriptions)
+; You are encouraged to use the dialplan groupcount functionality
+; to enforce call limits instead of using this channel-specific method.
+; You can still set limits per device in sip.conf or in a database by using
+; "setvar" to set variables that can be used in the dialplan for various limits.
+
+[general]
+context=public                  ; Default context for incoming calls. Defaults to 'default'
+;allowguest=no                  ; Allow or reject guest calls (default is yes)
+				; If your Asterisk is connected to the Internet
+				; and you have allowguest=yes
+				; you want to check which services you offer everyone
+				; out there, by enabling them in the default context (see below).
+;match_auth_username=yes        ; if available, match user entry using the
+                                ; 'username' field from the authentication line
+                                ; instead of the From: field.
+allowoverlap=no                 ; Disable overlap dialing support. (Default is yes)
+;allowoverlap=yes               ; Enable RFC3578 overlap dialing support.
+                                ; Can use the Incomplete application to collect the
+                                ; needed digits from an ambiguous dialplan match.
+;allowoverlap=dtmf              ; Enable overlap dialing support using DTMF delivery
+                                ; methods (inband, RFC2833, SIP INFO) in the early
+                                ; media phase.  Uses the Incomplete application to
+                                ; collect the needed digits.
+;allowtransfer=no               ; Disable all transfers (unless enabled in peers or users)
+                                ; Default is enabled. The Dial() options 't' and 'T' are not
+                                ; related as to whether SIP transfers are allowed or not.
+;realm=mydomain.tld             ; Realm for digest authentication
+                                ; defaults to "asterisk". If you set a system name in
+                                ; asterisk.conf, it defaults to that system name
+                                ; Realms MUST be globally unique according to RFC 3261
+                                ; Set this to your host name or domain name
+;domainsasrealm=no              ; Use domains list as realms
+                                ; You can serve multiple Realms specifying several
+                                ; 'domain=...' directives (see below). 
+                                ; In this case Realm will be based on request 'From'/'To' header
+                                ; and should match one of domain names.
+                                ; Otherwise default 'realm=...' will be used.
+;recordonfeature=automixmon	; Default feature to use when receiving 'Record: on' header
+				; from an INFO message. Defaults to 'automon'. Works with
+				; dynamic features. Feature must be usable on requesting
+				; channel for it to work. Setting this value to a blank
+				; will disable it.
+;recordofffeature=automixmon	; Default feature to use when receiving 'Record: off' header
+				; from an INFO message. Defaults to 'automon'. Works with
+				; dynamic features. Feature must be usable on requesting
+				; channel for it to work. Setting this value to a blank
+				; will disable it.
+
+; With the current situation, you can do one of four things:
+;  a) Listen on a specific IPv4 address.      Example: bindaddr=192.0.2.1
+;  b) Listen on a specific IPv6 address.      Example: bindaddr=2001:db8::1
+;  c) Listen on the IPv4 wildcard.            Example: bindaddr=0.0.0.0
+;  d) Listen on the IPv4 and IPv6 wildcards.  Example: bindaddr=::
+; (You can choose independently for UDP, TCP, and TLS, by specifying different values for
+; "udpbindaddr", "tcpbindaddr", and "tlsbindaddr".)
+; (Note that using bindaddr=:: will show only a single IPv6 socket in netstat.
+;  IPv4 is supported at the same time using IPv4-mapped IPv6 addresses.)
+;
+; You may optionally add a port number. (The default is port 5060 for UDP and TCP, 5061
+; for TLS).
+;   IPv4 example: bindaddr=0.0.0.0:5062
+;   IPv6 example: bindaddr=[::]:5062
+;
+; The address family of the bound UDP address is used to determine how Asterisk performs
+; DNS lookups. In cases a) and c) above, only A records are considered. In case b), only
+; AAAA records are considered. In case d), both A and AAAA records are considered. Note,
+; however, that Asterisk ignores all records except the first one. In case d), when both A
+; and AAAA records are available, either an A or AAAA record will be first, and which one
+; depends on the operating system. On systems using glibc, AAAA records are given
+; priority.
+
+udpbindaddr=0.0.0.0             ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
+                                ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
+
+; When a dialog is started with another SIP endpoint, the other endpoint
+; should include an Allow header telling us what SIP methods the endpoint
+; implements. However, some endpoints either do not include an Allow header
+; or lie about what methods they implement. In the former case, Asterisk
+; makes the assumption that the endpoint supports all known SIP methods.
+; If you know that your SIP endpoint does not provide support for a specific
+; method, then you may provide a comma-separated list of methods that your
+; endpoint does not implement in the disallowed_methods option. Note that 
+; if your endpoint is truthful with its Allow header, then there is no need 
+; to set this option. This option may be set in the general section or may
+; be set per endpoint. If this option is set both in the general section and
+; in a peer section, then the peer setting completely overrides the general
+; setting (i.e. the result is *not* the union of the two options).
+;
+; Note also that while Asterisk currently will parse an Allow header to learn
+; what methods an endpoint supports, the only actual use for this currently
+; is for determining if Asterisk may send connected line UPDATE requests and
+; MESSAGE requests. Its use may be expanded in the future.
+;
+; disallowed_methods = UPDATE
+
+;
+; Note that the TCP and TLS support for chan_sip is currently considered
+; experimental.  Since it is new, all of the related configuration options are
+; subject to change in any release.  If they are changed, the changes will
+; be reflected in this sample configuration file, as well as in the UPGRADE.txt file.
+;
+tcpenable=no                    ; Enable server for incoming TCP connections (default is no)
+tcpbindaddr=0.0.0.0             ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
+                                ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
+
+;tlsenable=no                   ; Enable server for incoming TLS (secure) connections (default is no)
+;tlsbindaddr=0.0.0.0            ; IP address for TLS server to bind to (0.0.0.0) binds to all interfaces)
+                                ; Optionally add a port number, 192.168.1.1:5063 (default is port 5061)
+                                ; Remember that the IP address must match the common name (hostname) in the
+                                ; certificate, so you don't want to bind a TLS socket to multiple IP addresses.
+                                ; For details how to construct a certificate for SIP see 
+                                ; http://tools.ietf.org/html/draft-ietf-sip-domain-certs
+
+;tcpauthtimeout = 30            ; tcpauthtimeout 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)
+
+;tcpauthlimit = 100             ; tcpauthlimit specifies the maximum number of
+				; unauthenticated sessions that will be allowed
+                                ; to connect at any given time. (default: 100)
+
+;websocket_write_timeout = 100  ; Default write timeout to set on websocket transports.
+                                ; This value may need to be adjusted for connections where
+                                ; Asterisk must write a substantial amount of data and the
+                                ; receiving clients are slow to process the received information.
+                                ; Value is in milliseconds; default is 100 ms.
+
+transport=udp                   ; Set the default transports.  The order determines the primary default transport.
+                                ; If tcpenable=no and the transport set is tcp, we will fallback to UDP.
+
+srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
+                                ; Note: Asterisk only uses the first host
+                                ; in SRV records
+                                ; Disabling DNS SRV lookups disables the
+                                ; ability to place SIP calls based on domain
+                                ; names to some other SIP users on the Internet
+                                ; Specifying a port in a SIP peer definition or
+                                ; when dialing outbound calls will supress SRV
+                                ; lookups for that peer or call.
+
+;pedantic=yes                   ; Enable checking of tags in headers,
+                                ; international character conversions in URIs
+                                ; and multiline formatted headers for strict
+                                ; SIP compatibility (defaults to "yes")
+
+; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
+;tos_sip=cs3                    ; Sets TOS for SIP packets.
+;tos_audio=ef                   ; Sets TOS for RTP audio packets.
+;tos_video=af41                 ; Sets TOS for RTP video packets.
+;tos_text=af41                  ; Sets TOS for RTP text packets.
+
+;cos_sip=3                      ; Sets 802.1p priority for SIP packets.
+;cos_audio=5                    ; Sets 802.1p priority for RTP audio packets.
+;cos_video=4                    ; Sets 802.1p priority for RTP video packets.
+;cos_text=3                     ; Sets 802.1p priority for RTP text packets.
+
+;maxexpiry=3600                 ; Maximum allowed time of incoming registrations (seconds)
+;minexpiry=60                   ; Minimum length of registrations (default 60)
+;defaultexpiry=120              ; Default length of incoming/outgoing registration
+;submaxexpiry=3600              ; Maximum allowed time of incoming subscriptions (seconds), default: maxexpiry
+;subminexpiry=60                ; Minimum length of subscriptions, default: minexpiry
+;mwiexpiry=3600                 ; Expiry time for outgoing MWI subscriptions
+;maxforwards=70			; Setting for the SIP Max-Forwards: header (loop prevention)
+				; Default value is 70
+;qualifyfreq=60                 ; Qualification: How often to check for the host to be up in seconds
+				; and reported in milliseconds with sip show settings.
+                                ; Set to low value if you use low timeout for NAT of UDP sessions
+				; Default: 60
+;qualifygap=100			; Number of milliseconds between each group of peers being qualified
+				; Default: 100
+;qualifypeers=1			; Number of peers in a group to be qualified at the same time
+				; Default: 1
+;keepalive=60                   ; Interval at which keepalive packets should be sent to a peer
+				; Valid options are yes (60 seconds), no, or the number of seconds.
+                                ; Default: 0
+;notifymimetype=text/plain      ; Allow overriding of mime type in MWI NOTIFY
+;buggymwi=no                    ; Cisco SIP firmware doesn't support the MWI RFC
+                                ; fully. Enable this option to not get error messages
+                                ; when sending MWI to phones with this bug.
+;mwi_from=asterisk              ; When sending MWI NOTIFY requests, use this setting in
+                                ; the From: header as the "name" portion. Also fill the
+			        ; "user" portion of the URI in the From: header with this
+			        ; value if no fromuser is set
+			        ; Default: empty
+;vmexten=voicemail              ; dialplan extension to reach mailbox sets the
+                                ; Message-Account in the MWI notify message
+                                ; defaults to "asterisk"
+
+; Codec negotiation
+;
+; When Asterisk is receiving a call, the codec will initially be set to the
+; first codec in the allowed codecs defined for the user receiving the call
+; that the caller also indicates that it supports. But, after the caller
+; starts sending RTP, Asterisk will switch to using whatever codec the caller
+; is sending.
+;
+; When Asterisk is placing a call, the codec used will be the first codec in
+; the allowed codecs that the callee indicates that it supports. Asterisk will
+; *not* switch to whatever codec the callee is sending.
+;
+;preferred_codec_only=yes       ; Respond to a SIP invite with the single most preferred codec
+                                ; rather than advertising all joint codec capabilities. This
+                                ; limits the other side's codec choice to exactly what we prefer.
+
+;disallow=all                   ; First disallow all codecs
+;allow=ulaw                     ; Allow codecs in order of preference
+;allow=ilbc                     ; 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.
+;
+; 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.
+;
+; This option may be specified globally, or on a per-user or per-peer basis.
+;
+;mohinterpret=default
+;
+; This option specifies which music on hold class to suggest to the peer channel
+; when this channel places the peer on hold. It may be specified globally or on
+; a per-user or per-peer basis.
+;
+;mohsuggest=default
+;
+;parkinglot=plaza               ; Sets the default parking lot for call parking
+                                ; This may also be set for individual users/peers
+                                ; Parkinglots are configured in features.conf
+;language=en                    ; Default language setting for all users/peers
+                                ; This may also be set for individual users/peers
+;tonezone=se			; Default tonezone for all users/peers
+                                ; This may also be set for individual users/peers
+
+;relaxdtmf=yes                  ; Relax dtmf handling
+;trustrpid = no                 ; If Remote-Party-ID should be trusted
+;sendrpid = yes                 ; If Remote-Party-ID should be sent (defaults to no)
+;sendrpid = rpid                ; Use the "Remote-Party-ID" header
+                                ; to send the identity of the remote party
+                                ; This is identical to sendrpid=yes
+;sendrpid = pai                 ; Use the "P-Asserted-Identity" header
+                                ; to send the identity of the remote party
+;rpid_update = no               ; In certain cases, the only method by which a connected line
+                                ; change may be immediately transmitted is with a SIP UPDATE request.
+                                ; If communicating with another Asterisk server, and you wish to be able
+                                ; transmit such UPDATE messages to it, then you must enable this option.
+                                ; Otherwise, we will have to wait until we can send a reinvite to
+                                ; transmit the information.
+;trust_id_outbound = no         ; Controls whether or not we trust this peer with private identity
+                                ; information (when the remote party has callingpres=prohib or equivalent).
+                                ; no - RPID/PAI headers will not be included for private peer information
+                                ; yes - RPID/PAI headers will include the private peer information. Privacy
+                                ;       requirements will be indicated in a Privacy header for sendrpid=pai
+                                ; legacy - RPID/PAI will be included for private peer information. In the
+                                ;       case of sendrpid=pai, private data that would be included in them
+                                ;       will be anonymized. For sendrpid=rpid, private data may be included
+                                ;       but the remote party's domain will be anonymized. The way legacy
+                                ;       behaves may violate RFC-3325, but it follows historic behavior.
+                                ; This option is set to 'legacy' by default
+;prematuremedia=no              ; Some ISDN links send empty media frames before 
+                                ; the call is in ringing or progress state. The SIP 
+                                ; channel will then send 183 indicating early media
+                                ; which will be empty - thus users get no ring signal.
+                                ; Setting this to "yes" will stop any media before we have
+                                ; call progress (meaning the SIP channel will not send 183 Session
+                                ; Progress for early media). Default is "yes". Also make sure that
+                                ; the SIP peer is configured with progressinband=never. 
+                                ;
+                                ; In order for "noanswer" applications to work, you need to run
+                                ; the progress() application in the priority before the app.
+
+;progressinband=never           ; If we should generate in-band ringing always
+                                ; use 'never' to never use in-band signalling, even in cases
+                                ; where some buggy devices might not render it
+                                ; Valid values: yes, no, never Default: never
+;useragent=Asterisk PBX         ; Allows you to change the user agent string
+                                ; The default user agent string also contains the Asterisk
+                                ; version. If you don't want to expose this, change the
+                                ; useragent string.
+;promiscredir = no              ; If yes, allows 302 or REDIR to non-local SIP address
+                                ; Note that promiscredir when redirects are made to the
+                                ; local system will cause loops since Asterisk is incapable
+                                ; of performing a "hairpin" call.
+;usereqphone = no               ; If yes, ";user=phone" is added to uri that contains
+                                ; a valid phone number
+;dtmfmode = rfc2833             ; Set default dtmfmode for sending DTMF. Default: rfc2833
+                                ; Other options:
+                                ; info : SIP INFO messages (application/dtmf-relay)
+                                ; shortinfo : SIP INFO messages (application/dtmf)
+                                ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
+                                ; auto : Use rfc2833 if offered, inband otherwise
+
+;compactheaders = yes           ; send compact sip headers.
+;
+;videosupport=yes               ; Turn on support for SIP video. You need to turn this
+                                ; on in this section to get any video support at all.
+                                ; You can turn it off on a per peer basis if the general
+                                ; video support is enabled, but you can't enable it for
+                                ; one peer only without enabling in the general section.
+                                ; If you set videosupport to "always", then RTP ports will
+                                ; always be set up for video, even on clients that don't
+                                ; support it.  This assists callfile-derived calls and
+                                ; certain transferred calls to use always use video when
+                                ; available. [yes|NO|always]
+
+;textsupport=no                 ; Support for ITU-T T.140 realtime text.
+                                ; The default value is "no".
+
+;maxcallbitrate=384             ; Maximum bitrate for video calls (default 384 kb/s)
+                                ; Videosupport and maxcallbitrate is settable
+                                ; for peers and users as well
+;authfailureevents=no           ; generate manager "peerstatus" events when peer can't
+                                ; authenticate with Asterisk. Peerstatus will be "rejected".
+;alwaysauthreject = yes         ; When an incoming INVITE or REGISTER is to be rejected,
+                                ; for any reason, always reject with an identical response
+                                ; equivalent to valid username and invalid password/hash
+                                ; instead of letting the requester know whether there was
+                                ; a matching user or peer for their request.  This reduces
+                                ; the ability of an attacker to scan for valid SIP usernames.
+                                ; This option is set to "yes" by default.
+
+;auth_options_requests = yes    ; Enabling this option will authenticate OPTIONS requests just like
+                                ; INVITE requests are.  By default this option is disabled.
+
+;accept_outofcall_message = no  ; Disable this option to reject all MESSAGE requests outside of a
+                                ; call.  By default, this option is enabled.  When enabled, MESSAGE
+                                ; requests are passed in to the dialplan.
+
+;outofcall_message_context = messages ; Context all out of dialog msgs are sent to. When this
+                                      ; option is not set, the context used during peer matching
+                                      ; is used. This option can be defined at both the peer and
+                                      ; global level.
+
+;auth_message_requests = yes    ; Enabling this option will authenticate MESSAGE requests.
+                                ; By default this option is enabled.  However, it can be disabled
+                                ; should an application desire to not load the Asterisk server with
+                                ; doing authentication and implement end to end security in the
+                                ; message body.
+
+;g726nonstandard = yes          ; If the peer negotiates G726-32 audio, use AAL2 packing
+                                ; order instead of RFC3551 packing order (this is required
+                                ; for Sipura and Grandstream ATAs, among others). This is
+                                ; contrary to the RFC3551 specification, the peer _should_
+                                ; be negotiating AAL2-G726-32 instead :-(
+;outboundproxy=proxy.provider.domain            ; send outbound signaling to this proxy, not directly to the devices
+;outboundproxy=proxy.provider.domain:8080       ; send outbound signaling to this proxy, not directly to the devices
+;outboundproxy=proxy.provider.domain,force      ; Send ALL outbound signalling to proxy, ignoring route: headers
+;outboundproxy=tls://proxy.provider.domain      ; same as '=proxy.provider.domain' except we try to connect with tls
+;outboundproxy=192.0.2.1                        ; IPv4 address literal (default port is 5060)
+;outboundproxy=2001:db8::1                      ; IPv6 address literal (default port is 5060)
+;outboundproxy=192.168.0.2.1:5062               ; IPv4 address literal with explicit port
+;outboundproxy=[2001:db8::1]:5062               ; IPv6 address literal with explicit port
+;                                               ; (could also be tcp,udp) - defining transports on the proxy line only
+;                                               ; applies for the global proxy, otherwise use the transport= option
+
+;supportpath=yes		; This activates parsing and handling of Path header as defined in RFC 3327. This enables
+				; Asterisk to route outgoing out-of-dialog requests via a set of proxies by using a pre-loaded
+				; route-set defined by the Path headers in the REGISTER request.
+				; NOTE: There are multiple things to consider with this setting:
+				;  * As this influences routing of SIP requests make sure to not trust Path headers provided
+				;    by the user's SIP client (the proxy in front of Asterisk should remove existing user
+				;    provided Path headers).
+				;  * When a peer has both a path and outboundproxy set, the path will be added to Route: header
+				;    but routing to next hop is done using the outboundproxy.
+				;  * If set globally, not only will all peers use the Path header, but outbound REGISTER
+				;    requests from Asterisk will add path to the Supported header.
+
+;rtsavepath=yes                 ; If using dynamic realtime, store the path headers
+
+;matchexternaddrlocally = yes     ; Only substitute the externaddr or externhost setting if it matches
+                                ; your localnet setting. Unless you have some sort of strange network
+                                ; setup you will not need to enable this.
+
+;dynamic_exclude_static = yes   ; Disallow all dynamic hosts from registering
+                                ; as any IP address used for staticly defined
+                                ; hosts.  This helps avoid the configuration
+                                ; error of allowing your users to register at
+                                ; the same address as a SIP provider.
+
+;contactdeny=0.0.0.0/0.0.0.0           ; Use contactpermit and contactdeny to
+;contactpermit=172.16.0.0/255.255.0.0  ; restrict at what IPs your users may
+                                       ; register their phones.
+;contactacl=named_acl_example          ; Use named ACLs defined in acl.conf
+
+;rtp_engine=asterisk            ; RTP engine to use when communicating with the device
+
+;
+; If regcontext is specified, Asterisk will dynamically create and destroy a
+; NoOp priority 1 extension for a given peer who registers or unregisters with
+; us and have a "regexten=" configuration item.
+; Multiple contexts may be specified by separating them with '&'. The
+; actual extension is the 'regexten' parameter of the registering peer or its
+; name if 'regexten' is not provided.  If more than one context is provided,
+; the context must be specified within regexten by appending the desired
+; context after '@'.  More than one regexten may be supplied if they are
+; separated by '&'.  Patterns may be used in regexten.
+;
+;regcontext=sipregistrations
+;regextenonqualify=yes          ; Default "no"
+                                ; If you have qualify on and the peer becomes unreachable
+                                ; this setting will enforce inactivation of the regexten
+                                ; extension for the peer
+;legacy_useroption_parsing=yes	; Default "no"      ; If you have this option enabled and there are semicolons
+                                                    ; in the user field of a sip URI, the field be truncated
+                                                    ; at the first semicolon seen. This effectively makes
+                                                    ; semicolon a non-usable character for peer names, extensions,
+                                                    ; and maybe other, less tested things.  This can be useful
+                                                    ; for improving compatability with devices that like to use
+                                                    ; user options for whatever reason.  The behavior is similar to
+                                                    ; how SIP URI's were typically handled in 1.6.2, hence the name.
+
+;send_diversion=no              ; Default "yes"     ; Asterisk normally sends Diversion headers with certain SIP
+                                                    ; invites to relay data about forwarded calls. If this option
+                                                    ; is disabled, Asterisk won't send Diversion headers unless
+                                                    ; they are added manually.
+
+; 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
+
+
+;use_q850_reason = no ; Default "no"
+                      ; Set to yes add Reason header and use Reason header if it is available.
+
+; When the Transfer() application sends a REFER SIP message, extra headers specified in
+; the dialplan by way of SIPAddHeader are sent out with that message. 1.8 and earlier did not
+; add the extra headers. To revert to 1.8- behavior, call SIPRemoveHeader with no arguments
+; before calling Transfer() to remove all additional headers from the channel. The setting
+; below is for transitional compatibility only.
+;
+;refer_addheaders=yes	; on by default
+
+;autocreatepeer=no             ; Allow any UAC not explicitly defined to register
+                               ; WITHOUT AUTHENTICATION. Enabling this options poses a high
+                               ; potential security risk and should be avoided unless the
+                               ; server is behind a trusted firewall.
+                               ; If set to "yes", then peers created in this fashion
+                               ; are purged during SIP reloads.
+                               ; When set to "persist", the peers created in this fashion
+                               ; are not purged during SIP reloads.
+
+;
+;------------------------ TLS settings ------------------------------------------------------------
+;tlscertfile=</path/to/certificate.pem> ; Certificate chain (*.pem format only) to use for TLS connections
+                                        ; The certificates must be sorted starting with the subject's certificate
+                                        ; and followed by intermediate CA certificates if applicable.
+                                        ; Default is to look for "asterisk.pem" in current directory
+
+;tlsprivatekey=</path/to/private.pem> ; Private key file (*.pem format only) for TLS connections.
+                                      ; If no tlsprivatekey is specified, tlscertfile is searched for
+                                      ; for both public and private key.
+
+;tlscafile=</path/to/certificate>
+;        If the server your connecting to uses a self signed certificate
+;        you should have their certificate installed here so the code can
+;        verify the authenticity of their certificate.
+
+;tlscapath=</path/to/ca/dir>
+;        A directory full of CA certificates.  The files must be named with
+;        the CA subject name hash value.
+;        (see man SSL_CTX_load_verify_locations for more info)
+
+;tlsdontverifyserver=[yes|no]
+;        If set to yes, don't verify the servers certificate when acting as
+;        a client.  If you don't have the server's CA certificate you can
+;        set this and it will connect without requiring tlscafile to be set.
+;        Default is no.
+
+;tlscipher=<SSL cipher string>
+;        A string specifying which SSL ciphers to use or not use
+;        A list of valid SSL cipher strings can be found at:
+;                http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
+;
+;tlsclientmethod=tlsv1     ; values include tlsv1, sslv3, sslv2.
+                           ; Specify protocol for outbound client connections.
+                           ; If left unspecified, the default is sslv2.
+;
+;--------------------------- SIP timers ----------------------------------------------------
+; These timers are used primarily in INVITE transactions.
+; The default for Timer T1 is 500 ms or the measured run-trip time between
+; Asterisk and the device if you have qualify=yes for the device.
+;
+;t1min=100                      ; Minimum roundtrip time for messages to monitored hosts
+                                ; Defaults to 100 ms
+;timert1=500                    ; Default T1 timer
+                                ; Defaults to 500 ms or the measured round-trip
+                                ; time to a peer (qualify=yes).
+;timerb=32000                   ; Call setup timer. If a provisional response is not received
+                                ; in this amount of time, the call will autocongest
+                                ; Defaults to 64*timert1
+
+;--------------------------- RTP timers ----------------------------------------------------
+; These timers are currently used for both audio and video streams. The RTP timeouts
+; are only applied to the audio channel.
+; The settings are settable in the global section as well as per device
+;
+;rtptimeout=60                  ; Terminate call if 60 seconds of no RTP or RTCP activity
+                                ; on the audio channel
+                                ; when we're not on hold. This is to be able to hangup
+                                ; a call in the case of a phone disappearing from the net,
+                                ; like a powerloss or grandma tripping over a cable.
+;rtpholdtimeout=300             ; Terminate call if 300 seconds of no RTP or RTCP activity
+                                ; on the audio channel
+                                ; when we're on hold (must be > rtptimeout)
+;rtpkeepalive=<secs>            ; Send keepalives in the RTP stream to keep NAT open
+                                ; (default is off - zero)
+
+;--------------------------- SIP Session-Timers (RFC 4028)------------------------------------
+; SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions.
+; This mechanism can detect and reclaim SIP channels that do not terminate through normal
+; signaling procedures. Session-Timers can be configured globally or at a user/peer level.
+; The operation of Session-Timers is driven by the following configuration parameters:
+;
+; * session-timers    - Session-Timers feature operates in the following three modes:
+;                            originate : Request and run session-timers always
+;                            accept    : Run session-timers only when requested by other UA
+;                            refuse    : Do not run session timers in any case
+;                       The default mode of operation is 'accept'.
+; * session-expires   - Maximum session refresh interval in seconds. Defaults to 1800 secs.
+; * session-minse     - Minimum session refresh interval in seconds. Defualts to 90 secs.
+; * session-refresher - The session refresher (uac|uas). Defaults to 'uas'.
+;                            uac - Default to the caller initially refreshing when possible
+;                            uas - Default to the callee initially refreshing when possible
+;
+; Note that, due to recommendations in RFC 4028, Asterisk will always honor the other
+; endpoint's preference for who will handle refreshes. Asterisk will never override the
+; preferences of the other endpoint. Doing so could result in Asterisk and the endpoint
+; fighting over who sends the refreshes. This holds true for the initiation of session
+; timers and subsequent re-INVITE requests whether Asterisk is the caller or callee, or
+; whether Asterisk is currently the refresher or not.
+;
+;session-timers=originate
+;session-expires=600
+;session-minse=90
+;session-refresher=uac
+;
+;--------------------------- SIP DEBUGGING ---------------------------------------------------
+;sipdebug = yes                 ; Turn on SIP debugging by default, from
+                                ; the moment the channel loads this configuration
+;recordhistory=yes              ; Record SIP history by default
+                                ; (see sip history / sip no history)
+;dumphistory=yes                ; Dump SIP history at end of SIP dialogue
+                                ; SIP history is output to the DEBUG logging channel
+
+
+;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
+; You can subscribe to the status of extensions with a "hint" priority
+; (See extensions.conf.sample for examples)
+; chan_sip support two major formats for notifications: dialog-info and SIMPLE
+;
+; You will get more detailed reports (busy etc) if you have a call counter enabled
+; for a device.
+;
+; If you set the busylevel, we will indicate busy when we have a number of calls that
+; matches the busylevel treshold.
+;
+; For queues, you will need this level of detail in status reporting, regardless
+; if you use SIP subscriptions. Queues and manager use the same internal interface
+; for reading status information.
+;
+; Note: Subscriptions does not work if you have a realtime dialplan and use the
+; realtime switch.
+;
+;allowsubscribe=no              ; Disable support for subscriptions. (Default is yes)
+;subscribecontext = default     ; Set a specific context for SUBSCRIBE requests
+                                ; Useful to limit subscriptions to local extensions
+                                ; Settable per peer/user also
+;notifyringing = no             ; Control whether subscriptions already INUSE get sent
+                                ; RINGING when another call is sent (default: yes)
+;notifyhold = yes               ; Notify subscriptions on HOLD state (default: no)
+                                ; Turning on notifyringing and notifyhold will add a lot
+                                ; more database transactions if you are using realtime.
+;notifycid = yes                ; Control whether caller ID information is sent along with
+                                ; dialog-info+xml notifications (supported by snom phones).
+                                ; Note that this feature will only work properly when the
+                                ; incoming call is using the same extension and context that
+                                ; is being used as the hint for the called extension.  This means
+                                ; that it won't work when using subscribecontext for your sip
+                                ; user or peer (if subscribecontext is different than context).
+                                ; This is also limited to a single caller, meaning that if an
+                                ; extension is ringing because multiple calls are incoming,
+                                ; only one will be used as the source of caller ID.  Specify
+                                ; 'ignore-context' to ignore the called context when looking
+                                ; for the caller's channel.  The default value is 'no.' Setting
+                                ; notifycid to 'ignore-context' also causes call-pickups attempted
+                                ; via SNOM's NOTIFY mechanism to set the context for the call pickup
+                                ; to PICKUPMARK.
+;callcounter = yes              ; Enable call counters on devices. This can be set per
+                                ; device too.
+
+;----------------------------------------- T.38 FAX SUPPORT ----------------------------------
+;
+; This setting is available in the [general] section as well as in device configurations.
+; Setting this to yes enables T.38 FAX (UDPTL) on SIP calls; it defaults to off.
+;
+; t38pt_udptl = yes            ; Enables T.38 with FEC error correction.
+; t38pt_udptl = yes,fec        ; Enables T.38 with FEC error correction.
+; t38pt_udptl = yes,redundancy ; Enables T.38 with redundancy error correction.
+; t38pt_udptl = yes,none       ; Enables T.38 with no error correction.
+;
+; In some cases, T.38 endpoints will provide a T38FaxMaxDatagram value (during T.38 setup) that
+; is based on an incorrect interpretation of the T.38 recommendation, and results in failures
+; because Asterisk does not believe it can send T.38 packets of a reasonable size to that
+; endpoint (Cisco media gateways are one example of this situation). In these cases, during a
+; T.38 call you will see warning messages on the console/in the logs from the Asterisk UDPTL
+; stack complaining about lack of buffer space to send T.38 FAX packets. If this occurs, you
+; can set an override (globally, or on a per-device basis) to make Asterisk ignore the
+; T38FaxMaxDatagram value specified by the other endpoint, and use a configured value instead.
+; This can be done by appending 'maxdatagram=<value>' to the t38pt_udptl configuration option,
+; like this:
+;
+; t38pt_udptl = yes,fec,maxdatagram=400 ; Enables T.38 with FEC error correction and overrides
+;                                       ; the other endpoint's provided value to assume we can
+;                                       ; send 400 byte T.38 FAX packets to it.
+;
+; FAX detection will cause the SIP 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 re-INVITE request.
+;
+; faxdetect = yes		; Default 'no', 'yes' enables both CNG and T.38 detection
+; faxdetect = cng		; Enables only CNG detection
+; faxdetect = t38		; Enables only T.38 detection
+;
+;----------------------------------------- OUTBOUND SIP REGISTRATIONS  ------------------------
+; Asterisk can register as a SIP user agent to a SIP proxy (provider)
+; Format for the register statement is:
+;       register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry]
+;
+;
+;
+; domain is either
+;	- domain in DNS
+; 	- host name in DNS
+;	- the name of a peer defined below or in realtime
+; The domain is where you register your username, so your SIP uri you are registering to
+; is username@domain
+;
+; If no extension is given, the 's' extension is used. The extension needs to
+; be defined in extensions.conf to be able to accept calls from this SIP proxy
+; (provider).
+;
+; A similar effect can be achieved by adding a "callbackextension" option in a peer section.
+; this is equivalent to having the following line in the general section:
+;
+;        register => username:secret@host/callbackextension
+;
+; and more readable because you don't have to write the parameters in two places
+; (note that the "port" is ignored - this is a bug that should be fixed).
+;
+; Note that a register= line doesn't mean that we will match the incoming call in any
+; other way than described above. If you want to control where the call enters your
+; dialplan, which context, you want to define a peer with the hostname of the provider's
+; server. If the provider has multiple servers to place calls to your system, you need
+; a peer for each server.
+;
+; Beginning with Asterisk version 1.6.2, the "user" portion of the register line may
+; contain a port number. Since the logical separator between a host and port number is a
+; ':' character, and this character is already used to separate between the optional "secret"
+; and "authuser" portions of the line, there is a bit of a hoop to jump through if you wish
+; to use a port here. That is, you must explicitly provide a "secret" and "authuser" even if
+; they are blank. See the third example below for an illustration.
+;
+;
+; Examples:
+;
+;register => 1234:password@mysipprovider.com
+;
+;     This will pass incoming calls to the 's' extension
+;
+;
+;register => 2345:password@sip_proxy/1234
+;
+;    Register 2345 at sip provider 'sip_proxy'.  Calls from this provider
+;    connect to local extension 1234 in extensions.conf, default context,
+;    unless you configure a [sip_proxy] section below, and configure a
+;    context.
+;    Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
+;    Tip 2: Use separate inbound and outbound sections for SIP providers
+;           (instead of type=friend) if you have calls in both directions
+;
+;register => 3456@mydomain:5082::@mysipprovider.com
+;
+;    Note that in this example, the optional authuser and secret portions have
+;    been left blank because we have specified a port in the user section
+;
+;register => tls://username:xxxxxx@sip-tls-proxy.example.org
+;
+;    The 'transport' part defaults to 'udp' but may also be 'tcp', 'tls', 'ws', or 'wss'.
+;    Using 'udp://' explicitly is also useful in case the username part
+;    contains a '/' ('user/name').
+
+;registertimeout=20             ; retry registration calls every 20 seconds (default)
+;registerattempts=10            ; Number of registration attempts before we give up
+                                ; 0 = continue forever, hammering the other server
+                                ; until it accepts the registration
+                                ; Default is 0 tries, continue forever
+;register_retry_403=yes         ; Treat 403 responses to registrations as if they were
+                                ; 401 responses and continue retrying according to normal
+                                ; retry rules.
+
+;----------------------------------------- OUTBOUND MWI SUBSCRIPTIONS -------------------------
+; Asterisk can subscribe to receive the MWI from another SIP server and store it locally for retrieval
+; by other phones. At this time, you can only subscribe using UDP as the transport.
+; Format for the mwi register statement is:
+;       mwi => user[:secret[:authuser]]@host[:port]/mailbox
+;
+; Examples:
+;mwi => 1234:password@mysipprovider.com/1234
+;mwi => 1234:password@myportprovider.com:6969/1234
+;mwi => 1234:password:authuser@myauthprovider.com/1234
+;mwi => 1234:password:authuser@myauthportprovider.com:6969/1234
+;
+; MWI received will be stored in the 1234 mailbox of the SIP_Remote context.
+; It can be used by other phones by following the below:
+; mailbox=1234@SIP_Remote
+;----------------------------------------- NAT SUPPORT ------------------------
+;
+; WARNING: SIP operation behind a NAT is tricky and you really need
+; to read and understand well the following section.
+;
+; When Asterisk is behind a NAT device, the "local" address (and port) that
+; a socket is bound to has different values when seen from the inside or
+; from the outside of the NATted network. Unfortunately this address must
+; be communicated to the outside (e.g. in SIP and SDP messages), and in
+; order to determine the correct value Asterisk needs to know:
+;
+; + whether it is talking to someone "inside" or "outside" of the NATted network.
+;   This is configured by assigning the "localnet" parameter with a list
+;   of network addresses that are considered "inside" of the NATted network.
+;   IF LOCALNET IS NOT SET, THE EXTERNAL ADDRESS WILL NOT BE SET CORRECTLY.
+;   Multiple entries are allowed, e.g. a reasonable set is the following:
+;
+;      localnet=192.168.0.0/255.255.0.0 ; RFC 1918 addresses
+;      localnet=10.0.0.0/255.0.0.0      ; Also RFC1918
+;      localnet=172.16.0.0/12           ; Another RFC1918 with CIDR notation
+;      localnet=169.254.0.0/255.255.0.0 ; Zero conf local network
+;
+; + the "externally visible" address and port number to be used when talking
+;   to a host outside the NAT. This information is derived by one of the
+;   following (mutually exclusive) config file parameters:
+;
+;   a. "externaddr = hostname[:port]" specifies a static address[:port] to
+;      be used in SIP and SDP messages.
+;      The hostname is looked up only once, when [re]loading sip.conf .
+;      If a port number is not present, use the port specified in the "udpbindaddr"
+;      (which is not guaranteed to work correctly, because a NAT box might remap the
+;      port number as well as the address).
+;      This approach can be useful if you have a NAT device where you can
+;      configure the mapping statically. Examples:
+;
+;        externaddr = 12.34.56.78          ; use this address.
+;        externaddr = 12.34.56.78:9900     ; use this address and port.
+;        externaddr = mynat.my.org:12600   ; Public address of my nat box.
+;        externtcpport = 9900   ; The externally mapped tcp port, when Asterisk is behind a static NAT or PAT. 
+;                               ; externtcpport will default to the externaddr or externhost port if either one is set. 
+;        externtlsport = 12600  ; The externally mapped tls port, when Asterisk is behind a static NAT or PAT.
+;                               ; externtlsport port will default to the RFC designated port of 5061.	
+;
+;   b. "externhost = hostname[:port]" is similar to "externaddr" except
+;      that the hostname is looked up every "externrefresh" seconds
+;      (default 10s). This can be useful when your NAT device lets you choose
+;      the port mapping, but the IP address is dynamic.
+;      Beware, you might suffer from service disruption when the name server
+;      resolution fails. Examples:
+;
+;        externhost=foo.dyndns.net       ; refreshed periodically
+;        externrefresh=180               ; change the refresh interval
+;
+;   Note that at the moment all these mechanism work only for the SIP socket.
+;   The IP address discovered with externaddr/externhost is reused for
+;   media sessions as well, but the port numbers are not remapped so you
+;   may still experience problems.
+;
+; NOTE 1: in some cases, NAT boxes will use different port numbers in
+; the internal<->external mapping. In these cases, the "externaddr" and
+; "externhost" might not help you configure addresses properly.
+;
+; NOTE 2: when using "externaddr" or "externhost", the address part is
+; also used as the external address for media sessions. Thus, the port
+; information in the SDP may be wrong!
+;
+; In addition to the above, Asterisk has an additional "nat" parameter to
+; address NAT-related issues in incoming SIP or media sessions.
+; In particular, depending on the 'nat= ' settings described below, Asterisk
+; may override the address/port information specified in the SIP/SDP messages,
+; and use the information (sender address) supplied by the network stack instead.
+; However, this is only useful if the external traffic can reach us.
+; The following settings are allowed (both globally and in individual sections):
+;
+;   nat = no                ; Do no special NAT handling other than RFC3581
+;   nat = force_rport       ; Pretend there was an rport parameter even if there wasn't
+;   nat = comedia           ; Send media to the port Asterisk received it from regardless
+;                           ; of where the SDP says to send it.
+;   nat = auto_force_rport  ; Set the force_rport option if Asterisk detects NAT (default)
+;   nat = auto_comedia      ; Set the comedia option if Asterisk detects NAT
+;
+; The nat settings can be combined. For example, to set both force_rport and comedia
+; one would set nat=force_rport,comedia. If any of the comma-separated options is 'no',
+; Asterisk will ignore any other settings and set nat=no. If one of the "auto" settings
+; is used in conjunction with its non-auto counterpart (nat=comedia,auto_comedia), then
+; the non-auto option will be ignored.
+;
+; The RFC 3581-defined 'rport' parameter allows a client to request that Asterisk send
+; SIP responses to it via the source IP and port from which the request originated
+; instead of the address/port listed in the top-most Via header. This is useful if a
+; client knows that it is behind a NAT and therefore cannot guess from what address/port
+; its request will be sent. Asterisk will always honor the 'rport' parameter if it is
+; sent. The force_rport setting causes Asterisk to always send responses back to the
+; address/port from which it received requests; even if the other side doesn't support
+; adding the 'rport' parameter.
+;
+; 'comedia RTP handling' refers to the technique of sending RTP to the port that the
+; the other endpoint's RTP arrived from, and means 'connection-oriented media'. This is
+; only partially related to RFC 4145 which was referred to as COMEDIA while it was in
+; draft form. This method is used to accomodate endpoints that may be located behind
+; NAT devices, and as such the address/port they tell Asterisk to send RTP packets to
+; for their media streams is not the actual address/port that will be used on the nearer
+; side of the NAT.
+;
+; IT IS IMPORTANT TO NOTE that if the nat setting in the general section differs from
+; the nat setting in a peer definition, then the peer username will be discoverable
+; by outside parties as Asterisk will respond to different ports for defined and
+; undefined peers. For this reason it is recommended to ONLY DEFINE NAT SETTINGS IN THE
+; GENERAL SECTION. Specifically, if nat=force_rport in one section and nat=no in the
+; other, then valid peers with settings differing from those in the general section will
+; be discoverable.
+;
+; In addition to these settings, Asterisk *always* uses 'symmetric RTP' mode as defined by
+; RFC 4961; Asterisk will always send RTP packets from the same port number it expects
+; to receive them on.
+;
+; The IP address used for media (audio, video, and text) in the SDP can also be overridden by using
+; the media_address configuration option. This is only applicable to the general section and
+; can not be set per-user or per-peer.
+;
+; media_address = 172.16.42.1
+;
+; 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_sip 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
+;
+; ICE/STUN/TURN usage can be enabled globally or on a per-peer basis using the icesupport
+; configuration option. When set to yes ICE support is enabled. When set to no it is disabled.
+; It is disabled by default.
+;
+; icesupport = yes
+
+;----------------------------------- MEDIA HANDLING --------------------------------
+; By default, Asterisk tries to re-invite media streams to an optimal path. If there's
+; no reason for Asterisk to stay in the media path, the media will be redirected.
+; This does not really work well in the case where Asterisk is outside and the
+; clients are on the inside of a NAT. In that case, you want to set directmedia=nonat.
+;
+;directmedia=yes                ; Asterisk by default tries to redirect the
+                                ; RTP media stream to go directly from
+                                ; the caller to the callee.  Some devices do not
+                                ; support this (especially if one of them is behind a NAT).
+                                ; The default setting is YES. If you have all clients
+                                ; behind a NAT, or for some other reason want Asterisk to
+                                ; stay in the audio path, you may want to turn this off.
+
+                                ; This setting also affect direct RTP
+                                ; at call setup (a new feature in 1.4 - setting up the
+                                ; call directly between the endpoints instead of sending
+                                ; a re-INVITE).
+
+                                ; Additionally this option does not disable all reINVITE operations.
+                                ; It only controls Asterisk generating reINVITEs for the specific
+                                ; purpose of setting up a direct media path. If a reINVITE is
+                                ; needed to switch a media stream to inactive (when placed on
+                                ; hold) or to T.38, it will still be done, regardless of this 
+                                ; setting. Note that direct T.38 is not supported.
+
+;directmedia=nonat              ; An additional option is to allow media path redirection
+                                ; (reinvite) but only when the peer where the media is being
+                                ; sent is known to not be behind a NAT (as the RTP core can
+                                ; determine it based on the apparent IP address the media
+                                ; arrives from).
+
+;directmedia=update             ; Yet a third option... use UPDATE for media path redirection,
+                                ; instead of INVITE. This can be combined with 'nonat', as
+                                ; 'directmedia=update,nonat'. It implies 'yes'.
+
+;directmedia=outgoing           ; When sending directmedia reinvites, do not send an immediate
+                                ; reinvite on an incoming call leg. This option is useful when
+                                ; peered with another SIP user agent that is known to send
+                                ; immediate direct media reinvites upon call establishment. Setting
+                                ; the option in this situation helps to prevent potential glares.
+                                ; Setting this option implies 'yes'.
+
+;directrtpsetup=yes             ; Enable the new experimental direct RTP setup. This sets up
+                                ; the call directly with media peer-2-peer without re-invites.
+                                ; Will not work for video and cases where the callee sends
+                                ; RTP payloads and fmtp headers in the 200 OK that does not match the
+                                ; callers INVITE. This will also fail if directmedia is enabled when
+                                ; the device is actually behind NAT.
+
+;directmediadeny=0.0.0.0/0      ; Use directmediapermit and directmediadeny to restrict 
+;directmediapermit=172.16.0.0/16; which peers should be able to pass directmedia to each other
+                                ; (There is no default setting, this is just an example)
+                                ; Use this if some of your phones are on IP addresses that
+                                ; can not reach each other directly. This way you can force 
+                                ; RTP to always flow through asterisk in such cases.
+;directmediaacl=acl_example     ; Use named ACLs defined in acl.conf
+
+;ignoresdpversion=yes           ; By default, Asterisk will honor the session version
+                                ; number in SDP packets and will only modify the SDP
+                                ; session if the version number changes. This option will
+                                ; force asterisk to ignore the SDP session version number
+                                ; and treat all SDP data as new data.  This is required
+                                ; for devices that send us non standard SDP packets
+                                ; (observed with Microsoft OCS). By default this option is
+                                ; off.
+
+;sdpsession=Asterisk PBX        ; Allows you to change the SDP session name string, (s=)
+                                ; Like the useragent parameter, the default user agent string
+                                ; also contains the Asterisk version.
+;sdpowner=root                  ; Allows you to change the username field in the SDP owner string, (o=)
+                                ; This field MUST NOT contain spaces
+;encryption=no                  ; Whether to offer SRTP encrypted media (and only SRTP encrypted media)
+                                ; on outgoing calls to a peer. Calls will fail with HANGUPCAUSE=58 if
+                                ; the peer does not support SRTP. Defaults to no.
+;encryption_taglen=80           ; Set the auth tag length offered in the INVITE either 32/80 default 80
+;
+;avpf=yes                       ; Enable inter-operability with media streams using the AVPF RTP profile.
+				; This will cause all offers and answers to use AVPF (or SAVPF). This
+				; option may be specified at the global or peer scope.
+;force_avp=yes			; Force 'RTP/AVP', 'RTP/AVPF', 'RTP/SAVP', and 'RTP/SAVPF' to be used for
+				; media streams when appropriate, even if a DTLS stream is present.
+;----------------------------------------- REALTIME SUPPORT ------------------------
+; For additional information on ARA, the Asterisk Realtime Architecture,
+; please read https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
+;
+;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 SIP UA registers successfully, the ip address,
+                                ; the origination port, the registration period, and the username of
+                                ; the UA will be set to database via realtime.
+                                ; If not present, defaults to 'yes'. Note: realtime peers will
+                                ; probably not function across reloads in the way that you expect, if
+                                ; you turn this option off.
+;rtautoclear=yes                ; Auto-Expire friends created on the fly on the same schedule
+                                ; as if it had just registered? (yes|no|<seconds>)
+                                ; If set to yes, when the registration expires, the friend will
+                                ; vanish from the configuration until requested again. If set
+                                ; to an integer, friends expire within this number of seconds
+                                ; instead of the registration interval.
+
+;ignoreregexpire=yes            ; Enabling this setting has two functions:
+                                ;
+                                ; For non-realtime peers, when their registration expires, the
+                                ; information will _not_ be removed from memory or the Asterisk database
+                                ; if you attempt to place a call to the peer, the existing information
+                                ; will be used in spite of it having expired
+                                ;
+                                ; For realtime peers, when the peer is retrieved from realtime storage,
+                                ; the registration information will be used regardless of whether
+                                ; it has expired or not; if it expires while the realtime peer
+                                ; is still in memory (due to caching or other reasons), the
+                                ; information will not be removed from realtime storage
+
+;----------------------------------------- SIP DOMAIN SUPPORT ------------------------
+; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
+; domains, each of which can direct the call to a specific context if desired.
+; By default, all domains are accepted and sent to the default context or the
+; context associated with the user/peer placing the call.
+; REGISTER to non-local domains will be automatically denied if a domain
+; list is configured.
+;
+; Domains can be specified using:
+; domain=<domain>[,<context>]
+; Examples:
+; domain=myasterisk.dom
+; domain=customer.com,customer-context
+;
+; In addition, all the 'default' domains associated with a server should be
+; added if incoming request filtering is desired.
+; autodomain=yes
+;
+; To disallow requests for domains not serviced by this server:
+; allowexternaldomains=no
+
+;domain=mydomain.tld,mydomain-incoming
+                                ; Add domain and configure incoming context
+                                ; for external calls to this domain
+;domain=1.2.3.4                 ; Add IP address as local domain
+                                ; You can have several "domain" settings
+;allowexternaldomains=no        ; Disable INVITE and REFER to non-local domains
+                                ; Default is yes
+;autodomain=yes                 ; Turn this on to have Asterisk add local host
+                                ; name and local IP to domain list.
+
+; fromdomain=mydomain.tld       ; When making outbound SIP INVITEs to
+                                ; non-peers, use your primary domain "identity"
+                                ; for From: headers instead of just your IP
+                                ; address. This is to be polite and
+                                ; it may be a mandatory requirement for some
+                                ; destinations which do not have a prior
+                                ; account relationship with your server.
+
+;------------------------------ Advice of Charge CONFIGURATION --------------------------
+; snom_aoc_enabled = yes;     ; This options turns on and off support for sending AOC-D and
+                              ; AOC-E to snom endpoints.  This option can be used both in the
+                              ; peer and global scope.  The default for this option is off.
+
+
+;------------------------------ 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".
+
+;-----------------------------------------------------------------------------------
+
+[authentication]
+; Global credentials for outbound calls, i.e. when a proxy challenges your
+; Asterisk server for authentication. These credentials override
+; any credentials in peer/register definition if realm is matched.
+;
+; This way, Asterisk can authenticate for outbound calls to other
+; realms. We match realm on the proxy challenge and pick an set of
+; credentials from this list
+; Syntax:
+;        auth = <user>:<secret>@<realm>
+;        auth = <user>#<md5secret>@<realm>
+; Example:
+;auth=mark:topsecret@digium.com
+;
+; You may also add auth= statements to [peer] definitions
+; Peer auth= override all other authentication settings if we match on realm
+
+;------------------------------------------------------------------------------
+; DEVICE CONFIGURATION
+;
+; SIP entities have a 'type' which determines their roles within Asterisk.
+; * For entities with 'type=peer':
+;   Peers handle both inbound and outbound calls and are matched by ip/port, so for
+;   The case of incoming calls from the peer, the IP address must match in order for
+;   The invitation to work. This means calls made from either direction won't work if
+;   The peer is unregistered while host=dynamic or if the host is otherise not set to
+;   the correct IP of the sender.
+; * For entities with 'type=user':
+;   Asterisk users handle inbound calls only (meaning they call Asterisk, Asterisk can't
+;   call them) and are matched by their authorization information (authname and secret).
+;   Asterisk doesn't rely on their IP and will accept calls regardless of the host setting
+;   as long as the incoming SIP invite authorizes successfully.
+; * For entities with 'type=friend':
+;   Asterisk will create the entity as both a friend and a peer. Asterisk will accept
+;   calls from friends like it would for users, requiring only that the authorization
+;   matches rather than the IP address. Since it is also a peer, a friend entity can
+;   be called as long as its IP is known to Asterisk. In the case of host=dynamic,
+;   this means it is necessary for the entity to register before Asterisk can call it.
+; 
+; Use remotesecret for outbound authentication, and secret for authenticating
+; inbound requests. For historical reasons, if no remotesecret is supplied for an
+; outbound registration or call, the secret will be used. 
+;
+; For device names, we recommend using only a-z, numerics (0-9) and underscore
+;
+; For local phones, type=friend works most of the time
+;
+; If you have one-way audio, you probably have NAT problems.
+; If Asterisk is on a public IP, and the phone is inside of a NAT device
+; you will need to configure nat option for those phones.
+; Also, turn on qualify=yes to keep the nat session open
+;
+; Configuration options available
+; --------------------
+; context
+; callingpres
+; permit
+; deny
+; secret
+; md5secret
+; remotesecret
+; transport
+; dtmfmode
+; directmedia
+; nat
+; callgroup
+; pickupgroup
+; language
+; allow
+; disallow
+; autoframing
+; insecure
+; trustrpid
+; trust_id_outbound
+; progressinband
+; promiscredir
+; useclientcode
+; accountcode
+; setvar
+; callerid
+; amaflags
+; callcounter
+; busylevel
+; allowoverlap
+; allowsubscribe
+; allowtransfer
+; ignoresdpversion
+; subscribecontext
+; template
+; videosupport
+; maxcallbitrate
+; rfc2833compensate
+; Note: app_voicemail mailboxes must be in the form of mailbox@context.
+; mailbox
+; session-timers
+; session-expires
+; session-minse
+; session-refresher
+; t38pt_usertpsource
+; regexten
+; fromdomain
+; fromuser
+; host
+; port
+; qualify
+; keepalive
+; defaultip
+; defaultuser
+; rtptimeout
+; rtpholdtimeout
+; sendrpid
+; outboundproxy
+; rfc2833compensate
+; callbackextension
+; timert1
+; timerb
+; qualifyfreq
+; t38pt_usertpsource
+; contactpermit         ; Limit what a host may register as (a neat trick
+; contactdeny           ; is to register at the same IP as a SIP provider,
+; contactacl            ; then call oneself, and get redirected to that
+;                       ; same location).
+; directmediapermit
+; directmediadeny
+; directmediaacl
+; unsolicited_mailbox
+; use_q850_reason
+; maxforwards
+; encryption
+; description		; Used to provide a description of the peer in console output
+; dtlsenable
+; dtlsverify
+; dtlsrekey
+; dtlscertfile
+; dtlsprivatekey
+; dtlscipher
+; dtlscafile
+; dtlscapath
+; dtlssetup
+; dtlsfingerprint
+; ignore_requested_pref ; Ignore the requested codec and determine the preferred codec
+;						; from the peer's configuration.
+;
+
+;------------------------------------------------------------------------------
+; DTLS-SRTP CONFIGURATION
+;
+; DTLS-SRTP support is available if the underlying RTP engine in use supports it.
+;
+; dtlsenable = yes                   ; Enable or disable DTLS-SRTP support
+; dtlsverify = yes                   ; Verify that provided peer certificate and fingerprint are valid
+;				     ; A value of 'yes' will perform both certificate and fingerprint verification
+;				     ; A value of 'no' will perform no certificate or fingerprint verification
+;				     ; A value of 'fingerprint' will perform ONLY fingerprint verification
+;				     ; A value of 'certificate' will perform ONLY certficiate verification
+; dtlsrekey = 60                     ; Interval at which to renegotiate the TLS session and rekey the SRTP session
+;                                    ; If this is not set or the value provided is 0 rekeying will be disabled
+; dtlscertfile = file                ; Path to certificate file to present
+; dtlsprivatekey = file              ; Path to private key for certificate file
+; dtlscipher = <SSL cipher string>   ; Cipher to use for TLS negotiation
+;                                    ; A list of valid SSL cipher strings can be found at:
+;                                    ; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
+; dtlscafile = file                  ; Path to certificate authority certificate
+; dtlscapath = path                  ; Path to a directory containing certificate authority certificates
+; dtlssetup = actpass                ; Whether we are willing to accept connections, connect to the other party, or both.
+;                                    ; Valid options are active (we want to connect to the other party), passive (we want to
+;                                    ; accept connections only), and actpass (we will do both). This value will be used in
+;                                    ; the outgoing SDP when offering and for incoming SDP offers when the remote party sends
+;                                    ; actpass
+; dtlsfingerprint = sha-1            ; The hash to use for the fingerprint in SDP (valid options are sha-1 and sha-256)
+
+;[sip_proxy]
+; For incoming calls only. Example: FWD (Free World Dialup)
+; We match on IP address of the proxy for incoming calls
+; since we can not match on username (caller id)
+;type=peer
+;context=from-fwd
+;host=fwd.pulver.com
+
+;[sip_proxy-out]
+;type=peer                        ; we only want to call out, not be called
+;remotesecret=guessit             ; Our password to their service
+;defaultuser=yourusername         ; Authentication user for outbound proxies
+;fromuser=yourusername            ; Many SIP providers require this!
+;fromdomain=provider.sip.domain
+;host=box.provider.com
+;transport=udp,tcp                ; This sets the default transport type to udp for outgoing, and will
+;                                 ; accept both tcp and udp. The default transport type is only used for
+;                                 ; outbound messages until a Registration takes place.  During the
+;                                 ; peer Registration the transport type may change to another supported
+;                                 ; type if the peer requests so.
+
+;usereqphone=yes                  ; This provider requires ";user=phone" on URI
+;callcounter=yes                  ; Enable call counter
+;busylevel=2                      ; Signal busy at 2 or more calls
+;outboundproxy=proxy.provider.domain  ; send outbound signaling to this proxy, not directly to the peer
+;port=80                          ; The port number we want to connect to on the remote side
+                                  ; Also used as "defaultport" in combination with "defaultip" settings
+
+;--- sample definition for a provider
+;[provider1]
+;type=peer
+;host=sip.provider1.com
+;fromuser=4015552299              ; how your provider knows you
+;remotesecret=youwillneverguessit ; The password we use to authenticate to them
+;secret=gissadetdu                ; The password they use to contact us
+;callbackextension=123            ; Register with this server and require calls coming back to this extension
+;transport=udp,tcp                ; This sets the transport type to udp for outgoing, and will
+;                                 ;   accept both tcp and udp. Default is udp. The first transport
+;                                 ;   listed will always be used for outgoing connections.
+;unsolicited_mailbox=4015552299   ; If the remote SIP server sends an unsolicited MWI NOTIFY message the new/old
+;                                 ;   message count will be stored in the configured virtual mailbox. It can be used
+;                                 ;   by any device supporting MWI by specifying <configured value>@SIP_Remote as the
+;                                 ;   mailbox.
+
+;
+; Because you might have a large number of similar sections, it is generally
+; convenient to use templates for the common parameters, and add them
+; the the various sections. Examples are below, and we can even leave
+; the templates uncommented as they will not harm:
+
+[basic-options](!)                ; a template
+        dtmfmode=rfc2833
+        context=from-office
+        type=friend
+
+[natted-phone](!,basic-options)   ; another template inheriting basic-options
+        directmedia=no
+        host=dynamic
+
+[public-phone](!,basic-options)   ; another template inheriting basic-options
+        directmedia=yes
+
+[my-codecs](!)                    ; a template for my preferred codecs
+        disallow=all
+        allow=ilbc
+        allow=g729
+        allow=gsm
+        allow=g723
+        allow=ulaw
+        ; Or, more simply:
+        ;allow=!all,ilbc,g729,gsm,g723,ulaw
+
+[ulaw-phone](!)                   ; and another one for ulaw-only
+        disallow=all
+        allow=ulaw
+        ; Again, more simply:
+        ;allow=!all,ulaw
+
+; and finally instantiate a few phones
+;
+; [2133](natted-phone,my-codecs)
+;        secret = peekaboo
+; [2134](natted-phone,ulaw-phone)
+;        secret = not_very_secret
+; [2136](public-phone,ulaw-phone)
+;        secret = not_very_secret_either
+; ...
+;
+
+; Standard configurations not using templates look like this:
+;
+;[grandstream1]
+;type=friend
+;context=from-sip                ; Where to start in the dialplan when this phone calls
+;recordonfeature=dynamicfeature1 ; Feature to use when INFO with Record: on is received.
+;recordofffeature=dynamicfeature2 ; Feature to use when INFO with Record: off is received.
+;callerid=John Doe <1234>        ; Full caller ID, to override the phones config
+                                 ; on incoming calls to Asterisk
+;description=Courtesy Phone      ; Description of the peer. Shown when doing 'sip show peers'.
+;host=192.168.0.23               ; we have a static but private IP address
+                                 ; No registration allowed
+;directmedia=yes                 ; allow RTP voice traffic to bypass Asterisk
+;dtmfmode=info                   ; either RFC2833 or INFO for the BudgeTone
+;call-limit=1                    ; permit only 1 outgoing call and 1 incoming call at a time
+                                 ; from the phone to asterisk (deprecated)
+                                 ; 1 for the explicit peer, 1 for the explicit user,
+                                 ; remember that a friend equals 1 peer and 1 user in
+                                 ; memory
+                                 ; There is no combined call counter for a "friend"
+                                 ; so there's currently no way in sip.conf to limit
+                                 ; to one inbound or outbound call per phone. Use
+                                 ; the group counters in the dial plan for that.
+                                 ;
+;mailbox=1234@default            ; mailbox 1234 in voicemail context "default"
+;disallow=all                    ; need to disallow=all before we can use allow=
+;allow=ulaw                      ; Note: In user sections the order of codecs
+                                 ; listed with allow= does NOT matter!
+;allow=alaw
+;allow=g723.1                    ; Asterisk only supports g723.1 pass-thru!
+;allow=g729                      ; Pass-thru only unless g729 license obtained
+;callingpres=allowed_passed_screen ; Set caller ID presentation
+                                 ; See function CALLERPRES documentation for possible
+                                 ; values.
+
+;[xlite1]
+; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
+; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
+;type=friend
+;regexten=1234                   ; When they register, create extension 1234
+;callerid="Jane Smith" <5678>
+;host=dynamic                    ; This device needs to register
+;directmedia=no                  ; Typically set to NO if behind NAT
+;disallow=all
+;allow=gsm                       ; GSM consumes far less bandwidth than ulaw
+;allow=ulaw
+;allow=alaw
+;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes
+;registertrying=yes              ; Send a 100 Trying when the device registers.
+
+;[snom]
+;type=friend                     ; Friends place calls and receive calls
+;context=from-sip                ; Context for incoming calls from this user
+;secret=blah
+;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
+;language=de                     ; Use German prompts for this user
+;host=dynamic                    ; This peer register with us
+;dtmfmode=inband                 ; Choices are inband, rfc2833, or info
+;defaultip=192.168.0.59          ; IP used until peer registers
+;mailbox=1234@context,2345@context ; Mailbox(-es) for message waiting indicator
+;subscribemwi=yes                ; Only send notifications if this phone
+                                 ; subscribes for mailbox notification
+;vmexten=voicemail               ; dialplan extension to reach mailbox
+                                 ; sets the Message-Account in the MWI notify message
+                                 ; defaults to global vmexten which defaults to "asterisk"
+;disallow=all
+;allow=ulaw                      ; dtmfmode=inband only works with ulaw or alaw!
+
+
+;[polycom]
+;type=friend                     ; Friends place calls and receive calls
+;context=from-sip                ; Context for incoming calls from this user
+;secret=blahpoly
+;host=dynamic                    ; This peer register with us
+;dtmfmode=rfc2833                ; Choices are inband, rfc2833, or info
+;defaultuser=polly               ; Username to use in INVITE until peer registers
+;defaultip=192.168.40.123
+                                 ; Normally you do NOT need to set this parameter
+;disallow=all
+;allow=ulaw                      ; dtmfmode=inband only works with ulaw or alaw!
+;progressinband=no               ; Polycom phones don't work properly with "never"
+
+
+;[pingtel]
+;type=friend
+;secret=blah
+;host=dynamic
+;insecure=port                   ; Allow matching of peer by IP address without
+                                 ; matching port number
+;insecure=invite                 ; Do not require authentication of incoming INVITEs
+;insecure=port,invite            ; (both)
+;qualify=1000                    ; Consider it down if it's 1 second to reply
+                                 ; Helps with NAT session
+                                 ; qualify=yes uses default value
+;qualifyfreq=60                  ; Qualification: How often to check for the
+                                 ; host to be up in seconds
+                                 ; Set to low value if you use low timeout for
+                                 ; NAT of UDP sessions
+;
+; Call group and Pickup group should be in the range from 0 to 63
+;
+;callgroup=1,3-4                 ; We are in caller groups 1,3,4
+;pickupgroup=1,3-5               ; We can do call pick-p for call group 1,3,4,5
+;namedcallgroup=engineering,sales,netgroup,protgroup ; We are in named call groups engineering,sales,netgroup,protgroup
+;namedpickupgroup=sales          ; We can do call pick-p for named call group sales
+;defaultip=192.168.0.60          ; IP address to use if peer has not registered
+;deny=0.0.0.0/0.0.0.0            ; ACL: Control access to this account based on IP address
+;permit=192.168.0.60/255.255.255.0
+;permit=192.168.0.60/24          ; we can also use CIDR notation for subnet masks
+;permit=2001:db8::/32            ; IPv6 ACLs can be specified if desired. IPv6 ACLs
+                                 ; apply only to IPv6 addresses, and IPv4 ACLs apply
+                                 ; only to IPv4 addresses.
+;acl=named_acl_example           ; Use named ACLs defined in acl.conf
+
+;[cisco1]
+;type=friend
+;secret=blah
+;qualify=200                     ; Qualify peer is no more than 200ms away
+;host=dynamic                    ; This device registers with us
+;directmedia=no                  ; Asterisk by default tries to redirect the
+                                 ; RTP media stream (audio) to go directly from
+                                 ; the caller to the callee.  Some devices do not
+                                 ; support this (especially if one of them is
+                                 ; behind a NAT).
+;defaultip=192.168.0.4           ; IP address to use until registration
+;defaultuser=goran               ; Username to use when calling this device before registration
+                                 ; Normally you do NOT need to set this parameter
+;setvar=CUSTID=5678              ; Channel variable to be set for all calls from or to this device
+;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep   ; This channel variable will
+                                                ; cause the given audio file to
+                                                ; be played upon completion of
+                                                ; an attended transfer to the
+                                                ; target of the transfer.
+
+;[pre14-asterisk]
+;type=friend
+;secret=digium
+;host=dynamic
+;rfc2833compensate=yes          ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
+                                ; You must have this turned on or DTMF reception will work improperly.
+;t38pt_usertpsource=yes         ; Use the source IP address of RTP as the destination IP address for UDPTL packets
+                                ; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
+                                ; external IP address of the remote device. If port forwarding is done at the client side
+                                ; then UDPTL will flow to the remote device.
diff --git a/configs/samples/sip_notify.conf.sample b/configs/samples/sip_notify.conf.sample
new file mode 100644
index 000000000..8224ee1ff
--- /dev/null
+++ b/configs/samples/sip_notify.conf.sample
@@ -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
diff --git a/configs/samples/skinny.conf.sample b/configs/samples/skinny.conf.sample
new file mode 100644
index 000000000..be88dc230
--- /dev/null
+++ b/configs/samples/skinny.conf.sample
@@ -0,0 +1,208 @@
+;
+; 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=
+
+; The imeddialkey option allows for a key to be used to immediately dial the already
+; entered number. This is useful where the dialplan includes variable length pattern
+; matching. Valid options are '#' and '*'. On devices with soft buttons, a button will
+; be available to immediately dial when a pattern than can be dialed has been entered.
+; Default is unset, that is no immediated dial key (softbutton still exists).
+;
+;immeddialkey=#
+
+; 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.
+;callfwdtimeout=20000        ; ms before cfwd_noans occurs (default 20 secs)
+;------------------------------- SPECIFIC LINE OPTIONS -----------------------------
+;setvar=        	     ; allows for the setting of chanvars.
+;-----------------------------------------------------------------------------------
+
+;[100]
+;nat=yes
+;callerid="Customer Support" <810-234-1212>
+; Note: app_voicemail mailboxes must be in the form of mailbox@context.
+;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 to the
+                                                ; target of the 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)
+
+; Service URLs attached to line buttons (eg phone directory)
+; See http://www.voip-info.org/wiki/view/Asterisk+Cisco+79XX+XML+Services
+; for intro to xml structure.
+;serviceurl=Directory,http://host/file.xml
+
+
+; 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
+
diff --git a/configs/samples/sla.conf.sample b/configs/samples/sla.conf.sample
new file mode 100644
index 000000000..b5e1ef614
--- /dev/null
+++ b/configs/samples/sla.conf.sample
@@ -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
+; --------------------------------------
+
diff --git a/configs/samples/smdi.conf.sample b/configs/samples/smdi.conf.sample
new file mode 100644
index 000000000..cca0a5cb9
--- /dev/null
+++ b/configs/samples/smdi.conf.sample
@@ -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
diff --git a/configs/samples/sorcery.conf.sample b/configs/samples/sorcery.conf.sample
new file mode 100644
index 000000000..7406214fb
--- /dev/null
+++ b/configs/samples/sorcery.conf.sample
@@ -0,0 +1,67 @@
+; Sample configuration file for Sorcery Data Access Layer
+
+;
+; Wizards
+;
+; Wizards are the persistence mechanism for objects. They are loaded as Asterisk modules and register
+; themselves with the sorcery core. All implementation specific details of how objects are persisted is isolated
+; within wizards.
+;
+
+;
+; Caching
+;
+; A wizard can optionally be marked as an object cache by adding "/cache" to the object type within the mapping.
+; If an object is returned from a non-object cache it is immediately given to the cache to be created. Multiple
+; object caches can be configured for a single object type.
+;
+
+;
+; Object Type Mappings
+;
+; To allow configuration of where and how an object is persisted object mappings can be defined within this file
+; on a per-module basis. The mapping consists of the object type, options, wizard name, and wizard configuration
+; data. This has the following format:
+;
+; object type [/options] = wizard name, wizard configuration data
+;
+; For example to configure an in-memory wizard for the 'bob' object type:
+;
+; bob = memory
+;
+; Or to configure the object type 'joe' from a configuration file:
+;
+; joe = config,joe.conf
+;
+; Note that an object type can have multiple mappings defined. Each mapping will be consulted in the order in which
+; it appears within the configuration file. This means that if you are configuring a wizard as a cache it should
+; appear as the first mapping so the cache is consulted before all other mappings.
+;
+
+;
+; The following object mappings are used by the unit test to test certain functionality of sorcery.
+;
+[test_sorcery_section]
+test=memory
+
+[test_sorcery_cache]
+test/cache=test
+test=memory
+
+;
+; The following object mapping is the default mapping of external MWI mailbox
+; objects to give persistence to the message counts.
+;
+;[res_mwi_external]
+;mailboxes=astdb,mwi_external
+
+;
+; The following object mappings set PJSIP objects to use realtime database mappings from extconfig
+; with the table names used when automatically generating configuration from the alembic script.
+;
+;[res_pjsip]
+;endpoint=realtime,ps_endpoints
+;auth=realtime,ps_auths
+;aor=realtime,ps_aors
+;domain_alias=realtime,ps_domain_aliases
+;identify=realtime,ps_endpoint_id_ips
diff --git a/configs/samples/ss7.timers.sample b/configs/samples/ss7.timers.sample
new file mode 100644
index 000000000..9cf9bd1ab
--- /dev/null
+++ b/configs/samples/ss7.timers.sample
@@ -0,0 +1,65 @@
+;;;;;	ITU-T Q.707 timers
+
+;mtp3_timer.q707_t1 = 4000
+;mtp3_timer.q707_t2 = 30000
+
+;;;;;	MTP3 timers as specified in ITU-T Q.704 or ANSI T1.111-2001
+
+mtp3_timer.t1 = 500
+mtp3_timer.t2 = 700
+mtp3_timer.t3 = 500
+mtp3_timer.t4 = 500
+mtp3_timer.t5 = 500
+mtp3_timer.t6 = 500
+mtp3_timer.t7 = 1000
+
+mtp3_timer.t10 = 60000
+
+mtp3_timer.t12 = 800
+mtp3_timer.t13 = 800
+mtp3_timer.t14 = 2000
+
+;	enable for ITU only. Timers after T17 are defined differently for ANSI
+;mtp3_timer.t19 = 67000
+;mtp3_timer.t21 = 63000
+;
+;mtp3_timer.t22 = 300000
+;mtp3_timer.t23 = 300000
+
+
+;;;;;	ISUP timers as specified in ITU-T Q.764 or ANSI T1.113-2000
+
+isup_timer.t1 = 15000
+;isup_timer.t2 = 180000		; ITU only
+
+;isup_timer.t5 = 300000		; for ITU
+;isup_timer.t5 = 60000		; for ANSI
+isup_timer.t6 = 30000
+isup_timer.t7 = 20000
+isup_timer.t8 = 10000
+
+;isup_timer.t10 = 4000		; ITU only
+
+isup_timer.t12 = 15000
+;isup_timer.t13 = 300000	; for ITU
+;isup_timer.t13 = 60000		; for ANSI
+isup_timer.t14 = 15000
+;isup_timer.t15 = 300000	; for ITU
+;isup_timer.t15 = 60000		; for ANSI
+isup_timer.t16 = 15000
+;isup_timer.t17 = 300000	; for ITU
+;isup_timer.t17 = 60000		; for ANSI
+isup_timer.t18 = 15000
+;isup_timer.t19 = 300000	; for ITU
+;isup_timer.t19 = 60000		; for ANSI
+isup_timer.t20 = 15000
+;isup_timer.t21 = 300000	; for ITU
+;isup_timer.t21 = 60000		; for ANSI
+isup_timer.t22 = 15000
+;isup_timer.t23 = 300000	; for ITU
+;isup_timer.t23 = 60000		; for ANSI
+
+isup_timer.t27 = 240000
+
+isup_timer.t33 = 12000
+;isup_timer.t35 = 15000		; ITU only
diff --git a/configs/samples/statsd.conf.sample b/configs/samples/statsd.conf.sample
new file mode 100644
index 000000000..806097300
--- /dev/null
+++ b/configs/samples/statsd.conf.sample
@@ -0,0 +1,8 @@
+[general]
+;enabled = yes			; When set to yes, statsd support is enabled
+;server = 127.0.0.1		; server[:port] of statsd server to use.
+				; If not specified, the port is 8125
+;prefix =			; Prefix to prepend to all metrics
+;add_newline = no		; Append a newline to every event. This is
+				; useful if you want to run a fake statsd
+				; server using netcat (nc -lu 8125)
diff --git a/configs/samples/telcordia-1.adsi b/configs/samples/telcordia-1.adsi
new file mode 100644
index 000000000..1486aa95e
--- /dev/null
+++ b/configs/samples/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
diff --git a/configs/samples/test_sorcery.conf.sample b/configs/samples/test_sorcery.conf.sample
new file mode 100644
index 000000000..c465dbf74
--- /dev/null
+++ b/configs/samples/test_sorcery.conf.sample
@@ -0,0 +1,14 @@
+; This is a res_sorcery_config compatible file for the sorcery unit tests
+
+[hey]
+bob=98
+joe=41
+
+[hey2]
+type=zombies
+bob=97
+joe=40
+
+[hey3]
+bob=96
+joe=39
diff --git a/configs/samples/udptl.conf.sample b/configs/samples/udptl.conf.sample
new file mode 100644
index 000000000..1d635da5c
--- /dev/null
+++ b/configs/samples/udptl.conf.sample
@@ -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
diff --git a/configs/samples/unistim.conf.sample b/configs/samples/unistim.conf.sample
new file mode 100644
index 000000000..c33426b0c
--- /dev/null
+++ b/configs/samples/unistim.conf.sample
@@ -0,0 +1,88 @@
+;
+; 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=0                ; 0 (default) = 31Jan, 1 = Jan31, 2 = month/day, 3 = 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
+;language=ru                 ; language used for audio files and onscreen messages translate
+;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
+;dtmf_duration=0             ; DTMF playback duration (in milliseconds) 0..150 (0 = off (default), 150 = maximum)
+;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
diff --git a/configs/samples/users.conf.sample b/configs/samples/users.conf.sample
new file mode 100644
index 000000000..50b80a1c5
--- /dev/null
+++ b/configs/samples/users.conf.sample
@@ -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
+
diff --git a/configs/samples/voicemail.conf.sample b/configs/samples/voicemail.conf.sample
new file mode 100644
index 000000000..f3d7d4f44
--- /dev/null
+++ b/configs/samples/voicemail.conf.sample
@@ -0,0 +1,469 @@
+;
+; Voicemail Configuration
+;
+
+; ********* NOTICE ************************************************************
+;
+; NOTE: Asterisk has to edit this file to change a user's password.  This does
+; not currently work with the "#include <file>" directive for Asterisk
+; configuration files, nor when using realtime static configuration.
+; Do not use them with this configuration file.
+;
+; NOTE: Mailboxes defined by app_voicemail MUST be referenced by the rest
+; of the system as mailbox@context.  The rest of the system cannot add
+; @default to mailbox identifiers for app_voicemail that do not specify a
+; context any longer.  It is a mailbox identifier format that should only
+; be interpreted by app_voicemail.
+;
+; ********* NOTICE ************************************************************
+
+[general]
+; Formats for writing Voicemail.  Note that when using IMAP storage for
+; voicemail, only the first format specified will be used.
+;format=g723sf|wav49|wav
+format=wav49|gsm|wav
+;
+; WARNING:
+; If you change the list of formats that you record voicemail in
+; when you have mailboxes that contain messages, you _MUST_ absolutely
+; manually go through those mailboxes and convert/delete/add the
+; the message files so that they appear to have been stored using
+; your new format list. If you don't do this, very unpleasant
+; things may happen to your users while they are retrieving and
+; manipulating their voicemail.
+;
+; In other words: don't change the format list on a production system
+; unless you are _VERY_  sure that you know what you are doing and are
+; prepared for the consequences.
+;
+; Who the e-mail notification should appear to come from
+serveremail=asterisk
+;serveremail=asterisk@linux-support.net
+; Should the email contain the voicemail as an attachment
+attach=yes
+; Maximum number of messages per folder.  If not specified, a default value
+; (100) is used.  Maximum value for this option is 9999.  If set to 0, a
+; mailbox will be greetings-only.
+;maxmsg=100
+; Maximum length of a voicemail message in seconds
+;maxsecs=180
+; Minimum length of a voicemail message in seconds for the message to be kept
+; The default is no minimum.
+;minsecs=3
+; Maximum length of greetings in seconds
+;maxgreet=60
+; How many milliseconds to skip forward/back when rew/ff in message playback
+skipms=3000
+; 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
+; Max number of failed login attempts
+maxlogins=3
+;
+; Move heard messages to the 'Old' folder automagically.  Defaults to on.
+;moveheard=yes
+;
+; Forward an urgent message as an urgent message.  Defaults to no so
+; sender can set the urgency on the envelope of the forwarded message.
+;forward_urgent_auto=no
+;
+; User context is where entries from users.conf are registered.  The
+; default value is 'default'
+;
+;userscontext=default
+;
+; If you need to have an external program, i.e. /usr/bin/myapp
+; called when a voicemail is left, delivered, or your voicemailbox
+; is checked, uncomment this.
+;externnotify=/usr/bin/myapp
+
+; If you would also like to enable SMDI notification then set smdienable to yes.
+; You will also need to make sure smdiport is set to a valid port as specified in
+; smdi.conf.
+;smdienable=yes
+;smdiport=/dev/ttyS0
+
+; If you need to have an external program, i.e. /usr/bin/myapp
+; called when a voicemail password is changed, uncomment this. The
+; arguments passed to the application are: <context> <mailbox> <newpassword>
+; Note: If this is set, the password will NOT be changed in voicemail.conf
+; If you would like to also change the password in voicemail.conf, use
+; the externpassnotify option below instead.
+;externpass=/usr/bin/myapp
+;externpassnotify=/usr/bin/myapp
+
+; If you would like to have an external program called when a user changes the
+; voicemail password for the purpose of doing validation on the new password,
+; then use this option.  The script can decide whether or not the new password
+; meets minimum password strength requirements before the Voicemail application
+; accepts the password.  If the script decides that the password is not acceptable,
+; the user will be informed that the new password does not meet minimum password
+; requirements, and they will be asked to enter another password.
+;
+; The arguments passed to this script are <mailbox> <context> <old pw> <new pw>.
+;
+; The script should print "VALID" to stdout to indicate that the new password
+; is acceptable.  If the password is considered too weak, the script should print
+; "INVALID" to stdout.
+;
+; There is an example script in the contrib/scripts/ directory, voicemailpwcheck.py,
+; which implements some basic password checking, and can be used as a starting point
+; for use with this option.
+;
+;externpasscheck=/usr/local/bin/voicemailpwcheck.py
+
+; For the directory, you can override the intro file if you want
+;directoryintro=dir-intro
+; The character set for voicemail messages can be specified here
+; default: ISO-8859-1
+;charset=UTF-8
+; The ADSI feature descriptor number to download to
+;adsifdn=0000000F
+; The ADSI security lock code
+;adsisec=9BDBF7AC
+; The ADSI voicemail application version number.
+;adsiver=1
+; Skip the "[PBX]:" string from the message title
+;pbxskip=yes
+; Change the From: string
+;fromstring=The Asterisk PBX
+; Permit finding entries for forward/compose from the directory
+;usedirectory=yes
+; Voicemail can be stored in a database using the ODBC driver.
+; The value of odbcstorage is the database connection configured
+; in res_odbc.conf.
+;odbcstorage=asterisk
+; The default table for ODBC voicemail storage is voicemessages.
+;odbctable=voicemessages
+;
+; Change the from, body and/or subject, variables:
+;     VM_NAME, VM_DUR, VM_MSGNUM, VM_MAILBOX, VM_CALLERID, VM_CIDNUM,
+;     VM_CIDNAME, VM_DATE
+; Additionally, on forwarded messages, you have the variables:
+;     ORIG_VM_CALLERID, ORIG_VM_CIDNUM, ORIG_VM_CIDNAME, ORIG_VM_DATE
+; You can select between two variables by using dialplan functions, e.g.
+;     ${IF(${ISNULL(${ORIG_VM_DATE})}?${VM_DATE}:${ORIG_VM_DATE})}
+;
+; Note: The emailbody config row can only be up to 512 characters due to a
+;       limitation in the Asterisk configuration subsystem.
+;emailsubject=[PBX]: New message ${VM_MSGNUM} 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 ${VM_NAME}:\n\n\tjust wanted to let you know you were just left a ${VM_DUR} long message (number ${VM_MSGNUM})\nin mailbox ${VM_MAILBOX} from ${VM_CALLERID}, on ${VM_DATE}, so you might\nwant to check it when you get a chance.  Thanks!\n\n\t\t\t\t--Asterisk\n
+;
+; Note: ${IF()} strips spacing at the beginning and end of its true and false
+; values, so a newline cannot be placed at either location.  The word 'so' is
+; therefore duplicated, in order for the newline to be interpreted correctly.
+;emailbody=Dear ${VM_NAME}:\n\n\tjust wanted to let you know you were just ${IF($["${VM_CIDNUM}" = "${ORIG_VM_CIDNUM}"]?left:forwarded)} a ${VM_DUR} long message (number ${VM_MSGNUM})\nin mailbox ${VM_MAILBOX} from ${VM_CALLERID}, on ${VM_DATE},\n${IF($["${VM_CIDNUM}" = "${ORIG_VM_CIDNUM}"]?so:(originally sent by ${ORIG_VM_CALLERID} on ${ORIG_VM_DATE})\nso)} you might want to check it when you get a chance.  Thanks!\n\n\t\t\t\t--Asterisk\n
+;
+; 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
+;pagerbody=New ${VM_DUR} long msg in box ${VM_MAILBOX}\nfrom ${VM_CALLERID}, on ${VM_DATE}
+;
+; 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 for pager use
+pagerdateformat=%A, %B %d, %Y at %r
+; Short 24h date format for pager use
+;pagerdateformat=%T %D
+;
+; Using the mailcmd option, you can specify what command is called for
+; outbound E-mail. The default is shown below.
+;
+;mailcmd=/usr/sbin/sendmail -t
+;
+;pollmailboxes=no    ;   If mailboxes are changed anywhere outside of app_voicemail,
+;                    ; then this option must be enabled for MWI to work.  This
+;                    ; enables polling mailboxes for changes.  Normally, it will
+;                    ; expect that changes are only made when someone called in
+;                    ; to one of the voicemail applications.
+;                    ;   Examples of situations that would require this option are
+;                    ; web interfaces to voicemail or an email client in the case
+;                    ; of using IMAP storage.
+;                    ; Default: no
+;pollfreq=30         ;   If the "pollmailboxes" option is enabled, this option
+;                    ; sets the polling frequency.  The default is once every
+;                    ; 30 seconds.
+;
+
+; -----------------------------------------------------------------------------
+; IMAP configuration settings only
+;   These settings are only applicable when Asterisk is compiled with IMAP support.
+;
+;imapgreetings=no        ; If using IMAP storage, specify whether voicemail greetings
+                         ; should be stored via IMAP. If no, then greetings are stored
+                         ; as if IMAP storage were not enabled.
+;greetingsfolder=INBOX   ; If imapgreetings=yes, then specify which folder to store
+                         ; your greetings in. If you do not specify a folder, then INBOX
+                         ; will be used
+;imapparentfolder=INBOX  ; Some IMAP server implementations store folders under INBOX
+                         ; instead of using a top level folder (ex. INBOX/Friends).  In
+                         ; this case, user imapparentfolder to set the parent folder. For
+                         ; example, Cyrus IMAP does NOT use INBOX as the parent. Default
+                         ; is to have no parent folder set.
+;imapserver=localhost    ; The address of the IMAP server
+;imapport=143            ; The port of the IMAP server
+;imapflags=ssl           ; Optional flags to pass to the IMAP server in the IMAP mailbox
+                         ; name.  For example, setting this to 'ssl' will enable OpenSSL
+                         ; encryption, assuming the IMAP libraries were compiled with
+                         ; OpenSSL support.
+;imapfolder=INBOX        ; The folder in which to store voicemail messages on the IMAP
+                         ; server.  By default, they are stored in INBOX.
+;authuser=user           ; The master user to use for connecting to the IMAP server, if
+                         ; the server is configured with a single user that has access to
+                         ; all mailboxes
+;authpassword=password   ; The password for the authuser, if used
+;imapopentimeout=60      ; The TCP open timeout (in seconds)
+;imapclosetimeout=60     ; The TCP close timeout (in seconds)
+;imapreadtimeout=60      ; The TCP read timeout (in seconds)
+;imapwritetimeout=60     ; The TCP write timeout (in seconds)
+
+; -----------------------------------------------------------------------------
+;
+
+; Each mailbox is listed in the form <mailbox>=<password>,<name>,<email>,<pager_email>,<options>
+; if the e-mail is specified, a message will be sent when a message is
+; received, to the given mailbox. If pager is specified, a message will be
+; sent there as well. If the password is prefixed by '-', then it is
+; considered to be unchangeable.
+;
+; Advanced options example is extension 4069
+; NOTE: All options can be expressed globally in the general section, and
+; overridden in the per-mailbox settings, unless listed otherwise.
+;
+; tz=central 		; Timezone from zonemessages below. Irrelevant if envelope=no.
+; locale=de_DE.UTF-8	; set the locale for generation of the date/time strings (make 
+			; sure the locales are installed in your operating system; e.g
+			; on Debian Linux you can use "dpkg-reconfigure locales").
+			; If you use UTF-8 locales, make sure to set the "charset" option
+			; to UTF-8 too. If you mix different locales for different users
+			; you should avoid words in the emaildateformat specification, e.g.:
+			; emaildateformat=%A, %d %B %Y, %H:%M:%S
+; attach=yes 		; Attach the voicemail to the notification email *NOT* the pager email
+; attachfmt=wav49	; Which format to attach to the email.  Normally this is the
+			; first format specified in the format parameter above, but this
+			; option lets you customize the format sent to particular mailboxes.
+			; Useful if Windows users want wav49, but Linux users want gsm.
+			; [per-mailbox only]
+; saycid=yes 		; Say the caller id information before the message. If not described,
+			;     or set to no, it will be in the envelope. When enabled, if a recorded file 
+			;     with the same name as the caller id exists in 
+			;     <astspooldir>/recordings/callerids, then that file will be played as a name
+			;     rather than saying each digit as a phone number.
+; cidinternalcontexts=intern	; Internal Context for Name Playback instead of
+			; extension digits when saying caller id.
+; sayduration=no 	; Turn on/off the duration information before the message. [ON by default]
+; saydurationm=2        ; Specify the minimum duration to say. Default is 2 minutes
+; dialout=fromvm ; Context to dial out from [option 4 from mailbox's advanced menu].
+                 ; If not specified, option 4 will not be listed and dialing out
+                 ; from within VoiceMailMain() will not be permitted.
+sendvoicemail=yes ; Allow the user to compose and send a voicemail while inside
+                  ; VoiceMailMain() [option 5 from mailbox's advanced menu].
+                  ; If set to 'no', option 5 will not be listed.
+; searchcontexts=yes	; Current default behavior is to search only the default context
+			; if one is not specified.  The older behavior was to search all contexts.
+			; This option restores the old behavior [DEFAULT=no]
+			; Note: If you have this option enabled, then you will be required to have
+			; unique mailbox names across all contexts. Otherwise, an ambiguity is created
+			; since it is impossible to know which mailbox to retrieve when one is requested.
+; callback=fromvm 	; Context to call back from
+			;     if not listed, calling the sender back will not be permitted
+; exitcontext=fromvm    ; Context to go to on user exit such as * or 0
+                        ;     The default is the current context.
+; review=yes 		; Allow sender to review/rerecord their message before saving it [OFF by default
+; operator=yes      ; Allow sender to hit 0 before/after/during leaving a voicemail to
+                    ; reach an operator.  This option REQUIRES an 'o' extension in the
+                    ; same context (or in exitcontext, if set), as that is where the
+                    ; 0 key will send you.  [OFF by default]
+; envelope=no 		; Turn on/off envelope playback before message playback. [ON by default]
+			;     This does NOT affect option 3,3 from the advanced options menu
+; delete=yes		; After notification, the voicemail is deleted from the server. [per-mailbox only]
+			;     This is intended for use with users who wish to receive their
+			;     voicemail ONLY by email. Note:  "deletevoicemail" is provided as an
+			;     equivalent option for Realtime configuration.
+; alias=Bongo       ; Use this additional string for comparison while looking
+                    ; for a match in the Directory application.  This option
+                    ; may be specified multiple times to specify additional
+                    ; strings [per-mailbox only]
+; volgain=0.0		; Emails bearing the voicemail may arrive in a volume too
+			;     quiet to be heard.  This parameter allows you to specify how
+			;     much gain to add to the message when sending a voicemail.
+			;     NOTE: sox must be installed for this option to work.
+; nextaftercmd=yes	; Skips to the next message after hitting 7 or 9 to delete/save current message.
+; forcename=yes		; Forces a new user to record their name.  A new user is
+			;     determined by the password being the same as
+			;     the mailbox number.  The default is "no".
+; forcegreetings=no	; This is the same as forcename, except for recording
+			;     greetings.  The default is "no".
+; hidefromdir=yes	; Hide this mailbox from the directory produced by app_directory
+			;     The default is "no".
+; tempgreetwarn=yes	; Remind the user that their temporary greeting is set
+
+; passwordlocation=spooldir
+                    ; Usually the voicemail password (vmsecret) is stored in
+                    ; this configuration file.  By setting this option you can
+                    ; specify where Asterisk should read/write the vmsecret.
+                    ; Supported options:
+                    ;   voicemail.conf:
+                    ;     This is the default option.  The secret is read from
+                    ;     and written to voicemail.conf (or users.conf).
+                    ;   spooldir:
+                    ;     The secret is stored in a separate file in the user's
+                    ;     voicemail spool directory in a file named secret.conf.
+                    ;     Please ensure that normal Linux users are not
+                    ;     permitted to access Asterisk's spool directory as the
+                    ;     secret is stored in plain text.  If a secret is not
+                    ;     found in this directory, the password in
+                    ;     voicemail.conf (or users.conf) will be used.
+                    ; Note that this option does not affect password storage for
+                    ; realtime users, which are still stored in the realtime
+                    ; backend.
+; messagewrap=no    ; Enable next/last message to wrap around to
+                    ; first (from last) and last (from first) message
+                    ; The default is "no".
+; minpassword=0 ; Enforce minimum password length
+
+; vm-password=custom_sound
+			;     Customize which sound file is used instead of the default
+			;     prompt that says: "password"
+; vm-newpassword=custom_sound
+			;     Customize which sound file is used instead of the default
+			;     prompt that says: "Please enter your new password followed by
+			;     the pound key."
+; vm-passchanged=custom_sound
+			;     Customize which sound file is used instead of the default
+			;     prompt that says: "Your password has been changed."
+; vm-reenterpassword=custom_sound
+			;     Customize which sound file is used instead of the default
+			;     prompt that says: "Please re-enter your password followed by
+			;     the pound key"
+; vm-mismatch=custom_sound
+			;     Customize which sound file is used instead of the default
+			;     prompt that says: "The passwords you entered and re-entered
+			;     did not match."
+; vm-invalid-password=custom_sound
+			;     Customize which sound file is used instead of the default
+			;     prompt that says: ...
+; vm-pls-try-again=custom_sound
+                        ; Customize which sound file is used instead of the
+                        ; default prompt that says "Please try again."
+; vm-prepend-timeout=custom_sound
+                        ; Customize which sound file is used when the user
+                        ; times out while recording a prepend message instead
+                        ; of the default prompt that says "then press pound"
+                        ; note that this will currently follow vm-pls-try-again.
+                        ; this behavior is subject to change in the near future.
+; listen-control-forward-key=#	; Customize the key that fast-forwards message playback
+; listen-control-reverse-key=*	; Customize the key that rewinds message playback
+; listen-control-pause-key=0	; Customize the key that pauses/unpauses message playback
+; listen-control-restart-key=2	; Customize the key that restarts message playback
+; listen-control-stop-key=13456789	; Customize the keys that interrupt message playback, probably all keys not set above
+
+; Maximum number of messages allowed in the 'Deleted' folder. If set to 0
+; or no then no deleted messages will be moved. If non-zero (max 9999) then up
+; to this number of messages will be automagically saved when they are
+; 'deleted' on a FIFO basis.
+; defaults to being off
+; backupdeleted=100
+
+
+[zonemessages]
+; 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
+;
+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'
+european=Europe/Copenhagen|'vm-received' a d b 'digits/at' HM
+
+
+
+[default]
+; Note: The rest of the system must reference mailboxes defined here as mailbox@default.
+
+1234 => 4242,Example Mailbox,root@localhost
+;4200 => 9855,Mark Spencer,markster@linux-support.net,mypager@digium.com,attach=no|serveremail=myaddy@digium.com|tz=central|maxmsg=10
+;4300 => 3456,Ben Rigas,ben@american-computer.net
+;4310 => -5432,Sales,sales@marko.net
+;4069 => 6522,Matt Brooks,matt@marko.net,,|tz=central|attach=yes|saycid=yes|dialout=fromvm|callback=fromvm|review=yes|operator=yes|envelope=yes|moveheard=yes|sayduration=yes|saydurationm=1
+;4073 => 1099,Bianca Paige,bianca@biancapaige.com,,delete=1|emailsubject=You have a new voicemail.|emailbody=Click on the attachment to listen.|rip=2010-06-04
+;4110 => 3443,Rob Flynn,rflynn@blueridge.net
+;4235 => 1234,Jim Holmes,jim@astricon.ips,,Tz=european
+
+
+;
+; Mailboxes may be organized into multiple contexts for
+; voicemail virtualhosting
+;
+
+[other]
+;The intro can be customized on a per-context basis
+;directoryintro=dir-company2
+1234 => 5678,Company2 User,root@localhost
+
+; example for our acme compartmentalized company
+;
+; Pete telecommutes from Chicago, so we'll customize timestamps for him.
+;
+;[acme]
+;111 => 7383,Pete,pete@acme-widgets.com,,tz=central
+;112 => 6262,Nancy,nancy@acme-widgets.com
+;
+
+; ---------------------------------------------------------------------------
+; IMAP user settings and overrides.  These are only applicable when Asterisk is
+; compiled with IMAP support.
+;
+; imapuser=username                ; The IMAP username of the mailbox to access
+; imappassword=password            ; The IMAP password of the user
+; imapvmshareid=xxxx               ; A shared mailbox ID to use for the IMAP mailbox
+                                   ; login, as opposed to the mailbox dialed
+; imapfolder                       ; Overrides the global imapfolder setting
+; imapserver                       ; Overrides the global imapserver setting
+; imapport                         ; Overrides the global imapport setting
+; imapflags                        ; Overrides the global imapflags setting
+
+;
+;[imapvm]
+;4324 => 7764,Ellis Redding,red@buxton.us,,imapuser=eredding|imappassword=g3tbusy|imapfolder=notinbox
+;4325 => 2392,Andrew Dufresne,andy@dufresne.info,,imapuser=adufresne|imappassword=rockh@mmer
diff --git a/configs/samples/vpb.conf.sample b/configs/samples/vpb.conf.sample
new file mode 100644
index 000000000..fecb3ec59
--- /dev/null
+++ b/configs/samples/vpb.conf.sample
@@ -0,0 +1,248 @@
+;
+; Voicetronix Voice Processing Board (VPB) telephony interface
+;
+; Configuration file
+;
+
+[general]
+;
+; Total number of Voicetronix cards in this machine
+;
+cards=0
+
+;
+; Which indication functions to use
+;    1 = use Asterisk functions
+;    0 = use VPB functions
+;
+indication=1
+
+;
+; Echo Canceller suppression threshold
+;    0    = no suppression threshold
+;    2048 = -18dB
+;    4096 = -24dB
+;
+;ecsuppthres=0
+
+;
+; Inter-digit delay timeout, used when collecting DTMF tones for dialling
+; from a station port.  Measured in milliseconds.
+;
+dtmfidd=3000
+
+;
+; How to play DTMF tones
+;    any value     = use Asterisk functions
+;    commented out = use VPB functions
+;
+;ast-dtmf=1
+
+;
+; How to detect DTMF tones
+;    any value     = use Asterisk functions
+;    commented out = use VPB functions
+;
+; NOTE: this setting is currently broken, and uncommenting it will
+; stop dialling from working.  Any volunteers to fix it?
+;ast-dtmf-det=1
+
+;
+; Use relaxed DTMF detection (ignored unless ast-dtmf-det is set)
+;
+relaxdtmf=1
+
+;
+; When we do a native bridge between two VPB channels:
+;    yes = only break the connection for '#' and '*'
+;    no  = break the connection for any DTMF
+;
+; NOTE: this is currently broken, and setting to no will segfault
+; Asterisk while dialling.  Any volunteers to fix it?
+;
+break-for-dtmf=yes
+
+;
+; The maximum period between received rings.  Measures in milliseconds.
+;
+timer_period_ring=4000
+
+
+[interfaces]
+;
+; Default language
+;
+language=en
+
+;
+; Default context
+;
+context=public
+
+;
+; Echo cancellation
+;     off  = no not use echo cancellation
+;     on   = use echo cancellation
+;
+echocancel=off
+
+;
+; Caller ID routines/signalling
+;   For FXO ports, select one of:
+;     on   = collect caller ID between 1st/2nd rings using VPB routines
+;     off  = do not use caller ID
+;     bell = bell202 as used in US, using Asterisk's caller ID routines
+;     v23  = v23 as used in the UK, using Asterisk's caller ID routines
+;   For FXS ports, set the channel's CID in '"name" <number>' format
+;
+; NOTE that other caller ID standards are supported in Asterisk, but are
+; not yet active in chan_vpb.  It should be reasonably trivial to add
+; support for the other standards (see the default chan_dahdi.conf for a
+; list of them) that Asterisk already handles.
+;
+callerid=bell
+
+;
+; Use a polarity reversal as the trigger for the start of caller ID,
+; rather than triggering after the first ring.
+;
+usepolaritycid=0
+
+;
+; Use loop drop to detect the end of a call.  On by default, but if you
+; experience unexpected hangups, try turning it off.
+;
+useloopdrop=1
+
+;
+; Use in-kernel bridging.  This will generally give lower delay audio if
+; bridging between two VPB channels.  It will not affect bridging
+; between VPB channels and other technologies.
+;
+usenativebridge=1
+
+;
+; Software transmit and receive gain.  Adjusting these will change the
+; volume of audio files that are played (tx) and recorded (rx).  It will
+; _not_ affect audio between channels in a native bridge.  It will,
+; however, affect the volume of audio between VPB channels and channels
+; using other technologies (such as VoIP channels).  Usually it's best to
+; leave these as they are.  If you're looking to get rid of echo, the
+; first thing to do is match your line impedance with the bal1/bal2/bal3
+; settings.
+;
+;txgain=0.0
+;rxgain=0.0
+
+;
+; Hardware transmit and receive gain.  Adjusting these will change the
+; volume of all audio on a channel.  The allowed range of settings is
+; -12.0 to 12.0 (measured in dB).
+;
+;txhwgain=0.0
+;rxhwgain=0.0
+
+;
+; Balance register settings, for matching the impedance of the card to
+; that of the connected equipment.  Only relevant for OpenLine and
+; OpenSwitch series cards.  Values should be in the range 0 - 255.
+;
+; We (Voicetronix) have determined the best codec balance values for
+; standard interfaces based on their US, Australian and European
+; specifications, shown below.
+;
+; US (600 ohm)
+;bal1=0xf8
+;bal2=0x1a
+;bal3=0x0c
+;
+; Australia (complex impedance)
+;bal1=0xf0
+;bal2=0x5d
+;bal3=0x79
+;
+; Europe (CTR-21)
+;bal1=0xf0
+;bal2=0x6e
+;bal3=0x75
+
+;
+; Logical groups can be assigned to allow outgoing rollover.  Groups range
+; from 0 to 63, and multiple groups can be specified.
+;
+group=1
+
+;
+; Ring groups (a.k.a. call groups) and pickup groups.  If a phone is
+; ringing and it is a member of a group which is one of your pickup
+; groups, then you can answer it by picking up and dialling *8#.  For
+; simple offices, just make these both the same.  Groups range from 0 to
+; 63.
+;
+callgroup=1
+pickupgroup=1
+
+;
+; If we haven't had a "grunt" (voice activity detection) for this many
+; seconds, then we hang up the line due to inactivity.  Default is one
+; hour.
+;
+grunttimeout=3600
+
+;
+; Type of line and line handling.  This setting will usually be overridden
+; on a per channel basis.  Valid settings are:
+;     fxo       = this is an FXO port
+;     immediate = this is an FXS port, with no dialtone or dialling
+;                   required (ie it is a "hotline")
+;     dialtone  = this is an FXS port, providing dialtone and dialling
+;
+mode=immediate
+
+;-------------------------------------------------------------------------
+; Channel definitions
+;
+; Each channel inherits the settings specified above, unless the are
+; overridden.  As a minimum, the board number and channel number must be
+; set, starting from 0 for the first board, and for the channels on each
+; board.  For example, board 0, channels 0 to 11, then board 1, channels
+; 0 to 11 for two OpenSwitch12 cards.
+;
+
+;
+; First board is an OpenSwitch12 card (jumpers at factory defaults)
+;
+;board=0
+;
+;mode=dialtone
+;context=from-handset
+;group=1
+;channel=0
+;channel=1
+;channel=2
+;channel=3
+;channel=4
+;channel=5
+;channel=6
+;channel=7
+;
+;mode=fxo
+;context=from-pstn
+;group=2
+;channel=8
+;channel=9
+;channel=10
+;channel=11
+
+;
+; Second board is an OpenLine4
+;
+;board=1
+;
+;mode=fxo
+;group=2
+;context=from-pstn
+;channel=0
+;channel=1
+;channel=2
+;channel=3
diff --git a/configs/samples/xmpp.conf.sample b/configs/samples/xmpp.conf.sample
new file mode 100644
index 000000000..dad0f79ef
--- /dev/null
+++ b/configs/samples/xmpp.conf.sample
@@ -0,0 +1,42 @@
+[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.
+;forceoldssl=no     ; Force the use of old-style SSL.
+;keepalive=
+