;
; 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