; ; 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