281 lines
38 KiB
Plaintext
281 lines
38 KiB
Plaintext
;!
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;! Automatically generated configuration file
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;! Filename: sccp.conf.annotated (/usr/local/asterisk-13-branch/etc/asterisk/sccp.conf.annotated)
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;! Generator: sccp config generate
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;! Creation Date: Sun Nov 1 01:27:41 2015
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;!
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;
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; general section
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;
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[general]
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;servername = Asterisk ; (REQUIRED) show this name on the device registration
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;keepalive = 60 ; (REQUIRED) Phone keep alive message every 60 secs. Used to check the voicemail and keep an open connection between server and phone (nat).
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; Don't set any lower than 60 seconds.
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;debug = core ; (MULTI-ENTRY) console debug level or categories
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; examples: debug = 11 | debug = mwi,event,core | debug = all | debug = none or 0
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; possible categories:
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; core, sccp, hint, rtp, device, line, action, channel, cli, config, feature, feature_button, softkey, indicate, pbx
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; socket, mwi, event, adv_feature, conference, buttontemplate, speeddial, codec, realtime, lock, newcode, high, all, none
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;context = default ; (REQUIRED) pbx dialplan context
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;dateformat = M/D/Y ; (SIZE: 7) M-D-Y in any order. Use M/D/YA (for 12h format)
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;bindaddr = 0.0.0.0 ; (REQUIRED) replace with the ip address of the asterisk server (RTP important param)
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;port = 2000 ; listen on port 2000 (Skinny, default)
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deny = 0.0.0.0/0.0.0.0
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permit = internal ; (REQUIRED) (MULTI-ENTRY) Deny every address except for the only one allowed. example: '0.0.0.0/0.0.0.0'
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; Accept class C 192.168.1.0 example '192.168.1.0/255.255.255.0'
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; You may have multiple rules for masking traffic.
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; Rules are processed from the first to the last.
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; This General rule is valid for all incoming connections. It's the 1st filter.
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; using 'internal' will allow the 10.0.0.0, 172.16.0.0 and 192.168.0.0 networks
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;localnet = internal ; (MULTI-ENTRY) All RFC 1918 addresses are local networks, example '192.168.1.0/255.255.255.0'
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;externip = 0.0.0.0 ; External IP Address of the firewall, required in case the PBX is running on a seperate host behind it. IP Address that we're going to notify in RTP media stream as the pbx source address.
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;firstdigittimeout = 16 ; Dialing timeout for the 1st digit
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;digittimeout = 8 ; More digits
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;digittimeoutchar = # ; You can force the channel to dial with this char in the dialing state
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;recorddigittimeoutchar = no ; You can force the channel to dial with this char in the dialing state
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;simulate_enbloc = yes ; Use simulated enbloc dialing to speedup connection when dialing while onhook (older phones)
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;ringtype = outside ; Ringtype for incoming calls (default='outside')
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;autoanswer_ring_time = 1 ; Ringing time in seconds for the autoanswer.
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;autoanswer_tone = 0x32 ; Autoanswer confirmation tone. For a complete list of tones: grep SKINNY_TONE sccp_protocol.h
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; not all the tones can be played in a connected state, so you have to try.
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;remotehangup_tone = 0x32 ; Passive hangup notification. 0 for none
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;transfer_tone = 0 ; Confirmation tone on transfer. Works only between SCCP devices
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;transfer_on_hangup = no ; Complete transfer on hangup, without pressing transfer a second time.
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; Will complete transfer, when the transferer puts the receiver on hook, after the destination has been reached.
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; To cancel the transfer, either press resume on the transfered channel, press the 'endcall' softkey, or have the receiving party hangup first.
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;dnd_tone = 0x0 ; Use 0x2D, 0x31, 0x32, 0x33 to activate dnd incoming call indication when dnd silent is active
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;callwaiting_tone = 0x2d ; Sets to 0 to disable the callwaiting tone
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;callwaiting_interval = 0 ; Callwaiting ring interval in seconds. Set to 0 to disable the callwaiting ringing interval.
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;musicclass = default ; Sets the default music on hold class
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;language = en ; Default language setting
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;callevents = yes ; Generate manager events when phone
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; Performs events (e.g. hold)
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;accountcode = skinny ; Accountcode to ease billing
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;sccp_tos = 0x68 ; Sets the default sccp signaling packets Type of Service (TOS) (defaults to 0x68 = 01101000 = 104 = DSCP:011010 = AF31)
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; Others possible values : [CS?, AF??, EF], [0x??], [lowdelay, throughput, reliability, mincost(solaris)], none
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;sccp_cos = 4 ; sets the default sccp signaling packets Class of Service (COS).
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;audio_tos = 0xB8 ; sets the default audio/rtp packets Type of Service (TOS) (defaults to 0xb8 = 10111000 = 184 = DSCP:101110 = EF)
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;audio_cos = 6 ; sets the default audio/rtp packets Class of Service (COS).
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;video_tos = 0x88 ; sets the default video/rtp packets Type of Service (TOS) (defaults to 0x88 = 10001000 = 136 = DSCP:100010 = AF41)
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;video_cos = 5 ; sets the default video/rtp packets Class of Service (COS).
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;echocancel = yes ; sets the phone echocancel for all devices
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;silencesuppression = no ; sets the silence suppression for all devices
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; we don't have to trust the phone ip address, but the ip address of the connection
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;earlyrtp = progress ; valid options: none, offhook, immediate, dial, ringout and progress.
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; The audio stream will be open in the progress and connected state by default. Immediate forces overlap dialing.
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; (POSSIBLE VALUES: ["Immediate","OffHook","Dialing","Ringout","Progress","None"])
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;dndFeature = on ; Turn on the dnd softkey for all devices. Valid values are 'off', 'on'.
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;private = yes ; permit the private function softkey
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;mwilamp = on ; Set the MWI lamp style when MWI active to on, off, wink, flash or blink
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; (POSSIBLE VALUES: ["Off","On","Wink","Flash","Blink"])
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;mwioncall = no ; Set the MWI on call.
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;blindtransferindication = ring ; moh or ring. the blind transfer should ring the caller or just play music on hold
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; (POSSIBLE VALUES: ["RING","MOH"])
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;cfwdall = yes ; activate the callforward ALL stuff and softkeys
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;cfwdbusy = yes ; activate the callforward BUSY stuff and softkeys
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;cfwdnoanswer = yes ; activate the callforward NOANSWER stuff and softkeys
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;nat = auto ; Global NAT support.
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; (POSSIBLE VALUES: ["Auto","Off","(Auto)Off","On","(Auto)On"])
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;directrtp = no ; This option allow devices to do direct RTP sessions.
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;allowoverlap = no ; Enable overlap dialing support. If enabled, starts dialing immediately and sends remaing digits as DTMF/inband.
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; Use with extreme caution as it is very dialplan and provider dependent.
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callgroup = "" ; We are in caller groups 1,3,4. Valid for all lines
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pickupgroup = "" ; We can do call pick-p for call group 1,3,4,5. Valid for all lines
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;directed_pickup_modeanswer = yes ; Automatically Answer when using Directed Pickup.
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;amaflags = default ; Sets the default AMA flag code stored in the CDR record
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;callanswerorder = oldestfirst ; oldestfirst or lastestfirst
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; (POSSIBLE VALUES: ["OldestFirst","LastFirst"])
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regcontext = "" ; SCCP Lines will we added to this context in asterisk for Dundi lookup purposes.
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; Do not set to an already created/used context. The context will be autocreated. You can share the sip/iax regcontext if you like.
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;devicetable = sccpdevice ; datebasetable for devices
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;linetable = sccpline ; datebasetable for lines
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;meetme = yes ; enable/disable conferencing via meetme (on/off), make sure you have one of the meetme apps mentioned below activated in module.conf
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; when switching meetme=on it will search for the first of these three possible meetme applications and set these defaults
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; - {'MeetMe', 'qd'},
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; - {'ConfBridge', 'Mac'},
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; - {'Konference', 'MTV'}
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;meetmeopts = qxd ; options to send the meetme application, defaults are dependent on meetme app see the list above
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; Other options (app_meetme: A,a,b,c,C,d,D,E,e,F,i,I,l,L,m,M,o,p,P,q,r,s,S,t,T,w,x,X,1) see meetme specific documentation
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;jbenable = no ; Enables the use of a jitterbuffer on the receiving side of a sccp channel.
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; An enabled jitterbuffer will be used only if the sending side can create and the receiving side can not accept jitter.
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; The sccp channel can accept jitter, thus a jitterbuffer on the receive sccp side will beused only if it is forced and enabled.
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;jbforce = no ; Forces the use of a jitterbuffer on the receive side of a sccp channel.
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;jblog = no ; Enables jitterbuffer frame logging.
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;jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
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;jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
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; resynchronized. Useful to improve the quality of the voice, with
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; big jumps in/broken timestamps, usually sent from exotic devices
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; and programs.
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;jbimpl = fixed ; (SIZE: 11) Jitterbuffer implementation, used on the receiving side of a
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; sccp channel. Two implementations are currently available
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; - 'fixed' (with size always equals to jbmaxsize)
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; - 'adaptive' (with variable size, actually the new jb of IAX2).
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;hotline_enabled = yes ; Setting the hotline Feature on a device, will make it connect to a predefined extension as soon as the Receiver
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; is picked up or the 'New Call' Button is pressed. No number has to be given. This works even on devices which
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; have no entry in the config file or realtime database.
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; The hotline function can be used in different circumstances, for example at a door, where you want people to be
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; able to only call one number, or for unprovisioned phones to only be able to call the helpdesk to get their phone
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; set up. If hotline_enabled = yes, any device which is not included in the configuration explicitly will be allowed
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; to registered as a guest device. All such devices will register on a single shared line called 'hotline'.
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;hotline_context = sccp ;
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;hotline_extension = 111 ;
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;hotline_label = hotline ;
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;fallback = no ; Immediately fallback to primairy/master server when it becomes available (master/slave asterisk cluster) (TokenRequest)
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; Possible values are: true/false/odd/even/script.
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; active/passive cluster: true on active/false on passive
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; active/active cluster: even on active1/off on active2
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; more complex cluster: use script. It will be called with three arguments, namely mac-address, ip-address, devicetype.
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; and it should return 'ACK' (without the quotes) to acknowledge the token, or a value for the number of seconds to backoff and try again.
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; Value can be changed online via CLI/AMI command 'sccp set fallback true/false/odd/even/script'
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;backoff_time = 60 ; Time to wait before re-asking to fallback to primairy server (Token Reject Backoff Time)
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;server_priority = 1 ; Server Priority for fallback: 1=Primairy, 2=Secundary, 3=Tertiary etc
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; For active-active (fallback=odd/even) use 1 for both
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;
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; device section
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;
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[default_device](!)
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device = "" ; (SIZE: 15) device type
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devicetype = "" ; (SIZE: 15) device type
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description = "" ; device description
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keepalive = "" ; set keepalive to 60
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;tzoffset = 0 ; time zone offset
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;disallow = all
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;allow = ulaw ; (MULTI-ENTRY) Same as entry in [general] section
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;allow = alaw
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;transfer = yes ; enable or disable the transfer capability. It does remove the transfer softkey
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;park = yes ; take a look to the compile how-to. Park stuff is not compiled by default.
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;cfwdall = no ; activate the call forward stuff and soft keys
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;cfwdbusy = no ; allow call forward when line is busy
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;cfwdnoanswer = no ; allow call forward when line if not being answered
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;dndFeature = yes ; allow usage do not disturb button
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dnd = "" ; allow setting dnd action for this device. Valid values are 'off', 'reject' (busy signal), 'silent' (ringer = silent) or 'user' (not used at the moment). . The value 'on' has been made obsolete in favor of 'reject'
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; (POSSIBLE VALUES: ["Off","Reject","Silent","UserDefined"])
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;force_dtmfmode = auto ; auto, skinny or rfc2833. Some phone models with bad firmware do send dtmf in a messed up order and need to be forced to skinny mode.
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; (POSSIBLE VALUES: ["AUTO","RFC2833","SKINNY"])
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deny = ""
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permit = "" ; (MULTI-ENTRY) Same as entry in [general] section
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; This device can register only using this ip address
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audio_tos = "" ; sets the audio/rtp packets Type of Service (TOS) (defaults to 0xb8 = 10111000 = 184 = DSCP:101110 = EF).
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; Others possible values : 0x??, lowdelay, throughput, reliability, mincost(solaris), none.
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audio_cos = "" ; sets the audio/rtp packets Class of Service (COS)
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video_tos = "" ; sets the video/rtp packets Type of Service (TOS) (defaults to 0x88 = 10001000 = 136 = DSCP:100010 = AF41).
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video_cos = "" ; sets the video/rtp packets Class of Service (COS).
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nat = "" ; Device NAT support. Currently nat is automatically detected in most cases.
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; (POSSIBLE VALUES: ["Auto","Off","(Auto)Off","On","(Auto)On"])
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directrtp = "" ; This option allow devices to do direct RTP sessions.
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earlyrtp = "" ; valid options: none, offhook, immediate, dial, ringout and progress.
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; The audio stream will be open in the progress and connected state by default. Immediate forces overlap dialing.
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; (POSSIBLE VALUES: ["Immediate","OffHook","Dialing","Ringout","Progress","None"])
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private = "" ; permit the private function softkey for this device
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privacy = "" ; permit the private function softkey for this device
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mwilamp = "" ; Set the MWI lamp style when MWI active to on, off, wink, flash or blink
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; (POSSIBLE VALUES: ["Off","On","Wink","Flash","Blink"])
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mwioncall = "" ; Set the MWI on call.
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meetme = "" ; enable/disable conferencing via app_meetme (on/off)
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meetmeopts = "" ; options to send the app_meetme application (default 'qd' = quiet,dynamic pin)
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; Other options (A,a,b,c,C,d,D,E,e,F,i,I,l,L,m,M,o,p,P,q,r,s,S,t,T,w,x,X,1) see app_meetme documentation
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;softkeyset = default ; use specified softkeyset with name 'default'
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;useRedialMenu = no ; show redial phone book list instead of dialing the last number (adv_feature). Requires a Phone Service block in SEP....cnf.xml to work correct on Java phones (See conf/tftp/SEP example files)
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;directed_pickup = yes ; enable/disable Pickup button to do directed pickup from a specific extension.
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directed_pickup_context = "" ; context where direct pickup search for extensions. if not set current contect will be use.
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;directed_pickup_modeanswer = yes ; on = asterisk way, the call has been answered when picked up.
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monitor = "" ;
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allowoverlap = "" ; Allow for Overlap dialing (Continue dialing after the first part of the number has already been send to the pstn)
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setvar = "" ; (MULTI-ENTRY) extra variables to be set on line initialization multiple entries possible (for example the sip number to use when dialing outside)
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; format setvar=param=value, for example setvar=sipno=12345678
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permithost = "" ; (MULTI-ENTRY) permit/deny but by resolved hostname
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addon = "" ; One of 7914, 7915, 7916
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button = "" ; (MULTI-ENTRY) Buttons come in the following flavours (empty, line, speeddial, service, feature).
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; Examples (read the documentation for more examples/combinations):
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; - button = line,1234
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; - button = line,1234,default
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; - button = empty
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; - button = line,98099@11:Phone1
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; - button = line,98099@12:Phone2#ButtonLabel!silent ; append cidnum:'12' and cidname:'Phone2' to line-ci with label 'ButtonLabel', don't ring when dialed directly
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; - button = line,98099@+12:Phone2@ButtonLabel!silent ; same as the previous line
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; - button = line,98099@=12:Phone2!silent ; overwrite line-cid instead of appending
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; - button = speeddial,Phone 2 Line 1, 98021, 98021@hints
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; - button = feature,cfwdall,1234
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; - button = feature,PDefault,ParkingLot,default ; feature, name, feature_type, parkinglotContext [,RetrieveSingle]
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; - button = feature,PDefault,ParkingLot,default,RetrieveSingle ; feature, name, feature_type, parkinglotContext [,RetrieveSingle]
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;allowRinginNotification = no ; allow ringin notification for hinted extensions. experimental configuration param that may be removed in further version
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;conf_allow = yes ; Allow the use of conference
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;conf_play_general_announce = yes ; Playback General Announcements (like: 'You are Entering/Leaving the conference')
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;conf_play_part_announce = yes ; Playback Personal/Participant Announcements, (like: 'You have been muted / You have been kicked')
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;conf_mute_on_entry = no ; Mute new participants from the start
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;conf_music_on_hold_class = default ; Play music on hold of this class when no moderator is listening on the conference. If set to an empty string, no music on hold will be played.
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;conf_show_conflist = yes ; Automatically show conference list to the moderator
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backgroundImage = "" ; Set the Background Image after device registered. Image must be set as URI to a http served file.
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ringtone = "" ; Set the Ring Tone after device registered. Ring Tone must be set as URI to a http served file.
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imageversion = "" ; (SIZE: 31) ImageVersion to be loaded on the device.
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;
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; line section
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;
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[default_line](!)
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id = "" ; (SIZE: 7) id
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pin = "" ; (SIZE: 7) pin
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description = "" ; description
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context = "" ; pbx dialing context
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defaultSubscriptionId_name = "" ; (SIZE: 79) Name used on a shared line when no name is specified on the line button for the device
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defaultSubscriptionId_number = "" ; (SIZE: 79) Number used on a shared line when no name is specified on the line button for the device
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mailbox = "" ; Mailbox to store messages in. Format 'mailbox@context' or 'mailbox' when you use 'default' context
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vmnum = "" ; Number to dial to get to the users Mailbox
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adhocNumber = "" ; Adhoc Number or Private-line automatic ring down (PLAR):
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; Adhoc/PLAR circuits have statically configured endpoints and do not require the user dialing to connect calls.
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; - The adhocNumber is dialed as soon as the Phone is taken off-hook or when the new-call button is pressed.
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; - The number will not be dialed when choosing a line; so when you choose a line you can enter a number manually.
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meetme = "" ; enable/disable conferencing via meetme, make sure you have one of the meetme apps mentioned below activated in module.conf.
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; When switching meetme=on it will search for the first of these three possible meetme applications and set these defaults.
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; Meetme=>'qd', ConfBridge=>'Mac', Konference=>'MTV'
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meetmenum = "" ; This extension will receive meetme requests, SCCP_MEETME_ROOM channel variable will
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; contain the room number dialed into simpleswitch (this parameter is going to be removed).
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meetmeopts = "" ; options to send the meetme application, defaults are dependent on meetme app see the list above.
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; Other options (app_meetme: A,a,b,c,C,d,D,E,e,F,i,I,l,L,m,M,o,p,P,q,r,s,S,t,T,w,x,X,1) see conferencing app for specific documentation
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transfer = "" ; per line transfer capability
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;incominglimit = 6 ; allow x number of incoming calls (call waiting)
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echocancel = "" ; sets the phone echocancel for this line
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silencesuppression = "" ; sets the silence suppression for this line
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language = "" ; sets the language setting per line
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musicclass = "" ; sets the music on hold class per line
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accountcode = "" ; accountcode for this line to make billing per call possible
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amaflags = "" ; sets the AMA flags stored in the CDR record for this line
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callgroup = "" ; sets the caller groups this line is a member of
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pickupgroup = "" ; sets the pickup groups this line is a member of (this phone can pickup calls from remote phones which are in this caller group
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namedcallgroup = "" ; sets the named caller groups this line is a member of (ast111)
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namedpickupgroup = "" ; sets the named pickup groups this line is a member of (this phone can pickup calls from remote phones which are in this caller group (ast111)
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parkinglot = "" ; parkinglot assigned to this line
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trnsfvm = "" ; extension to redirect the caller to for voice mail
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secondary_dialtone_digits = "" ; digits to indicate an external line to user (secondary dialtone) (max 9 digits)
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;secondary_dialtone_tone = 0x22 ; outside dialtone frequency
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setvar = "" ; (MULTI-ENTRY) extra variables to be set on line initialization multiple entries possible (for example the sip number to use when dialing outside)
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; format setvar=param=value, for example setvar=sipno=12345678
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dnd = "" ; allow setting dnd action for this line. Valid values are 'off', 'reject' (busy signal), 'silent' (ringer = silent) or 'user' (not used at the moment). . The value 'on' has been made obsolete in favor of 'reject'
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; (POSSIBLE VALUES: ["Off","Reject","Silent","UserDefined"])
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regexten = "" ; SCCP Lines will we added to the regcontext with this number for Dundi look up purpose
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; If regexten is not filled in the line name (categoryname between []) will be used
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;
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; softkey section
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;
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;[mysoftkeyset]
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;type = softkeyset ; (SIZE: -1) This should be set to softkeyset
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;onhook = redial,newcall,cfwdall,dnd,pickup,gpickup,private ; (SIZE: 15) displayed when we are on hook
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;connected = hold,endcall,park,vidmode,select,cfwdall,cfwdbusy,idivert ; (SIZE: 15) displayed when we have a connected call
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;onhold = resume,newcall,endcall,transfer,conflist,select,dirtrfr,idivert,meetme ; (SIZE: 15) displayed when we have a call on hold
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;ringin = answer,endcall,transvm,idivert ; (SIZE: 15) displayed when we have an incoming call
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;offhook = redial,endcall,private,cfwdall,cfwdbusy,pickup,gpickup,meetme,barge ; (SIZE: 15) displayed when the phone is taken off hook
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;conntrans = hold,endcall,transfer,conf,park,select,dirtrfr,vidmode,meetme,cfwdall,cfwdbusy ; (SIZE: 15) displayed when we are connected and could transfer a call
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;digitsfoll = back,endcall,dial ; (SIZE: 15) displayed when one or more digits have been entered, more are expected
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;connconf = conflist,newcall,endcall,hold,vidmode ; (SIZE: 15) displayed when we are in a conference
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;ringout = empty,endcall,transfer,cfwdall,idivert ; (SIZE: 15) displayed when We are calling someone
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;offhookfeat = redial,endcall ; (SIZE: 15) displayed wenn we went offhook using a feature
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;onhint = redial,newcall,pickup,gpickup,barge ; (SIZE: 15) displayed when a hint is activated
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;onstealable = redial,newcall,cfwdall,pickup,gpickup,dnd,intrcpt ; (SIZE: 15) displayed when there is a call we could steal on one of the neighboring phones
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;holdconf = resume,newcall,endcall,join ; (SIZE: 15) displayed when we are a conference moderator, have the conference on hold and have another active call
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uriaction = "" ; (MULTI-ENTRY) (SIZE: 7) softkey uri action to replace default handling. Format: uriaction = softkeyname, uri[,uri...]
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; . URI can be an embedded cisco action (like Key:Service, Play:1041.raw) or a URLIf uri is a url the following parameters will be added to it: devicename, linename, channelname, callid, linkedid, uniqueid, appid, transactionid
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