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