;
; SCCPv3 -- An implementation of Skinny Client Control Protocol
;
; Federico Santulli <[email protected]>
;
; See http://www.chan-sccp.org for more information about
; the SCCPv3 project. Please do not directly contact
; any of the maintainers of this project for assistance;
; the project provides a web site, tracker, and mailing list
; for your use.
;
; This program is free software, distributed under the terms of
; the GNU General Public License Version 2. See the LICENSE file
; at the top of the source tree.
;
[general]
servername=SCCPv3 ; name showed on the device after registration
allowguest=no ; Allow or reject guest device registration
; this feature allow unknown devices to not be rejected on
; register but not linking any line so as CCM behaviour
keepalive=60 ; global phone keep-alive timeout
;verbose=3 ; Console Log Level
;debug=3 ; Console Debug Level
context=sccp ; Asterisk default extension's context
dateformat=D.M.Y ; M-D-Y in any order. Use M/D/YA (for 12h format)
bindport=5060 ; TCP Port to bind to (SCCP standard port is 2000)
bindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
; You can specify port here too, like 123.123.123.123:2000
; See qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters.
tos_sccp=cs3 ; Sets TOS for SCCP packets.
tos_audio=ef ; Sets TOS for RTP audio packets.
;tos_video=af41 ; Sets TOS for RTP video packets. (not yet implemented)
cos_sccp=3 ; Sets 802.1p priority for SCCP packets.
cos_audio=5 ; Sets 802.1p priority for RTP audio packets.
;cos_video=4 ; Sets 802.1p priority for RTP video packets. (not yet implemented)
;disallow=all ; First disallow all codecs
;allow=alaw ; Allow codecs in order of preference
;allow=ulaw ; see Asterisk doc/rtp-packetization for framing options
;allow=g729 ; Most phones supports only alaw/ulaw/g729
;
; 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
; 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-device or per-line basis.
;
;mohsuggest=default
;
firstdigittimeout = 16 ; dialing timeout for the 1st digit
digittimeout = 8 ; dialing timeout between digits
;digittimeoutchar = # ; you can force the channel to dial with this char in the dialing state
autoanswer_ring_time = 0 ; ringing time in seconds for the autoanswer, the default is 0
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
callwaiting_tone = 0x2d ; sets to 0 to disable the callwaiting tone
language=en ; Default language setting
;callevents=no ; generate manager events when phone
; performs events (e.g. hold)
;accountcode=sccp ; accountcode to ease billing
deny=0.0.0.0/0.0.0.0 ; Deny every address except for the only one allowed.
permit=192.168.1.0/255.255.255.0 ; Accept class C 192.168.1.0
; 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.
;localnet = 192.168.1.0/255.255.255.0 ; All RFC 1918 addresses are local networks
;externip = 1.2.3.4 ; IP Address that we're going to notify in RTP media stream
;externhost = domain.mydns.org ; Hostname (if dynamic) that we're going to notify in RTP media stream
;externrefresh = 60 ; expire time in seconds for the hostname (dns resolution)
dnd = on ; turn on the dnd softkey for all devices. Valid values are "off", "on" (busy signal), "reject" (busy signal), "silent" (ringer = silent)
rtptos = 184 ; sets the default rtp packets TOS
echocancel = on ; sets the phone echocancel for all devices
silencesuppression = off ; sets the silence suppression for all devices
;callgroup=1,3-4 ; We are in caller groups 1,3,4. Valid for all lines
;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5. Valid for all lines
;amaflags = ; Sets the default AMA flag code stored in the CDR record
trustphoneip = no ; The phone has a ip address. It could be private, so if the phone is behind NAT
; we don't have to trust the phone ip address, but the ip address of the connection
tos = 0x68 ; call control packets tos (0x68 Assured forwarding)
;earlyrtp = none ; valid options: none, offhook, dial, ringout. default is none.
; The audio strem will be open in the progress and connected state.
private = on ; permit the private function softkey
privacy = full ; full = disable hints notification on devices, on = hints showed depending on private key, off = hints always showed
;mwilamp = on ; Set the MWI lamp style when MWI active to on, off, wink, flash or blink
;mwioncall = off ; Set the MWI on call.
;blindtransferindication = ring ; moh or ring. the blind transfer should ring the caller or just play music on hold
;protocolversion = 3 ; skinny version protocol. Just for testing. 0 to 11
;cfwdall = off ; activate the callforward ALL stuff and softkeys
;cfwdbusy = off ; activate the callforward BUSY stuff and softkeys
;cfwdnoanswer = off ; activate the callforward NOANSWER stuff and softkeys
;
;************************
;SCCP Realtime Support
;************************
; you should specify both tables to get realtime support working
;
rt-devices = sccpdevices ; datebasetable for devices
rt-lines = sccplines ; datebasetable for lines
;
;
; this is for custom softkey template
;
[softkeys]
onhook = redial,newcall,cfwdall,dnd
connected = hold,endcall,park,select,cfwdall,cfwdbusy,idivert
onhold = resume,newcall,endcall,transfer,confrn,select,dirtrfr,idivert
ringin = answer,endcall,idivert
offhook = redial,endcall,private,cfwdall,cfwdbusy,pickup,gpickup,meetme,barge
conntrans = hold,endcall,transfer,confrn,park,select,dirtrfr,cfwdall,cfwdbusy
digitsfoll = backspace,endcall
connconf = hold,endcall,join
ringout = endcall,transfer,cfwdall,idivert
offhookfeat = redial,endcall
onhint = pickup,barge
; This is a 7940 device
[SEP001F6C7E8C0A]
type = 7940 ; device type (see below)
imageversion = SCCP65.8-4-1SR2 ; useful to upgrade old firmwares (the ones that do not load *.xml from the tftp server)
;keepalive = 60 ; set 0 to disable the keepalive check.
;tzoffset = +2 ;
;addon = 7915 ; first addon. Allowed value are 7914, 7915, 7916. Please check your device for addons compatibility.
;addon = 7915 ; second addon.
button = line,1001,default ; this is default line
button = speeddial,snom,Dial1000 ; this is second line
;button = empty ; this is an empty button
;button = speeddial,1000,Dial1000 ; this is a speeddial button
;button = speeddial,1001,Dial1001,default ; this is an hotline button
;button = empty ; this is an empty button
;button = serviceurl,http://www.google.it,GoGoogle ; this is a service url
;button = feature,Displayname,ID ; this is a feature button
transfer = on ; enable or disable the transfer capability. It does remove the transfer softkey
conference = on ; enable or disable the conference capability. It does remove the conference softkey
park = on ; take a look to the compile howto. Park stuff is not compiled by default
cfwdall = off ; activate the callforward stuff and softkeys
cfwdbusy = off
cfwdnoanswer = off
pickupexten = off ; enable Pickup function to direct pickup an extension
;pickupcontext = sccp ; context where direct pickup search for extensions. if not set it will be ignored.
pickupmodeanswer = on ; on = asterisk way, the call has been answered when picked up
; off = call manager way, the phone who picked up the call rings the call
private = on ; permit the private function softkey for this device
privacy = full ; full = disable hints notification on devices, on = hints showed depending on private key, off = hints always showed
dnd = on ; turn on the dnd softkey for this device. Valid values are "off", "on" (busy signal), "reject" (busy signal), "silent" (ringer = silent) or user to toggle on phone
;mwilamp = on ; Set the MWI lamp style when MWI active to on, off, wink, flash or blink
;mwioncall = off ; Set the MWI on call.
mohinterpret=default ; Sets the default music on hold class
mohsuggest=default ; Suggested music on hold when other channels put on hold
dtmfmode = inband ; inband or outofband. outofband is the native cisco dtmf tone play.
; Some phone model does not play dtmf tones while connected (bug?), so the default is inband
;disallow=all ; First disallow all codecs
;allow=alaw ; Allow codecs in order of preference
;allow=g729 ; see Asterisk doc/rtp-packetization for framing options
deny=0.0.0.0/0.0.0.0 ; Same as general
permit=192.168.1.5/255.255.255.255 ; This device can register only using this ip address
trustphoneip = no ; The phone has a ip address. It could be private, so if the phone is behind NAT
; we don't have to trust the phone ip address, but the ip address of the connection
;earlyrtp = none ; valid options: none, offhook, dial, ringout. default is none.
; The audio strem will be open in the progress and connected state.
;setvar=testvar=value ; This is a device test variable
;
; This is a sample line
;
[7940]
label = 7940 ; button line label (7960, 7970, 7940, 7920)
context=sccp
description = Office ; top diplay description
echocancel = on ; sets the phone echocancel for this line
silencesuppression = off ; sets the silence suppression for this line
incominglimit = 2 ; more than 1 incoming call = call waiting.
perdevicelimit = 2
secondary_dialtone_digits = 9 ; digits for the secondary dialtone (max 9 digits)
secondary_dialtone_tone = 0x22 ; outside dialtone
cid_name = Office ; caller id name
cid_num = 7940 ; caller id number
transfer = on ; per line transfer capability. on, off, 1, 0
conference = on ; per line conference capability.
trnsfvm = 1000 ; extension to redirect the caller (e.g for voicemail)
mailbox = 1001 ; voicemail.conf (syntax: vmbox[@context][:folder])
vmnum = 600 ; speeddial for voicemail administration, just a number to dial
meetmenum = 700 ; this extension will receive meetme requests, SCCP_MEETME_ROOM channel variable will
; contain the room number dialed into simpleswitch.
mohinterpret=default ; Sets the default music on hold class
mohsuggest=default ; Suggested music on hold when other channels put on hold
language=en ; Default language setting
accountcode=79651 ; accountcode to ease billing
rtptos = 184 ; sets the the rtp packets TOS for this line
;callgroup=1,3-4 ; We are in caller groups 1,3,4. Valid for this line
;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5. Valid for this line
;amaflags = ; Sets the default AMA flag code stored in the CDR record for this line
;setvar=testvar2=value ; this is a line test variable
