=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2006.07.04 23:20:58 =~=~=~=~=~=~=~=~=~=~=~=
cat /etc/asterisk/za[K[Ksccp.conf
; (SCCP*)
;
; An implementation of Skinny Client Control Protocol (SCCP)
;
; Sergio Chersovani (mlists@c-net.it)
; http://chan-sccp.belios.de
;
[general]
servername = Asterisk ; show this name on the device registration
keepalive = 60 ; phone keep alive message evey 60 secs. Used to check the voicemail
debug = 1 ; console debug level. 1 => 10
context = users
dateFormat = D.M.Y ; M-D-Y in any order. Use M/D/YA (for 12h format)
bindaddr = 192.168.0.106 ; replace with the ip address of the asterisk server (RTP important param)
port = 2000 ; listen on port 2000 (Skinny, default)
disallow=all ; First disallow all codecs
allow=g729
allow=alaw ; Allow codecs in order of preference
allow=ulaw ;
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
autoanswer_ring_time = 1 ; 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
musicclass=default ; Sets the default music on hold class
language=en ; Default language setting
;accountcode=skinny ; accountcode to ease billing
deny=0.0.0.0/0.0.0.0 ; Deny every address except for the only one allowed.
permit=192.168.0.0/255.255.255.0 ; 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 = 192.168.0.0/255.255.255.0 ; All RFC 1918 addresses are local networks
externip = 202.125.141.11 ; IP Address that we're going to notify in RTP media stream
;externhost = mydomain.dyndns.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
;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. 2 to 6
;cfwdall = off ; activate the callforward ALL stuff and softkeys
;cfwdbusy = off ; activate the callforward BUSY stuff and softkeys
[devices]
type = 7935 ; device type (see below)
autologin = line79052,,line79052 ; lines list. You can add an empty line for an empty button (7960, 7970, 7940, 7920)
description = Phone7905 ; internal description. Not important
;keepalive = 60 ; set 0 to disable the keepalive check.
;tzoffset = +2
transfer = on ; enable or disable the transfer capability. It does remove the transfer softkey
park = on ; take a look to the compile howto. Park stuff is not compiled by default
speeddial = ; you can add an empty speedial if you want an empty button (7960, 7970, 7920)
speeddial = 1000,name ; speeddial number and name
cfwdall = off ; activate the callforward stuff and softkeys
cfwdbusy = off
dtmfmode = inband ; inband or outofband. outofband is the native cisco dtmf tone play.
;==== 7935 =======
type = 7935
description = Bedroom
tzoffset = 0
autologin = 2123
device => SEP00E075240F9D
; Some phone model does not play dtmf tones while connected (bug?), so the default is inband
imageversion = P00405000700 ; useful to upgrade old firmwares (the ones that do not load *.xml from the tftp server)
;deny=0.0.0.0/0.0.0.0 ; Same as general
;permit=192.168.0.0/255.255.255.255 ; This device can register only using this ip address
dnd = on ; turn on the dnd softkey for this device. Valid values are "off", "on" (busy signal), "reject" (busy signal), "silent" (ringer = silent)
trustphoneip = yes ; 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 = ringout ; 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 for this device
;mwilamp = on ; Set the MWI lamp style when MWI active to on, off, wink, flash or blink
;mwioncall = off ; Set the MWI on call.
device => SEP00119341E684 ; device name SEP<MAC>
[lines]
id = 2123 ; future use
pin = 1234 ; future use
label = 7935 ; button line label (7960, 7970, 7940, 7920)
description = Line 79051 ; top diplay description
context = users
incominglimit = 2 ; more than 1 incoming call = call waiting.
transfer = on ; per line transfer capability. on, off, 1, 0
mailbox = 1001 ; voicemail.conf (syntax: vmbox[@context][:folder])
vmnum = 600 ; speeddial for voicemail administration, just a number to dial
cid_name = 7935 CID ; caller id name
cid_num = 79051
trnsfvm = 1000 ; extension to redirect the caller (e.g for voicemail)
secondary_dialtone_digits = 9 ; digits for the secondary dialtone (max 9 digits)
secondary_dialtone_tone = 0x22 ; outside dialtone
musicclass=default ; Sets the default music on hold class
language=en ; Default language setting
;accountcode=79501 ; accountcode to ease billing
rtptos = 184 ; sets the the rtp packets TOS for this line
echocancel = on ; sets the phone echocancel for this line
silencesuppression = off ; sets the silence suppression 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
line => 2123
; phone types
; 12 -- Cisco IP Phone 12SP+ (or other 12 variants)
; 30 -- Cisco IP Phone 30VIP (or other 30 variants)
; 7902 -- Cisco IP Phone 7902G
; 7905 -- Cisco IP Phone 7905G
; 7910 -- Cisco IP Phone 7910
; 7912 -- Cisco IP Phone 7912G
; 7935 -- Cisco IP Conference Station 7935
; 7936 -- Cisco IP Conference Station 7936
; 7920 -- Cisco IP Wireless Phone 7920
; 7940 -- Cisco IP Phone 7940
; 7960 -- Cisco IP Phone 7960
; 7970 -- Cisco IP Phone 7970
; 7914 -- Cisco IP Phone 7960 with a 7914 addon
; ata -- Cisco ATA-186 or Cisco ATA-188
; kirk -- Kirk telecom ip phones
]0;root@LHR-PBX:~[root@LHR-PBX root]#