XML_infoNONEdev_sshUserIdciscoHelp.dev_sshPasswordciscoHelp.dev_deviceProtocolSCCPHelp.sccp_xml_aboutXML Base ver: 11.2, Sccp ver: 431Help.autoanswer_tone0x32sccp-customAutoanswer Tone: The tone the phone plays back when it picks up the phone in autoanswer mode. Default is '0x32'. Silence is '0x00'. There are lots of tones, all expressed as '0XNN' where 'NN' is a hexadecimal number.remotehangup_tone0x32sccp-customRemote Hangup Tone: The tone played by the phone when it received a remote hang-up signal. Use '0' to disable the tone.transfer_tone0x32sccp-customTransfer Tone: The tone played when a call is transferred. Use '0' to disable the tone.callwaiting_tone0x2Dsccp-customCall Waiting Tone: The tone played when a call is waiting. If you set this one to '0', you will not get a tone in your current call if a new call comes in, so you might want to disable call waiting for this line instead.sccp_tos0x68sccp-customsccp_cos0x4sccp-customSCCP Type Of Service / Class Of Service: SCCP Type or Class of Service - this is modifiable, but don't.audio_tos0xB8sccp-customaudio_cos0x6sccp-customAudio Type Of Service / Class Of Service: Audio Type or Class of Service - this is modifiable, but don't.video_tos0x88sccp-customvideo_cos0x5sccp-customVideo Type Of Service / Class Of Service: Video Type or Class of Service - this is modifiable, but don't.linetablesccplinesccp-customLine Table: This is the linetable for your realtime configuration. Don't change this unless you know what you are doing and have made all the appropriate changes in the rest of your Asterisk config files. If you do not want to use the realtime database anymore, you can set this to blank. NOT RECOMMENDED.devicetablesccpdevicesccp-customDevice Table: This is the devicetable for your realtime configuration. Don't change this unless you know what you are doing and have made all the appropriate changes in the rest of your Asterisk config files. There are two reasonable settings for this - the sccpdevice table or the sccpdeviceconfig view. If you do not want to use the realtime database anymore, you can set this to blank. NOT RECOMMENDED.servernameVtServername: This is the type of server - usually, it will be Asterisk.bindaddr0.0.0.0sccp-customport2000sccp-customBind Address: The address to bind to for SCCP. In general, it should be set to '0.0.0.0'. If you find problems with one-way audio, you can set this to a specific address in the server. Note that '127.0.0.1' is always wrong here. Also note that placing in your external IP address will cause the chan_sccp module to not load. A manual edit of the /etc/asterisk/sccp.conf file will be needed to fix and reload the module. : externipsccp-customExternal 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. / keepalive60sccp-customnumberTime between Keep Alive checks. Valid range is 60-300 seconds. After much trial-and-error, the minimum (60) seems to work just fine.natautoGlobal NAT support (default Auto)debugNONEDebug: Enable debugging level in SCCP module.displayconfigsccpgeneralHelp!permitInternal0.0.0.0/0.0.0.0+Add Allow RangeAllow network settings. Blank fields will be ignored used Network 0.0.0.0/0.0.0.0 to resolve any existing connections. You can use the 'internal' connections only from the networks connected to the server. localnetInternal0.0.0.0/0.0.0.0+Add Internal RangeLocal network settings. Blank fields will be ignored used Network 0.0.0.0.deny0.0.0.0/0.0.0.0+Add Deny networkAll RFC 1918 addresses are local networks. Should always be at least '0.0.0.0/0.0.0.0'.languageEnglishSCCP Language: This is the language for your hints and other features of the phone. If you don't have any languages installed or are using a single language, you can leave this blank.netlangEnglish_United_StatesThe Network locales allows the phone to play tones (ringing, busy etc.) native to the phone's country.devlangEnglish_United_StatesThe user locale allows the phone to display text (menu items, soft keys etc.) native to the phone's language.firstdigittimeout16sccp-customnumberFirst Digit Timeout: The amount of time after your first digit to start dialing automatically. This can be over-ridden with settings in your dialplan.xml or by using the 'immediate dial' button.digittimeout8sccp-customnumberDigit Timeout: The amount of time to wait after the second (or subsequent) dialed digit. Override rules are the same as for firstdigittimeout.autoanswer_ring_time0sccp-customnumberAutoanswer Ring Time: The amount of time the phones will ring when being called as Intercom or Paging mode.echocanceloffEcho Cancel: Echo Cancellation (On or Off).silencesuppressionoffSilence Suppression: Slience Suppression on the phone.privateonPrivate Calling Enabled: Place a call with privacy Options (no Caller ID) turned on. Needs to be supported in Asterisk to work through SIP and DAHDI trunks.directed_pickup_modeansweroffDirected Pickup Mode (Answer): If a call is sent with the "directed pickup" flag, the phone will answer when set to "Yes".callanswerorderoldestfirstCall Answer Order: Which call should be answered first? The most common choice is "oldestfirst", but other orders are supported.mwilampOnSet the MWI lamp style when MWI active to on, off, wink, flash or blinkmwioncalloffSet the MWI on call.directrtpoffThis option set global allow devices to do direct RTP sessions (default Off)earlyrtpnoneImmediateThe audio strem will be open in the progress and connected state. Valid options: NONE, progress, offhook, dial, ringout. Default may be Progress.simulate_enbloconUse simulated enbloc dialing to speedup connection when dialing while onhook (older phones)cfwdalloffActivate the callforward softkeys. Default is OncfwdbusyoffActivate the callforward busy softkeys. Default is OndndFeatureonDo Not Disturb. Default is Offntp_config_enabledoffEnabling NTP settings in device configuration.ntp_serverpool.ntp.orgpool.ntp.orgNTP Server name or IPntp_server_modeunicastConfigure NTP Server protocol time syncronizationdateformatDate Format: The date format for the on-screen display. Can be one of the following: (D-M-YA, M.D.Y, Y/M/D) where 'D' is Day, 'M' is Month, 'Y' is Year, 'A' is 24-hour, 'a' is 12-hour, and the separators can be '.','-','/'ntp_timezonesccp-customDate Format: Time zoneHelp_srstYou can also configure a Survivable Remote Site Telephony (SRST) reference as the last device in the CallManager(CM) list. This proves especially useful in a remote site configuration where the phones and Asterisk are connected over a WAN network. SRST provides users with fallback support for the IP phones that cannot access the primary, secondary, or tertiary Asterisk Node in the CallManager List because of an Asterisk Node failure or loss of connectivity across the WAN. For the remote sites attached to multiple-service routers across the WAN, SRST ensures that your remote users receive continuous (although minimal) service by providing call handling support directly from the SRST router.When IP phones lose contact with primary, secondary, and tertiary Asterisk Nodes (CM's), they must establish a connection to a local SRST router to sustain the call-processing capability necessary to place and receive calls. The IP phone retains the IP address of the local SRST router as a default router in the Network Configuration area of the Settings menu. The Settings menu supports a maximum of five default router entries; however, the cnf.xml accommodates a maximum of three entries. When a secondary Asterisk Node is not available on the network, the local SRST Router's IP address is retained as the standby connection for Asterisk Node during normal operation.srst_OptiondisableYou enabled/configured a Survivable Remote Site Telephony (SRST) reference as the last device in the CallManager(CM) list.srst_userModifiablefalseThe user can change the personal SRST configuration on the devicesrst_isSecurefalseThe user can change the personal SRST configuration on the devicesrst_NameEnableYou enabled configure a Survivable Remote Site Telephony (SRST) reference as the last device in the CallManager(CM) list.srst_ip3/Add ServerHelp.srst_sip3/Add ServerHelp.dev_servicesURLdev_authenticationURLThe above is simply a dummy authentication page. It literally contains one word: AUTHORIZED (it receives UserId, Password, and devicename in the url - if you truly wish to implement special auth)dev_idleURLURL of CiscoIPPhoneImage. Requires a non-zero setting in idleTimeout.dev_informationURLdev_messagesURLdev_directoryURLThis is the URL for a CiscoIPPhoneMenu which gets appended to the end of the Missed/Received/Placed calls. I don't use it (I find it makes more sense to put my phone book under services)dev_proxyServerURLdev_idleTimeout60sccp-customnumberHelp_id2Setting 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 :
First time configuration
This will make it easier to register new devices and assign numbers
At a door
Where you want people to be able to only call one number
For unprovisioned phones
To only be able to call the helpdesk to get their phone set up
Be careful with this function. Especially not to the internet. So i would restring the deny/permit to 'internal' by default in that case.
Allow =
Specify a list of your networks, for example: 192.168.1.0
deny =
0.0.0.0/0.0.0.0
permit =
internal
hotline_context =
sccp
hotline_label =
hotline
hotline_enabledoffHotline Enabled: This allows unregistered extensions to connect to the system and dial the number listed below.hotline_extension*111sccp-customHotline Extension: The number that gets called when a hotline is picked up. hinthotline_labelHot Linesccp-customHotline Label: The label on the devicehotline_contextdefaultsccp-customHotline Context: This is the context through which the phone will connect. It should probably match your other contest. The default is "from-internal" but "from-internal-xfer" would also make sense by limiting the options for the person using the phone.contextfrom-internalsccp-customContext: This is the context in which your phones will operate. It should match the context you are using for the rest of your phones (if you have any). The FreePBX default is 'from-internal'regcontextsccpregistrationsccp-customIf regcontext is specified in sccp.conf chan-sccp-b will dynamically create and destroy a NoOp priority 1 extension for a given peer/line which registers with the server. If the context is not specified in extension.conf, then it will be dynamically created when an chan-sccp-b agent registersmusicclassNONEform-controlMusic Class: Available MOH Classes. These are the MOH classes listed in your current server.dial_templateNONEsccp-customHelp.autoselectline_enabledoffEnables and Disables auto line selection.meetmeoffEnable/Disable conferencing via meetme (on/off), make sure you have one of the meetme apps mentioned below activated in module.confmeetmeoptsqxdsccp-customOther 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 documentationautocall_selectoffbackgroundImageAccessoffI'm guessing on this one, but on some devices, the background image on the display can be modified at the device. I think this is the thing that allows that to take.provision_hidetftp_path/tftpbootsccp-customPath to tftp home directoryprovision_showtftp_rewrite_path/tftpbootsccp-customUse path from provision index.cnf file. You must first make sure that you have properly configured the "Provision"tftp_rewriteoffproproSupport the use of regular-expression-based filename remappingcreatelangdirnoSay 'yes' if you need to create cisco default language directory in tftp path.mac000000000sccp-customThe MAC address of the phone. You must specify 12 characters in the format XXXX.XXXX.XXXX or XX-XX-XX-XX-XX-XX or XXXXXXXXXXXXtype7911hw_select sccp-customThe type of phone: 7911, 7940, 7960, etc. Important note: the 'G' models are handled as the base model (e.g., 7962G is handled as 7962). In the Display mode, this field is read-only because the MAC address and the model number are a pair.addonNONEhw_select sccp-customAddon: Addons are model specific and only work with certain base phones. This phone model is identified as being a phone that does not accept sidecars. Update devmodel if this is not correct.mac000000000sccp-customThe MAC address of the phonetype7911hw_select sccp-customThe type of phone: 7911, 7940, 7960, etc. Important note: the 'G' models are handled as the base model (e.g., 7962G is handled as 7962). In the Display mode, this field is read-only because the MAC address and the model number are a pair.type79XXsccp-customThe type of phone: 7911, 7940, 7960, etc. Important note: the 'G' models are handled as the base model (e.g., 7962G is handled as 7962). In the Display mode, this field is read-only because the MAC address and the model number are a pair.addonNONEsccp-customAddon: Addons are model specific and only work with certain base phones. This phone model is identified as being a phone that does not accept sidecars. Update devmodel if this is not correct.description000000000sccp-customThe information in the upper right corner of the device screensoftkeysetdefaultSystem Default Softkeytzoffset00Time Zone offsetnetlangEnglish_United_StatesThe Network locales allows the phone to play tones (ringing, busy etc.) native to the phone's country.devlangRussian_RussiaThe user locale allows the phone to display text (menu items, soft keys etc.) native to the phone's language.backgroundImagesccp-customFor phones that can display background images - display this one. Default is [empty]ringtonesccp-customThe ringtone that the phone will default to. Can be overridden in the phone. The files RINGLIST.XML provice the basic phone ring tones, while DISTINCTIVERINGLIST.XML defines the list of possible ring tones for your other line types. They, along with the actual 'raw' ringtones, are stored in the /tftpboot/ringtones directory with the rest of the config files.mac000000000sccp-customtransferoffTransfer allowedcfwdallNONEActivate the callforward softkeys. Default is OncfwdbusyNONEActivate the callforward busy softkeys. Default is OndndFeatureNONEDo Not Disturb. Default is Offdirected_pickuponoffEnable Pickup function to direct pickup an extension. Default is Onconf_allowonoffAllow the use of conferencepickup_hidedirected_pickup_contextsccp-customContext where direct pickup search for extensions. if not set current contact will be use.directed_pickup_modeansweronOn (Default)= the call has been answered when picked up. Off = call manager way, the phone who picked up the call rings the callOptions: 'Immediate Answer' or 'Show CallerID' ??????useRedialMenuoffYou can specifying 'useRedialMenu = yes' in the sccp.conf device section and the redial softkey will cause the "placed calls" list instead of immediately calling the last dialed number.force_dtmfmodeautoSome phone models with bad firmware do send dtmf in a messed up order and need to be forced to skinny mode.mwioncallNONESet the MWI on call.mwilampNONESet the MWI lamp style when MWI active to on, off, wink, flash or blinkconf_hideconf_play_general_announceonPlayback General Announcements (like: 'You are Entering/Leaving the conference')conf_play_part_announceonPlayback Personal/Participant Announcements, (like: 'You have been muted / You have been kicked')conf_mute_on_entryoffMute new participants from the startconf_show_conflistonAutomatically show conference list to the moderatorconf_music_on_hold_classdefaultform-controlPlay 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.custom_network_accpermitHelp.custom_network_typeipv4Help.custom_network_vsccp-customcustom_network_msccp-customHelp.deny/sccp_hw_net_inherit+Add Deny networkAll RFC 1918 addresses are local networks. Should always be at least '0.0.0.0/0.0.0.0'.permitNONEInherit
falsesccp_hw-ar_permit-grinternalsccp_hw_net_inherit+Add Allow networkAllow network settings. Blank fields will be ignored used Network 0.0.0.0.natNONEDevice NAT support (default Auto)directrtpNONEAutoThis option allow devices to do direct RTP sessions (default Off)earlyrtpNONEImmediateThe audio stream will be open in the progress and connected state. Valid options: none, progress, offhook, dial, ringout. Default may be Progress.dialtemplate_nameform-controlHelp.Help_id1Specifies a pattern to match dialed digits against. Note: TEMPLATE must be in uppercase.Rules:
match:
Pattern to match, consists of one or more elements
0 1 2 3 4 5 6 7 8 9
Match digit
.
Match one digit, # or *
*
Match zero or more digits, # or *
\*
Match a literal *
,
Play secondary dial-tone specified by tone
timeout:
Number of seconds to wait for more digits if this pattern matches
line:
Only apply template to the specified line (optional)
rewrite:
Rewrite the matched digits before dialing, consists of one or more elements (optional)
0 1 2 3 4 5 6 7 8 9
Replace with digit
%0
The entire match
%1 %2 %3 %4 %5
Replace with group of digits matched, grouping is done by consecutive literal digit or . elements
%%
A literal %
.
Each . is replaced by the digit that was matched by the corresponding . in the pattern
tone:
Secondary dial-tone to play when a , is matched, up to 3 can be specified (optional)
dialtemplate*/10/*titletext1
empty
Bellcore-Alerting
Bellcore-Inside
Bellcore-Outside
Bellcore-Busy
Bellcore-BusyVerify
Bellcore-Reorder
Bellcore-CallWaiting
Bellcore-Hold
Bellcore-Reminder
Bellcore-Confirmation
Bellcore-Stutter
Bellcore-Permanent
Bellcore-None
Cisco-Zip
Cisco-ZipZip
Cisco-BeepBonk
Bellcore-dr1
Bellcore-dr2
Bellcore-dr3
Bellcore-dr4
Bellcore-dr5
Bellcore-dr6
CallWaiting-2
CallWaiting-3
CallWaiting-4
Allow network settings. Blank fields will be ignored used Network 0.0.0.0.