XML_info NONE dev_sshUserId cisco Help. dev_sshPassword cisco Help. dev_deviceProtocol SCCP Help. sccp_xml_about XML Base ver: 11.2, Sccp ver: 433 Help. servername Asterisk XML Servername: This is the name of the server; normally it will be Asterisk. bindaddr 0.0.0.0 sccp-custom port 2000 sccp-custom Bind 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. : externip 0.0.0.0 sccp-custom externhost sccp-custom External IP Address of the firewall, required in case the PBX is running on a separate host behind it. IP Address that we're going to notify in RTP media stream as the pbx source address. Resolve Hostname (if dynamic) that we're going to resolve when setting up the RTP media stream (only active if externip=0.0.0.0 and host is natted.) 1 / externhost keepalive 60 sccp-custom number Time between Keep Alive checks. Valid range is 60-300 seconds. After much trial-and-error, the minimum (60) seems to work just fine. nat auto Global NAT support (default Auto) debug NONE Debug: Enable debugging level in SCCP module. displayconfig sccpgeneral Help! siptftp off Help! deny 0.0.0.0/0.0.0.0 + 5 Add Deny network All RFC 1918 addresses are local networks. Should always be at least '0.0.0.0/0.0.0.0'. permit Internal 0.0.0.0/0.0.0.0 + 5 Add Allow Range Allow 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. localnet Internal 0.0.0.0/0.0.0.0 + 5 Add Internal Range Local network settings. Blank fields will be ignored used Network 0.0.0.0. ccm_address + 5 Add Address This function is useful when the server has many interfaces, but devices must connect only to some interfaces. language English SCCP 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. devlang English_United_States The user locale allows the phone to display text (menu items, soft keys etc.) native to the phone's language. If No language packs found is shown, you need to add locales in the tftp server netlang United_States The Network locales allows the phone to play tones (ringing, busy etc.) native to the phone's country. If No language packs found is shown, you need to add locales in the tftp server firstdigittimeout 16 sccp-custom number First 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. digittimeout 8 sccp-custom number Digit Timeout: The amount of time to wait after the second (or subsequent) dialed digit. Override rules are the same as for firstdigittimeout. autoanswer_ring_time 0 sccp-custom number Autoanswer Ring Time: The amount of time the phones will ring when being called as Intercom or Paging mode. private on Private 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. callanswerorder oldestfirst Call Answer Order: Which call should be answered first? The most common choice is "oldestfirst", but other orders are supported. callhistory_answered_elsewhere Missed Calls mwilamp On Set the MWI lamp style when MWI active to on, off, wink, flash or blink mwioncall Set the MWI on call. earlyrtp The audio stream will be open in the progress and connected state. Valid options: Yes/ No simulate_enbloc Use simulated enbloc dialing to speedup connection when dialing while onhook (older phones) phonepersonalization This needs to be enabled to allow the server to push background or ringtones to the phone in the SEPXXXXXXXXXX.cnf.xml of each phone: autoanswer_tone 0x32 sccp-custom Autoanswer 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_tone 0x32 sccp-custom Remote Hangup Tone: The tone played by the phone when it received a remote hang-up signal. Use '0' to disable the tone. transfer_tone 0x32 sccp-custom Transfer Tone: The tone played when a call is transferred. Use '0' to disable the tone. callwaiting_tone 0x2D sccp-custom Call 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. cfwdnoanswer_timeout 30 sccp-custom number Timeout after which callforward noanswer (when active) will be triggered. default is 30 seconds cfwdall off Activate the callforward softkeys. Default is On cfwdbusy off Call Forward Activate the callforward busy softkeys. Default is On cfwdnoanswer off Activate the callforward NOANSWER stuff and softkeys. Default is On dndFeature on Do Not Disturb. Default is Off transfer Allow the use of the transfer feature ntp_config_enabled off Enable NTP settings in device configuration. ntp_server pool.ntp.org pool.ntp.org NTP Server name or IP address ntp_server_mode unicast Configure NTP Server protocol time synchronization dateformat Date 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_timezone System Time zone - this is set by FreePBX and can be changed in Settings : Advanced Settings logserver sccp-custom Log server Address: x.x.x.x:9001 daysdisplaynotactive sccp-custom Comma separated list of days that the phone's display is not active and will be automatically turned off after the time specified by "displayIdleTimeout" has elapsed (1 - Sunday, 7 - Saturday) displayontime sccp-custom Time in HH:MM format to automatically turn on the phone display. displayonduration sccp-custom Time duration in HH:MM format to automatically turn off the phone display after it was turned on. displayidletimeout sccp-custom Timeout in HH:MM format to automatically turn off phone display if outside the time specified by "daysDisplayNotActive", "displayOnTime" and "displayOnTimeout". settingsaccess off Whether a user can access the phone settings. videocapability on webaccess off Enable the phone's HTTP server. webadmin off Enable remote administration using the phone's HTTP server. 8821 model only. pcport off Disable the PC (computer) port spantopcport off Forward packets sent and received on the SW (network) port to the PC (computer) port. voicevlanaccess off Allow devices connected to the PC (computer) port to access the voice VLAN. enablecdpswport off Enable Cisco Discovery Protocol on the SW (network) port. enablecdppcport off Enable Cisco Discovery Protocol on the PC (computer) port. enablelldpswport off Enable Link Layer Discovery Protocol on the SW (network) port. enablelldppcport off Enable Link Layer Discovery Protocol on the PC (computer) port. Help_srst You 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_Option disable You enabled/configured a Survivable Remote Site Telephony (SRST) reference as the last device in the CallManager(CM) list. srst_userModifiable false The user can change the personal SRST configuration on the device srst_isSecure false The user can change the personal SRST configuration on the device srst_Name Enable You enabled configure a Survivable Remote Site Telephony (SRST) reference as the last device in the CallManager(CM) list. srst_ip 3 / Add Server Help. srst_sip 3 / Add Server Help. dev_servicesURL dev_authenticationURL The 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_informationURL dev_messagesURL dev_directoryURL This 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_idleURL URL of CiscoIPPhoneImage. Requires a non-zero setting in idleTimeout. dev_proxyServerURL dev_idleTimeout 60 sccp-custom number dev_secureservicesURL dev_secureauthenticationURL dev_secureinformationURL dev_securemessagesURL dev_securedirectoryURL dev_secureidleURL sccp_tos 0x68 sccp-custom sccp_cos 0x4 sccp-custom SCCP Signalling Type Of Service / Class Of Service - this is modifiable, but it is not recommended to change audio_tos 0xB8 sccp-custom audio_cos 0x6 sccp-custom Audio Type Of Service / Class Of Service - this is modifiable, but it is not recommended to change video_tos 0x88 sccp-custom video_cos 0x5 sccp-custom Video Type Of Service / Class Of Service - this is modifiable, but it is not recommended to change Help_id2 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 : 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. It is recommended to restrict to internal network use; set the deny/permit to 'internal' by default. This is insecure and should be switched off before going into production. A PLAR phone is still able to ring any other extension by entering the number on the phone before going off-hook. So either setup a secure context for these types of phones or switch off hotline. 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_enabled "no" Hotline Enabled: This allows unregistered extensions to connect to the system and dial the number listed below. hotline_extension *111 sccp-custom Hotline Extension: The number that gets called when a hotline is picked up. hint hotline_label Hot Line sccp-custom Hotline Label: The label on the device hotline_context default sccp-custom Hotline 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. context from-internal sccp-custom Context: 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' regcontext sccpregistration sccp-custom If 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 registers directed_pickup_context sccp-custom Context where direct pickup search for extensions. if not set current context will be use. pickup_modeanswer off Directed Pickup Mode (Answer): If a call is sent with the "directed pickup" flag, the phone will answer when set to "Yes". directed_pickup off Enable/disable Pickup button to do directed pickup from a specific extension. transfer_on_hangup off 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 transferred channel, press the 'endcall' softkey, or have the receiving party hangup first. dial_template NONE sccp-custom Help. autoselectline_enabled on Enables and Disables auto line selection. meetme off Enable/Disable conferencing via meetme (on/off), make sure you have one of the meetme apps mentioned below is activated in module.conf meetmeopts qxd sccp-custom 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 autocall_select on backgroundImageAccess true I'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 happen. callLogBlfEnabled 2 Show numbers you can redial, but also include their current device state, so you know when they are currently busy. Note that the other phonebook entries will now also monitor the remove device state and show the current device state echocancel yes Echo Cancel: Echo Cancellation (Yes or No). silencesuppression no Silence Suppression: Silence Suppression on the phone. musicclass NONE form-control Music Class: Available MOH Classes. These are the MOH classes listed in your current server. Help_tftp_rewrite This section is responsible for the modes of operation of external services: tftp http. * off - Flat model cisco. All data is in the directory TFTP Server Path * Provision - The directory structure is strictly defined in the manager. Requires tftpd mapping support. The mode is determined by Sccp_manager based on the presence of a mapping file in TFTP defaults and must be the same as the tftp server, or the phones will not find required files To change the mode, add or remove the -m option. provision_hide tftp_path sccp-custom Path to tftp home directory. This was determined by the Sccp_manager installer and should only be changed if the tftp root has been modified since installation provision_show tftp_rewrite_path sccp-custom Use path from provision index.cnf file. You must first make sure that you have properly configured the "Provision" tftp_rewrite off Provision Off Support the use of regular-expression-based filename remapping createlangdir no Say 'yes' if you need to create cisco default language directory in tftp path. getExternalData no If this is set to yes, Sccp_manager will try to get data files from dkgroot/Provision (languages, ring tones, firmware etc). This is enabled at your own risk - Sccp_Manager cannot vouch for the security or accuracy of these files, and this option is included to assist in getting up to speed as rapidly as possible. system_rouminguser no Experimental Function Roaming Users mac sccp-custom The MAC address of the phone. You must specify 12 characters in the format XXXX.XXXX.XXXX, XX:XX:XX:XX:XX:XX, XX-XX-XX-XX-XX-XX or XXXXXXXXXXXX type 7911 hw_select form-control The 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. addon NONE hw_select form-control Addon: 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. mac sccp-custom The MAC address of the phone type 79XX sccp-custom The 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. addon NONE hw_select form-control Addon: 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. description 000000000 sccp-custom The information in the upper right corner of the device screen. Only English letters and digits ! defaultLine lineSelect form-control Select the default line for this device softkeyset default System Default Softkey tzoffset 00 Time Zone offset devlang English_United_States The user locale allows the phone to display text (menu items, soft keys etc.) native to the phone's language. If No language packs found is shown, you need to add locales in the tftp server netlang English_United_States The Network locales allows the phone to play tones (ringing, busy etc.) native to the phone's country. If No language packs found is shown, you need to add locales in the tftp server phonepersonalization This needs to be enabled to allow the server to push background or ringtones to the phone in the SEPXXXXXXXXXX.cnf.xml of each phone: backgroundImage sccp-custom For phones that can display background images - display this one. Default is [empty] ringtone sccp-custom The ringtone that the phone will default to. Can be overridden in the phone. The files RINGLIST.XML provides 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. description 000000000 sccp-custom The information in the upper right corner of the device screen. Only English letters and digits ! defaultLine lineSelect form-control Select the default line for this device softkeyset default System Default Softkey tzoffset 00 Time Zone offset devlang English_United_States The user locale allows the phone to display text (menu items, soft keys etc.) native to the phone's language. If No language packs found is shown, you need to add locales in the tftp server netlang English_United_States The Network locales allows the phone to play tones (ringing, busy etc.) native to the phone's country. If No language packs found is shown, you need to add locales in the tftp server phonepersonalization This needs to be enabled to allow the server to push background or ringtones to the phone in the SEPXXXXXXXXXX.cnf.xml of each phone: backgroundImage sccp-custom For phones that can display background images - display this one. Default is [empty] ringtone sccp-custom The ringtone that the phone will default to. Can be overridden in the phone. The files RINGLIST.XML provides 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. mac 000000000 sccp-custom transfer off Transfer allowed park on Take a look to the compile how-to. Park stuff is not compiled by default. cfwdall off Activate the callforward softkeys. Default is On cfwdbusy off Activate the callforward busy softkeys. Default is On cfwdnoanswer on Activate the callforward NOANSWER stuff and softkeys. Default is On dndFeature off Do Not Disturb. Default is Off conf_allow on off Allow the use of conference useRedialMenu off You 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_dtmfmode auto Some phone models with bad firmware do send dtmf in a messed up order and need to be forced to skinny mode. mwioncall on Set the MWI on call. mwilamp on Set the MWI lamp style when MWI active to on, off, wink, flash or blink callhistory_answered_elsewhere Missed Calls Do not update history for this Phonebook if answered by another extension conf_hide conf_play_general_announce on Playback General Announcements (like: 'You are Entering/Leaving the conference') conf_play_part_announce on Playback Personal/Participant Announcements, (like: 'You have been muted / You have been kicked') conf_mute_on_entry off Mute new participants from the start conf_show_conflist on Automatically show conference list to the moderator conf_music_on_hold_class default form-control 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. custom_network_acc permit Help. custom_network_type ipv4 Help. custom_network_v sccp-custom custom_network_m sccp-custom Help. deny / sccp_hw_net_inherit + 5 Add Deny network All RFC 1918 addresses are local networks. Should always be at least '0.0.0.0/0.0.0.0'. permit NONE sccp_hw-ar_permit-grinternal sccp_hw_net_inherit + 5 Add Allow network Allow network settings. Blank fields will be ignored used Network 0.0.0.0. nat auto Device NAT support (default Auto) earlyrtp The audio stream will be open in the progress and connected state. Valid options: Yes/ No keepalive 60 sccp-custom number Time between Keep Alive checks. Valid range is 60-300 seconds. After much trial-and-error, the minimum (60) seems to work just fine. dialtemplate_name name_ruls form-control Help. Help_id1 Specifies 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/* title text1 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. id 000000000 sccp-custom Help! description Help! pin Help! password Help! roaminglogin off auto_logout off homedevice NONE sccp-custom Help! work_time off work_group Help! time_extend 30 Help! mac 000000000 sccp-custom The MAC address of the phone. You must specify 12 characters in the format XXXX.XXXX.XXXX, XX:XX:XX:XX:XX:XX, XX-XX-XX-XX-XX-XX or XXXXXXXXXXXX type 7911 form-control hw_select The 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. addon NONE hw_select form-control Addon: 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. _dialrules dialplan Dial templates control what number patterns are valid and how many seconds to wait for extra digits before dialing. setvar col-md-9 + 3 mailbox sccp-custom Name or id of linked maibox: Voicemail has to be enabled for this to be available incominglimit sccp-custom The SCCP channel number for this port. If you are planning to use this line as 'Shared' or use on several devices - leave this field blank or set limit to 10 calls. context sccp-custom Context: 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' namedcallgroup Sets the groups this line is a member of. Incoming calls to this line are given these groups as an identifier. This is a comma separated list of group names. (It is not recommended to use numeric names) namedpickupgroup This is the list of groups that this line can pickup from. This is a comma separated list of group names directed_pickup_context sccp-custom Context where direct pickup searches for extensions. if not set current context will be use. directed_pickup Enable/disable directed Pickup button (to do directed pickup from a specific extension). pickup_modeanswer off When using directed pickup, if this flag is set to Yes, the call will automatically be answered upon pickup secondary_dialtone_digits sccp-custom Digits to indicate an external line to user (secondary dialtone) Sample 9 or 8 (max 9 digits) secondary_dialtone_tone Outside dialtone frequency (default 0x22) transfer Allow the use of the transfer feature echocancel yes Echo Cancel: Echo Cancellation (Yes or No). dnd reject DND: Means how will dnd react when it is set on the device level dnd can have three states: off / busy(reject) / silent / Cycle Cycle - DND that cycles through all three states off -> reject -> silent -> off (this is the normal behaviour) Reject - Users can only switch off and on (in reject/busy mode) Silent - Users can only switch off and on (in silent mode) silencesuppression no Silence Suppression: Silence Suppression on the phone. musicclass NONE form-control Music Class: Available MOH Classes. These are the MOH classes listed in your current server. pin Help!