SCCP Hiden Server config Phone ssh login dev_sshUserId cisco Help. Phone ssh password dev_sshPassword cisco Help. Phone protocol dev_deviceProtocol SCCP Help. SCCP Server config SCCP Server Name servername Vt Servername: This is the type of server - usually, it will be Asterisk. Bind Address : [ Port ] 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. : External Ip externip sccp-custom 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. / SCCP SERVER Keepalive 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. debug SCCP DEBUG none No Core SCCP Device All Debug: Enable debugging level in SCCP module. SCCP Networks Alow Networks / Mask permit 0.0.0.0/0.0.0.0 Add Alow network Alow network settings. Blank fields will be ignored used Network 0.0.0.0. Deny Networks / Mask deny 0.0.0.0/0.0.0.0 Add Deny network All RFC 1918 addresses are local networks. Should always be at least '0.0.0.0/0.0.0.0'. Local Networks / Mask localnet 0.0.0.0/0.0.0.0 Add local networks Local network settings. Blank fields will be ignored used Network 0.0.0.0. SCCP Language SCCP System Language 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. SCCP Network Device Language netlang English_United_States The Network locales allows the phone to play tones (ringing, busy etc.) native to the phone's country. SCCP Phone Device Language devlang English_United_States The user locale allows the phone to display text (menu items, soft keys etc.) native to the phone's language. SCCP Device config First Digit Timeout 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. Digit Timeout 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 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. Autoanswer Tone 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. Remote Hangup Tone 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 transfer_tone 0x32 sccp-custom Transfer Tone: The tone played when a call is transferred. Use '0' to disable the tone. Call Waiting 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. echocancel Echo Cancel no On Off Echo Cancel: Echo Cancellation (On or Off). silencesuppression Silence Suppression no On Off Silence Suppression: Slience Suppression on the phone. private Private Calling yes On Off 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. directed_pickup_modeanswer Directed Pickup no On Off Directed Pickup Mode (Answer): If a call is sent with the "directed pickup" flag, the phone will answer when set to "Yes". callanswerorder Call Answer Order oldestfirst Oldestfirst Latestfirst Call Answer Order: Which call should be answered first? The most common choice is "oldestfirst", but other orders are supported. mwilamp Message Lamp On On Off Wink Flash Blink Set the MWI lamp style when MWI active to on, off, wink, flash or blink mwioncall Message Lamp On Call no Yes No Set the MWI on call. SCCP NTP Service config ntp_config_enabled NTP Server Enabled no Yes No Enabling NTP settings in device configuration. NTP Server name (ip) ntp_server pool.ntp.org pool.ntp.org NTP Server name or IP ntp_server_mode Call Answer Order unicast Unicast Broadcast Multicast Manycast Configure NTP Server protocol time syncronization SCCP Time Service config Display Day Format dateformat D.M.Y 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 '.','-','/' Time Zone ntp_timezone UTC Date Format: Time zone SCCP Phone SRST HELP SRST 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 SRST Server Enabled no Yes No TempletDefault You enabled configure a Survivable Remote Site Telephony (SRST) reference as the last device in the CallManager(CM) list. srst_userModifiable SRST User Modifiable false Yes No The user can change the personal SRST configuration on the device srst_isSecure SRST Security false Yes No The user can change the personal SRST configuration on the device SRST Config Name srst_Name Enable You enabled configure a Survivable Remote Site Telephony (SRST) reference as the last device in the CallManager(CM) list. SRSt Server list srst_ip 3 / Add Server Help. SRST Sip Server list srst_sip 3 / Add Server Help. SCCP URL Service config Phone Service URL dev_servicesURL Phone authentication URL 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) Phone Idle information URL dev_idleURL URL of CiscoIPPhoneImage. Requires a non-zero setting in idleTimeout. Phone information URL dev_informationURL Phone messages URL dev_messagesURL Phone directory service URL 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) Phone proxyServer URL dev_proxyServerURL Phone Idle Timeout dev_idleTimeout 60 sccp-custom number SCCP QoS config SCCP ToS / CoS sccp_tos 0x68 sccp-custom sccp_cos 0x4 sccp-custom SCCP Type Of Service / Class Of Service: SCCP Type or Class of Service - this is modifiable, but don't. SCCP Audio ToS / CoS audio_tos 0xB8 sccp-custom audio_cos 0x6 sccp-custom Audio Type Of Service / Class Of Service: Audio Type or Class of Service - this is modifiable, but don't. SCCP Video ToS / CoS video_tos 0x88 sccp-custom video_cos 0x5 sccp-custom Video Type Of Service / Class Of Service: Video Type or Class of Service - this is modifiable, but don't. SCCP Line config Help_id2HELP SCCP HotLine 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 configurationThis will make it easier to register new devices and assign numbers At a doorWhere you want people to be able to only call one number For unprovisioned phonesTo 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. Alow =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 Hotline no Enabled Disabled Hotline Enabled: This allows unregistered extensions to connect to the system and dial the number listed below. Hotline extension hotline_extension *111 sccp-custom Hotline Extension: The number that gets called when a hotline is picked up. hint Hotline Label hotline_label Hot Line sccp-custom Hotline Label: The label on the device Hotline Context 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. System Context 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' Dynamically registration Context 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 Music On Hold Class musicclass none form-control Music Class: Available MOH Classes. These are the MOH classes listed in your current server. Default Dial Plan dial_template none sccp-custom Help. autoselectline_enabled Autoselect Line no Enabled Disabled Enables and disables auto line selection. meetme Meetme Conference no Enabled Disabled enable/disable conferencing via meetme (on/off), make sure you have one of the meetme apps mentioned below activated in module.conf Meetme Options 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 Auto Call Select no Enabled Disabled backgroundImageAccess Background Image no Enabled Disabled 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 take. SCCP Server Path config SCCP TFTP SERVER Path tftp_path /tftpboot sccp-custom Path to tftp home directory SCCP SERVER MySQL Device Table Name devicetable sccpdevice sccpdevice sccpdeviceconfig sccp-custom Device 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. SCCP SERVER MySQL Line Table Name linetable sccpline sccp-custom Line 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. createlangdir Create tftp empty language dir no Yes No Say 'yes' if you need to create cisco default language directory in tftp path. General Device configuration Device SEP ID.[XXXXXXXXXXXX]=MAC mac 000000000 sccp-custom The MAC address of the phone. You must specify 12 characters in the format XXXX.XXXX.XXXX or XX-XX-XX-XX-XX-XX or XXXXXXXXXXXX Type type 7911 model hw_select 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 addon none model hw_select sccp-custom 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. General Device configuration Device SEP ID mac 000000000 sccp-custom The MAC address of the phone Type type 7911 model hw_select 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. Type 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 addon none sccp-custom 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. Device configuration Device Label description 000000000 sccp-custom The information in the upper right corner of the device screen Soft Keyset softkeyset default System default softkey Time Zone offset tzoffset 00 Time Zone offset SCCP Network Device Language netlang English_United_States The Network locales allows the phone to play tones (ringing, busy etc.) native to the phone's country. SCCP Phone Device Language devlang Russian_Russia The user locale allows the phone to display text (menu items, soft keys etc.) native to the phone's language. Background Image backgroundImage sccp-custom For phones that can display background images - display this one. Default is [empty] Ringtone ringtone sccp-custom The 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/ directory with the rest of the config files. Buttons Configuration Device MAC SEP. mac 000000000 sccp-custom Enable/Disable Softkeys transfer Transfer no On Off Transfer allowed cfwdall Call Forward no On Off Activate the callforward softkeys. Default is On cfwdbusy Call Forward busy no On Off Activate the callforward busy softkeys. Default is On DND DND Button on Device On On Off Do Not Disturb. Default is Off directed_pickup Pickup/Gpickup yes On Off Enable Pickup function to direct pickup an extension. Default is On conf_allow Conference yes Yes No Allow the use of conference Pickup Context directed_pickup_context sccp-custom Context where direct pickup search for extensions. if not set current contect will be use. directed_pickup_modeanswer Mode Answer yes On Off On (Default)= the call has been answered when picked up. Off = call manager way, the phone who picked up the call rings the call Options: 'Immediate Answer' or 'Show CallerID' ?????? Device Properies useRedialMenu Redial using Phonebook no On 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. dtmfmode Dual-Tone Multi-Frequency (DTMF) outofband outofband inband Dual-Tone Multi-Frequency: outofband is the native cisco dtmf tone play mwilamp Message Waiting Indication (MWI) On On Off Wink Flash Blink Set the MWI lamp style when MWI active to on, off, wink, flash or blink Conference conf_play_general_announce Playback Conference Anncouncements yes Yes No Playback General Announcements (like: 'You are Entering/Leaving the conference') conf_play_part_announce Playback Participant Anncouncements yes Yes No Playback Personal/Participant Announcements, (like: 'You have been muted / You have been kicked') conf_mute_on_entry Start Muted no Yes No Mute new participants from the start conf_show_conflist Show conference list yes Yes No Automatically show conference list to the moderator Music On Hold Class 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. Network Alow Networks / Mask permit / Add Alow network Alow network settings. Blank fields will be ignored used Network 0.0.0.0. Deny Networks / Mask deny / Add Deny network All RFC 1918 addresses are local networks. Should always be at least '0.0.0.0/0.0.0.0'. nat NAT none Auto Off On Device NAT support (default Auto) directrtp Direct RTP none Auto Off On This option allow devices to do direct RTP sessions (default Off) earlyrtp Early RTP none Progress Offhook Immediate Dial Ringout None The audio strem will be open in the progress and connected state. Valid options: none, progress, offhook, dial, ringout. Default may be Progress. Template Name dialtemplate_name form-control Help. Device Template Function HELP SCCP Server Name 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 9Match 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 9Replace with digit %0The entire match %1 %2 %3 %4 %5Replace 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 Alow network settings. Blank fields will be ignored used Network 0.0.0.0.