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64
Technical.note/ATA.txt
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https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cata/186_188/2_15_ms/english/administration/guide/sccp/sccp/sccpaape.html
Äëÿ ðàáîòû Cisco ATA-186, 188
îáíîâëåíèå ïðîøèâêè
--- linux
/sata186us.linux -any -d3 ATA030204SCCP090202A.zup
Äëÿ ðàáîòû Cisco ATA-186, 188 ìîæåò ïîòðåáîâàòüñÿ ôàéë atadefault.cfg
---------- Config
cfgfmt.linux atadefault.txt atadefault.cfg
----------------------begin atadefault.txt ---------------------
#txt
UIPassword:0
UseTftp:1
TftpURL:0
cfgInterval:3600
EncryptKey:0
ToConfig:0
upgradecode:0,0x301,0x0400,0x0200,0.0.0.0,69,0,none
upgradelang:0,0x301,0x0400,0x0200,0.0.0.0,69,0,none
Dhcp:1
StaticIp:0
StaticRoute:0
StaticNetMask:0
CA0orCM0:172.30.122.41:2000
CA1orCM1:0
CA0UID:0
CA1UID:0
EPID0orSID0:.
EPID1orSID1:.
PrfCodec:1
LBRCodec:3
AudioMode:0x00350035
NumTxFrames:2
CallerIdMethod:0x00019e60
ConnectMode:0x90000400
DNS1IP:0.0.0.0
DNS2IP:0.0.0.0
UDPTOS:0xA0
RingCadence:2,4,25
DialTone:2,31538,30831,1380,1740,1,0,0,0
BusyTone:2,30467,28959,1191,1513,0,4000,4000,0
ReorderTone:2,30467,28959,1191,1513,0,2000,2000,0
RingBackTone:2,30831,30467,1943,2111,0,16000,32000,0
CallWaitTone:1,30831,0,5493,0,0,2400,2400,4800
ConfirmTone:1,30467,0,5970,0,0,480,480,1920
MediaPort:16384
UseMGCP:0
MGCPPort:2427
RetxIntvl:500
RetxLim:7
MGCPVer:MGCP1.0
NPrintf:0
TraceFlags:0x00000000
SigTimer:0x00000064
CodecName:PCMU,PCMA,G723,G729
OpFlags:0x2
VLANSetting:0x0000002b
-----------------------end atadefault.txt ------------------

49
Technical.note/Conferencing.txt
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Conference - NOT CONFERENCE BRIDGE !!!!! ( Sccp Conference)
Conference Introduction
The integrated conference solution build in chan-sccp-b is based on asterisk's ConfBridge functionality. In stead of having to memorize the confbridge voice menu and having to press DTMF keys to control your conference we have opted to include a visual Cisco-XML menu, which give you (the Moderator) the ability to Kick, Mute and Promote another user to become an additional Moderator.
Note: You need to './configure --enable-conference ...' when you built the chan_sccp.so module. Note: A conference always requires at least one moderator.
Conference Settings
The standard conference settings are setup per device and contain:
param default description
conf_allow yes Allow the use of conference
conf_play_general_announce yes Playback General Announcements (like: 'You are Entering/Leaving the conference')
conf_play_part_announce yes Playback Personal/Participant Announcements, (like: 'You have been muted / You have been kicked')
conf_mute_on_entry no Mute new participants from the start, when they enter the conference (Preventing them to talk amongst one another). The Moderator will have to UnMute a participant manually to allow them to speak. Useful in a classroom setting.
conf_music_on_hold_class 'default' 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.
conf_show_conflist yes Automatically show conference list to the moderator
Creating a New Conference (Conf Softkey)
Using the Conference Button makes it possible to set up a Simple Conference between 3 or more participant. (The actual minimum to start a conference is 2, but that doesn't make a lot of sense now does it.)
You already have 2 or more lines connected (One is active and the other(s) is/are on Hold), which you would like to put in a Conference; Simply Press the Conf Softkey Button.
If you do not already have these lines connected, that call some people first and then start the conference. It does not make sense to be conferencing on your own.
Conference List (ConfList Softkey)
When conf_show_conflist=yes or you press the ConfList Softkey, you will be presented with a Cisco-XML Menu, showing you all currently connected Participant. Something like this:
7970_Conference.png
You can use the Softkeys underneath the menu, for example:
Softkey Description
EndConf Hangup all participants and end the current conference
Kick Through a specific participant out of the conference (Call is hungup)
Mute Do not allow a specific participant to speak (The hear a voiceprompt stating that they have been muted (if conf_play_part_announce = yes), and the mute status is displayed on their display (if they have an sccp device))
Unmute Allow a specific participant to speak (The hear a voiceprompt stating that they have been unmuted (if conf_play_part_announce = yes), and the mute status is displayed on their display (if they have an sccp device)).
Promote Make a specific participant a moderator as well (giving them control over the conference as well). You can leave the conference by hanging up, without the conference being terminated.
Exit Leave the ConfList Menu, but remain connected to the conference. This makes it possible to put the conference onhold and invite someone new for examples. You need to press the ConfList Softkey to get back into the ConfList Menu.
Adding another Participant after the conference has already started (Join Softkey)
If you do need to add a person after having started the conference, then you need to exit the conflist menu, put the conference on hold and dial the new "future" participant, once that person has picked up, you press the join button on that new call and this new participant will be added to the conference and you will automatically resume the conference you where in before.
Once the conference is started you will be presented with the conflist menu which will allow you to control the conference directly from your phone (kick / mute participant and even promote one of the participant to become a secondary moderator, so that they can take over control of the conference and you are free to leave).
Q & A:
The Conference Softkey just created a two person conference
Question: Creating a conference call on my 7961 does not seem to work. Once I hit the Conference softbutton, it will create a conference but put me and the other person directly into the conference without giving me any way to call a third party.
Solution: Just put the first person on hold, dial the second person (and a third, fourth etc) and then press the conference button. All of the calls connected to your phone will automatically be put into the conference.

24
Technical.note/RedialMenu.txt
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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.
CallListStateUpdate (java phones)
If you add/enable the 'callLogBlfEnabled' xml entry in SEPXXX.cnf.xml under commonProfile, like so:
<commonProfile>
<callLogBlfEnabled>3</callLogBlfEnabled>
</commonProfile>
and you have added hints for your local extension in your dialplan, like:
exten => _XX.,hint,SCCP/${EXTEN}
Then the placed calls list will include the status of the remote extension, like this:
PlacedCalls
Which does 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.
Note: the hints for the extension need to be in the same context as the device/global context, for callLogBlfEnabled to work
# This does not apply to phones 7940. Be careful with these keys the phone may not boot !!!

554
Technical.note/SEP0000000000.cnf.xml_annotated
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<?xml version="1.0" encoding="UTF-8"?>
<device>
<fullConfig>true</fullConfig>
<deviceProtocol>SCCP</deviceProtocol>
<sshUserId>cisco</sshUserId>
<sshPassword>cisco</sshPassword>
<!--
After using the above login, you will be prompted again for a username/password.
Use debug/debug for this second login, and type 'help' for commands.
(also see note about sshAccess below)
-->
<devicePool>
<revertPriority>0</revertPriority>
<name>Default</name>
<dateTimeSetting>
<name>Netherlands</name>
<dateTemplate>D/M/YA</dateTemplate>
<timeZone>W. Europe Standard/Daylight Time</timeZone>
<ntps>
<ntp>
<name>pool.ntp.org</name>
<ntpMode>Unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>
<callManagerGroup>
<name>Default</name>
<tftpDefault>true</tftpDefault>
<members>
<member priority="0">
<callManager>
<name>Asterisk</name>
<description>Primary Asterisk Server</description>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<!--
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
<mgcpPorts>
<listen>2427</listen>
<keepAlive>2428</keepAlive>
</mgcpPorts>
-->
</ports>
<processNodeName>x.x.x.x</processNodeName>
</callManager>
</member>
<member priority="1">
<callManager>
<name>Asterisk 1</name>
<description>Secundary Asterisk Server</description>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<!--
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
<mgcpPorts>
<listen>2427</listen>
<keepAlive>2428</keepAlive>
</mgcpPorts>
-->
</ports>
<processNodeName>x.x.x.x</processNodeName>
<!--
'processNodeName' is the ip address of your asterisk server.
-->
</callManager>
</member>
</members>
</callManagerGroup>
<srstInfo>
<name>Enable</name>
<srstOption>Enable</srstOption>
<userModifiable>true</userModifiable>
<ipAddr1>x.x.x.x</ipAddr1>
<port1>2000</port1>
<ipAddr2></ipAddr2>
<port2>2000</port2>
<ipAddr3></ipAddr3>
<port3>2000</port3>
<sipIpAddr1>192.168.5.101</sipIpAddr1>
<sipPort1>5060</sipPort1>
<sipIpAddr2></sipIpAddr2>
<sipPort2>5060</sipPort2>
<sipIpAddr3></sipIpAddr3>
<sipPort3>5060</sipPort3>
<isSecure>false</isSecure>
</srstInfo>
<mlppDomainId>-1</mlppDomainId>
<mlppIndicationStatus>Default</mlppIndicationStatus>
<preemption>Default</preemption>
<connectionMonitorDuration>120</connectionMonitorDuration>
</devicePool>
<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>2</callLogBlfEnabled>
</commonProfile>
<vendorConfig>
<ehookEnable>1</ehookEnable>
<!--
Headset Hookswitch Control
This allows the headset to take the phone off-hook (with appropriate
cables/connections etc)
-->
<disableSpeaker>false</disableSpeaker>
<!--
Disable only the speakerphone functionality.
Disabling speakerphone functionality will not affect the handset.
true = Disabled.
false = Enabled (default).
-->
<disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
<!--
Enables and disables the speakerphone and headset.
true = Disabled.
false = Enabled (default).
-->
<pcPort>0</pcPort>
<!--
Enables and disables the Ethernet switch port on the phone so the IP phone can
have access to an Ethernet connection for a PC connection through the phone.
0 = Enabled (default).
1 = Disabled.
-->
<settingsAccess>1</settingsAccess>
<!--
Enables and disables the Settings button on an IP phone.
Indicates whether the Settings button on the phone is functional. When Settings Access is enabled, you can change the phone network configuration, ring type, and volume on the phone. When Settings Access is disabled, the Settings button is completely disabled; no options appear when you press the button. Also, you cannot adjust the ringer volume or save any volume settings. By default, Settings Access is enabled.
0 = Disabled.
1 = Enabled (default). The phone user can modify features by using the Settings menu.
2 = Restricted. The phone user is allowed to access User Preferences and volume settings only.
-->
<garp>0</garp>
<!--
Enables and disables IP phone response to gratuitous Address Resolution
Protocol (ARP) messages from the IP phone's Ethernet interface.
Indicates whether the phone will learn MAC addresses from Gratuitous ARP responses. Disabling the phones ability to accept Gratuitous ARP will prevent applications which use this mechanism for monitoring and recording of voice streams from working. If monitoring capability is not desired, change this setting to Disabled.
0 = Disabled.
1 = Enabled (default).
-->
<voiceVlanAccess>0</voiceVlanAccess>
<!--
Enables and disables spanning, which is the IP phone's access to the voice
VLAN of the PC to which the IP phone's Ethernet port is connected.
0 = Enabled (default).
1 = Disabled.
-->
<videoCapability>1</videoCapability>
<!--
a 7975G doesn't have any "real" video capability, it requires software
called video advantage to stream webcam/video from a pc connected to
the pc-port on the back of the phone. Enabled here just as a
curiosity for now.
Make sure this is only set when video is available. Seems to cause sporadic issues if not.
-->
<autoSelectLineEnable>0</autoSelectLineEnable>
<!--
Enables and disables auto line selection.
0 = Disabled.
1 = Enabled (default).
-->
<webAccess>0</webAccess>
<!--
Enables and disables web access that allows phone users to configure settings and features on User Option web pages.
This parameter indicates whether the phone will accept connections from a web browser or other HTTP client.
Disabling the web server functionality of the phone will block access to the phones internal web pages.
These pages provide statistics and configuration information.
Features, such as QRT ( Quality Report Tool ), will not function properly without access to the phones web pages.
This setting will also affect any serviceability application such as CiscoWorks 2000 that relies on web access.
The following options are available.
Disabled: Phone does not accept any HTTP connection. HTTP server is disabled on the phone.
Read Only: Phone displays web pages but does not allow any configuration.
Full: Phone displays web pages and allows configuration.
0 = Enabled (default).
1 = Disabled.
2 = Read Only.
-->
<daysDisplayNotActive>1,7</daysDisplayNotActive> <!-- sunday = 1, sat = 7 -->
<displayOnTime>08:00</displayOnTime>
<displayOnDuration>12:00</displayOnDuration>
<displayIdleTimeout>00:10</displayIdleTimeout>
<displayOnWhenIncomingCall>1</displayOnWhenIncomingCall>
<forwardingDelay>1</forwardingDelay>
<loggingDisplay>1</loggingDisplay>
<!-- <headsetWidebandUIControl>1</headsetWidebandUIControl> -->
<!--
Enables or disables wideband headset option on supported IP phones.
If the headsetWidebandUIControl parameter is set to Enable (0), the option set
in the phone UI, by the phone user, has priority over the value set for this
parameter.
0 = Enabled (default). Enables wideband headset on phone.
1 = Disabled. Disables wideband headset on phone.
-->
<!-- <handsetWidebandUIControl>1</handsetWidebandUIControl> -->
<!--
Enables or disables control of handset options by phone user.
0 = Enabled (default). Allows phone user to select either narrowband or wideband handset in the phone UI.
1 = Disabled.
-->
<!-- <headsetWidebandEnable>1</headsetWidebandEnable> -->
<!--
Enables or disables control of headset option by phone user.
0 = Enabled (default). Allows phone user to select either narrowband or wideband headset
1 = Disabled.
-->
<!-- <handsetWidebandEnable>1</handsetWidebandEnable> -->
<!--
Enables or disables wideband handset option on supported IP phones.
If the handsetWidebandUIControl parameter is set to Enable (1), the option
set in the phone UI, by the phone user, has priority over the value set for
this parameter.
0 = Phone default (default), equal to disabled or enabled and set by manufacturer.
1 = Enabled. Enables wideband handset on phone.
2 = Disabled. Disables wideband headset on phone.
-->
<spanToPCPort>1</spanToPCPort>
<!--
Enables and disables the path between the Ethernet switch port of an IP phone
and a connection to a PC.
0 = Enabled (default).
1 = Disabled.
-->
<g722CodecSupport>2</g722CodecSupport>
<!--
Enables and disables the registration of the G.722 codec on the IP phone.
0 = Phone default (default), equal to disabled or enabled and set by manufacturer.
1 = Disabled. Disables G.722-64K2 codec on phone.
2 = Enabled. Enables G.722-64K codec on phone.
See also advertiseG722Codec
-->
<peerFirmwareSharing>1</peerFirmwareSharing>
<enableCdpSwPort>1</enableCdpSwPort>
<enableCdpPcPort>1</enableCdpPcPort>
<enableLldpSwPort>1</enableLldpSwPort>
<enableLldpPcPort>1</enableLldpPcPort>
<lldpAssetId></lldpAssetId>
<powerPriority>1</powerPriority>
<!--
I have the above turned on just for curiosity.
-->
<sshAccess>0</sshAccess> <!-- 0 = enabled -->
<sshPort>22</sshPort>
<!--
The above 2 lines are *required* to enable ssh on this phone, it is off
by default.
-->
<adminPassword></adminPassword>
<!--
Password to access the web interface on the phone
string
length = 32
-->
<loadServer></loadServer>
<!--
Indicates that the phone will use an alternative server to obtain firmware loads and upgrades, rather than the defined TFTP server.
This option enables you to indicate a local server to be used for firmware upgrades, which can assist in reducing install times, particularly for upgrades over a WAN. Enter the hostname or the IP address (using standard IP addressing format) of the server. The indicated server must be running TFTP services and have the load file in the TFTP path. If the load file is not found, the load will not install. The phone will not be redirected to the TFTP server.
If this field is left blank, the phone will use the designated TFTP server to obtain its load files and upgrades
string
length = 256
-->
<WlanProfile1>0</WlanProfile1>
<!--
Only on 7921/7925/7926
Indicates whether WLAN Profile 1 can be modified by the user.
If the profile is locked, the user can not modify it.
To allow the user to edit only the username and password in the profile, set it to restricted.
0 = Unlocked
1 = Locked
2 = Restricted
-->
<WlanProfile2>0</WlanProfile1>
<WlanProfile3>0</WlanProfile1>
<WlanProfile4>0</WlanProfile1>
<specialNumbers></specialNumbers>
<!--
Only on 7921/7925/7926
Special numbers are telephone numbers that do not require unlocking the phone to call.
For example, in the United States, the 911 emergency number is a good special number candidate so that it can be dialed without unlocking the phone.
You may enter one or more special numbers in this field.
To enter more than one special number, use a comma as separator.
For example, if you want to enter 411, 511, and 911 as special numbers, enter 411,511,911 in the field without spaces.
string
length = 16
comma separated
-->
<PushToTalkURL></PushToTalkURL>
<!--
Only on 7921/7925/7926
This parameter specifies the URL which the phone contacts for application services.
-->
<sendKeyAction></sendKeyAction>
<!--
This field controls the behavior of the "Send" (green) key when it is pressed. If the "Onhook Dialing" option is selected, the phone displays a list of the last numbers dialed. If the "Offhook Dialing" option is selected, the phone sends a SCCP "offhook" message to the Cisco Unified CallManager.
-->
<phoneBookWebAccess></phoneBookWebAccess>
<!--
This field controls the accessibility of the local phone book via the web and works in conjunction with the 'Web Access' parameter. When 'Web Access' is 'Disabled', the local phone book is not accessible via the web regardless of this parameter setting. When 'Web Access' is 'Read-Only' or 'Full', access to the local phone book is as follows: 'Deny All' will not allow web access for reading or writing of the local phone book; 'Allow Admin' will allow only the phone administrator to read from and write to the local phone book.
-->
<unlockSettingsSequence></unlockSettingsSequence>
<!--
Only on 7921/7925/7926
This parameter determines if a "**#" (star-star-pound) key sequence is required for write-access to the phone's settings menu. A value of 'Enabled' requires the user to enter **# first before making changes to the settings menu. If the value is 'Disabled', the user is allowed to make modifications to the settings menu without having to enter the unlock key sequence.
-->
<appButtonTimer></appButtonTimer>
<!--
Amount of time you must hold down the Application Button to activate the application. The timer values are in seconds. A value of 0 indicates that a simple push of the Application Button will active the application. For non-zero values, the application is activated after the specified timer value expires.
-->
<appButtonPriority></appButtonPriority>
<!--
This parameter indicates the priority of the Application Button relative to all other tasks on the phone. The priorities are defined as follows.
Low: The Application Button only works when the phone is idle and on the main screen.
Medium: The Application Button takes precedence over all tasks on the phone except when the keypad is locked.
High: The Application Button takes precedence over all tasks on the phone.
-->
<outOfRangeAlert></outOfRangeAlert>
<!--
Only on 7921/7925/7926
This parameter controls the frequency of audible alerts when the phone is out of range of an access point. The phone does not play audible alerts when the parameter value is "disabled." The phone can beep once or regularly at a selected interval (10, 30, or 60 seconds) when it is out of range of an access point. Once the phone is within range of an access point, audible alerts stop.
-->
<scanningMode></scanningMode>
<!--
Only on 7921/7925/7926
This parameter controls when the phone performs scanning. The parameter values are as follow.
Auto: Phone scans when it is in a call or when the received strength signal indicator (RSSI) is low.
Single AP: Phone never scans except when the basic service set (BSS) is lost.
Continuous: Phone scans continuously even when it is not in a call.
-->
<restrictDataRates></restrictDataRates>
<!--
This parameter enables or disables the restriction of the upstream and downstream PHY rates according to CCX V4 Traffic Stream Rate Set IE (S54.2.6).
-->
<powerOffWhenCharging></powerOffWhenCharging>
<!--
Only on 7921/7925/7926
This parameter indicates whether the phone powers off when it is connected to a charger or placed in a charging station.
-->
<cdpEnable></cdpEnable>
<!--
This parameter enables or disables the Cisco Discovery Protocol (CDP) on the phone.
-->
<g722CodecSupport></g722CodecSupport>
<!--
This parameter indicates whether the phone will advertise the G.722 codec to the Cisco Unified CallManager. Codec negotiation involves two steps: first, the phone must advertise the supported codec(s) to the Cisco Unified CallManager (not all endpoints support the same set of codecs). Second, when the Cisco Unified CallManager gets the list of supported codecs from all phones involved in the call attempt, it chooses a commonly-supported codec based on various factors, including the region pair setting. Valid values specify Use System Default (this phone will defer to the setting specified in the enterprise parameter, Advertise G.722 Codec), Disabled (this phone will not advertise G.722 to the Cisco Unified CallManager) or Enabled (this phone will advertise G.722 to the Cisco Unified CallManager).
-->
<homeScreen></homeScreen>
<!--
This parameter sets the phone's default home screen.
-->
<fipsMode></fipsMode>
<!--
This parameter sets the Federal Information Processing Standards (FIPS) mode for the phone. The Cisco 7926 is a FIPS 140-2 level 1 compliant device when this option is enabled.
-->
<autoSelectLineEnable></autoSelectLineEnable>
<!--
When enabled, indicates that the phone will shift the call focus to incoming calls on all lines. When disabled, the phone will only shift the focus to incoming calls on the currently used line.
-->
<bluetooth></bluetooth>
<!--
Indicates whether the Bluetooth device on the phone is enabled or disabled.
-->
<fileSystemVerificationEnable></fileSystemVerificationEnable>
<!--
This parameter indicates whether the phone will perform a file system integrity check as part of the firmware upgrade process. Enable this option to troubleshoot file system issues. This feature may impact phone performance if it is enabled.
-->
<barCodeSymbologyGroup></barCodeSymbologyGroup>
<!--
Only on 7921/7925/7926
-->
<scannerCommands></scannerCommands>
<!--
Only on 7921/7925/7926
-->
<minimumRingVolume></minimumRingVolume>
<!--
-->
<java></java>
<!--
-->
<headsetHandsetMonitor>0</handsetHeadsetMonitor>
<!--
0: will not enable the handset monitoring and only the headset will be active
1: will enable activate both the headset as well as the handset during a call
-->
<detectCMConnectionFailure>0</detectCMConnectionFailure>
<!--
Switch from measured keepalive packet delivery time falure detection to delayed failure detection
See: https://github.com/chan-sccp/chan-sccp/wiki/Unregister%2C-ReRegistering%2C-Keepalive-issues
See: http://www.cisco.com/c/en/us/support/docs/collaboration-endpoints/unified-ip-phone-7900-series/200410-Troubleshoot-IP-Phone-Unregistration-A.html
0: Normal Failure Detection
1: Delayed Failure Detection
-->
</vendorConfig>
<versionStamp>{Jan 01 2003 00:00:00}</versionStamp>
<loadInformation>P00308010100</loadInformation>
<!--
The firmware version that the phone looks for at boot.
Can be either the actual firmware version like P00308010100 or SCCP70.8-5-4S or a reference to the termXX.default file.
The later is actually much easier to maintain, just unpack the new firmware, which replaces the termXX.default.loads file.
You don't have to update all your SEP... files because it just refers them to the loads file, from which they get the
new firmware version number.
-->
<userLocale>
<name>Dutch_Netherlands</name>
<langCode>nl</langCode>
<winCharSet>iso-8859-1</winCharSet>
</userLocale>
<networkLocale>Netherlands</networkLocale>
<networkLocaleInfo>
<name>Dutch_Netherlands</name>
<langCode>nl</langCode>
<uid>64</uid>
<version>4.0(1)</version>
</networkLocaleInfo>
<!--
This is something to do with dialtones. The above config gives Dutch dialtones.
-->
<authenticationURL>http://x.x.x.x/cisco_menu/authentication.php</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)
-->
<informationURL>http://x.x.x.x/cisco_menu/help/help.php</informationURL>
<messagesURL></messagesURL>
<proxyServerURL></proxyServerURL>
<servicesURL>http://x.x.x.x/cisco_menu/menu.php</servicesURL>
<directoryURL>http://x.x.x.x/cisco_menu/directory/menu.php</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)
-->
<idleURL>http://x.x.x.x/cisco_menu/idle.php</idleURL>
<!--
URL of CiscoIPPhoneImage.
Requires a non-zero setting in idleTimeout.
-->
<idleTimeout>3600</idleTimeout>
<deviceSecurityMode>1</deviceSecurityMode>
<phonePersonalization>1</phonePersonalization>
<singleButtonBarge>1</singleButtonBarge>
<joinAcrossLines>1</joinAcrossLines>
<autoCallPickupEnable>false</autoCallPickupEnable>
<blfAudibleAlertSettingOfIdleStation>1</blfAudibleAlertSettingOfIdleStation>
<blfAudibleAlertSettingOfBusyStation>1</blfAudibleAlertSettingOfBusyStation>
<advertiseG722Codec>1</advertiseG722Codec>
<mobility>
<handoffdn></handoffdn>
<dtmfdn></dtmfdn>
<ivrdn></ivrdn>
<dtmfHoldCode>*81</dtmfHoldCode>
<dtmfExclusiveHoldCode>*82</dtmfExclusiveHoldCode>
<dtmfResumeCode>*83</dtmfResumeCode>
<dtmfTxfCode>*84</dtmfTxfCode>
<dtmfCnfCode>*85</dtmfCnfCode>
</mobility>
<dscpForSCCPPhoneConfig>104</dscpForSCCPPhoneConfig>
<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
<dscpForCm2Dvce>184</dscpForCm2Dvce>
<transportLayerProtocol>4</transportLayerProtocol>
<capfAuthMode>0</capfAuthMode>
<capfList>
<capf>
<phonePort>3804</phonePort>
</capf>
</capfList>
<certHash></certHash>
<encrConfig>false</encrConfig>
<phoneServices>
<provisioning>0</provisioning>
<!-- provisioning:
0 = Use PhoneServices
1 = Use ServicerURL / DirectoryURL and MessagesURL (Above)
2 = Use Both (Merge Both)
-->
<!-- Below
type = 0 -> applicaton menu
type = 1 -> contacts menu
type = 2 -> messages menu
-->
<phoneService type="0" category="0">
<name>Corporate Directory</name>
<url>Application:Cisco/CorporateDirectory</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="1" category="0">
<name>Missed Calls</name>
<url>Application:Cisco/MissedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="1" category="0">
<name>Received Calls</name>
<url>Application:Cisco/ReceivedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="1" category="0">
<name>Placed Calls</name>
<url>Application:Cisco/PlacedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="1" category="0">
<name>Personal Directory</name>
<url>Application:Cisco/PersonalDirectory</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="2" category="0">
<name>Voicemail</name>
<url>Application:Cisco/Voicemail</url>
<vendor></vendor>
<version></version>
</phoneService>
</phoneServices>
</device>

8
Technical.note/SEPXML.txt
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@ -1,8 +0,0 @@
http://usecallmanager.nz/sepmac-cnf-xml.html
http://usecallmanager.nz/line-keys-xml.html
http://usecallmanager.nz/user-locale.html
https://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP
https://www.voip-info.org/wiki/view/Asterisk+phone+cisco+7970+SIP
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cuipph/all_models/xsi/8_5_1/xsi_dev_guide/xmlobjects.html
https://habrahabr.ru/post/176019/
https://learningnetwork.cisco.com/thread/14585

57
Technical.note/backgroundImage.txt
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@ -1,57 +0,0 @@
https://github.com/chan-sccp/chan-sccp/wiki/Adding-custom-background-images
http://www.voicecerts.com/2011/08/changing-cisco-ip-phone-background.html
https://silver-golem.livejournal.com/432591.html
> Real Time !
http://silver-golem.livejournal.com/431942.html
General Information
Cisco IP Phones support either colored or monochrom background images in various resolutions (depending on model). The background can either be set up in sccp.conf server side or the user can be enabled to select a background image from a defined list of backgrounds. The image has to be a graphic file with .PNG extension. Other requirements apply, depending on phone model (see below).
Set background image server side
Background images can be set up server side in sccp.conf for most modern Cisco IP Phones by using a parameter in the device section. This image is pushed to the phone upon every restart.
[SEPXXXXXXXXX]
...
backgroundImage=http://PATH-TO-BACKGROUND-IMAGE/filename.png
!!!> Phone personalization needs to be set to allow the server to push background or ringtones to the phone in the SEPXXXXXXXXXX.cnf.xml of each phone:
------------------------------------
<phonePersonalization>1</phonePersonalization>
------------------------------------
Enable user to pick a custom background image
When a user is allowed to pick his own background image in the user settings (true in device section of the SEPXXXX.xml config file), the phone searches for the List.xml (case-sensitive) file in the following directories. Depending on the phone model, the required file properties are as follows:
Phone Model Image Size Thumbnail Size Directory
7906 / 7911 95x34 23x8 /Desktops/95x34x1
7941 / 7961 320x196 80x49 /Desktops/320x196x4
7942 / 7962 320x196 80x49 /Desktops/320x196x4
7945 / 7965 320x212 80x53 /Desktops/320x212x16
7970 / 7971 320x212 80x53 /Desktops/320x212x12
7975 320x216 80x53 /Desktops/320x216x16
7985 800x600 not supported /Desktops/800x600x16
8941 / 8945 640x480 123x111 /Desktops/640x480x24
The Image file is used for the background of the phone display. An additional thumbnail is used as a preview image on the phone settings menu (on 7985 only the filename). The List.xml has to be in the above model-depending directory. The file has a Cisco IPPhoneImage syntax, example:
<CiscoIPPhoneImageList>
<ImageItem Image="TFTP:Desktops/640x480x24/sccp-tn.png"
URL="TFTP:Desktops/640x480x24/sccp.png"/>
<ImageItem Image="TFTP:Desktops/640x480x24/sccp_2-tn.png"
URL="TFTP:Desktops/640x480x24/sccp_2.png"/>
</CiscoIPPhoneImageList>
While the resolution is fix, the phones are able to reduce the color depth if the original image uses too many colors.
Note: This can also be done using the SEP....cnf.xml file
<device>
...
<commonProfile>
<defaultBackground>TFTP/HTTP/HTTPS URL</defaultBackground>
<backgroundImageAccess>true/false</backgroundImageAccess> <!-- allowing users to change the background -->
</commonProfile>
<phonePersonalization>1</phonePersonalization> <!-- allowing users to change the background -->
...
</device>

11
Technical.note/dialplan.txt
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@ -1,11 +0,0 @@
Убрать коменты
в sccpgeneral.xml
- <item type="SLA-disabled" id="6"><label>Default Dial Plan</label>
+ <item type="SLA" id="6"><label>Default Dial Plan</label>
в Sccp_manager.class.php
// "sccpdialplan" => array(
// "name" => _("SCCP Dial Plan information"),
// "page" => 'views/server.dialtemplate.php'
// )

1
Technical.note/freepbx.txt
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@ -1 +0,0 @@
https://github.com/chan-sccp/chan-sccp/wiki/Monitor-and-Pickup-Incoming-Calls-via-Speeddial-Using-Custom-Devstate

58
Technical.note/help.tftprewrite
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@ -1,58 +0,0 @@
diff --git a/src/sccp_config.c b/src/sccp_config.c
index e15d19db..c78bcbf2 100644
--- a/src/sccp_config.c
+++ b/src/sccp_config.c
@@ -3062,6 +3062,7 @@ int sccp_manager_config_metadata(struct mansession *s, const struct message *m)
uint i;
const char *id = astman_get_header(m, "ActionID");
const char *req_segment = astman_get_header(m, "Segment");
+ const char *req_listresult = astman_get_header(m, "ListResult");
uint comma = 0;
if (sccp_strlen_zero(req_segment)) { // return all segments
@@ -3180,11 +3181,22 @@ int sccp_manager_config_metadata(struct mansession *s, const struct message *m)
sccpConfigSegment = &sccpConfigSegments[i];
const SCCPConfigOption *config = sccpConfigSegment->config;
- astman_append(s, "Response: Success\r\n");
- if (!ast_strlen_zero(id)) {
- astman_append(s, "ActionID: %s\r\n", id);
+ if (sccp_strcaseequals(req_listresult, "yes")) {
+ //astman_append(s, "Response: Follows\r\n\r\n");
+ //astman_append(s, "EventList: Start\r\n");
+ astman_send_listack(s, m, "SCCPConfigMetaData Follows", "Start");
+ astman_append(s, "Event: SCCPConfigMetaData\r\n");
+ if (!ast_strlen_zero(id)) {
+ astman_append(s, "ActionID: %s\r\n", id);
+ }
+ } else if (sccp_strcaseequals(req_listresult, "freepbx")) {
+ astman_append(s, "Response: Follows\r\n");
+ } else {
+ astman_append(s, "Response: Success\r\n");
+ if (!ast_strlen_zero(id)) {
+ astman_append(s, "ActionID: %s\r\n", id);
+ }
}
-
astman_append(s, "JSON: {");
astman_append(s, "\"Segment\":\"%s\",", sccpConfigSegment->name);
astman_append(s, "\"Options\":[");
@@ -3296,8 +3308,17 @@ int sccp_manager_config_metadata(struct mansession *s, const struct message *m)
comma = 1;
}
}
- astman_append(s, "]}\r\n\r\n");
+ astman_append(s, "]}\r\n");
total++;
+ if (sccp_strcaseequals(req_listresult, "yes")) {
+ astman_append(s,
+ "\r\nEvent: SCCPConfigMetaDataComplete\r\n"
+ "EventList: Complete\r\n"
+ "ListItems: %d\r\n\r\n", total);
+ } else if (sccp_strcaseequals(req_listresult, "freepbx")) {
+ astman_append(s, "--END COMMAND--\r\n");
+ }
+ astman_append(s, "\r\n");
}
}
}

15
Technical.note/make_sccp.txt
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@ -1,15 +0,0 @@
git clone https://github.com/chan-sccp/chan-sccp chan-sccp_develop
./configure --enable-indications --enable-conference --enable-advanced-functions --enable-distributed-devicestate
make
make install
load = chan_sccp.so
noload = chan_skinny.so
preload = func_db.so
preload = res_odbc.so
preload = res_config_odbc.so
preload = cdr_adaptive_odbc.so
preload = app_voicemail.so

280
Technical.note/sccp.conf.annotated
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@ -1,280 +0,0 @@
;!
;! Automatically generated configuration file
;! Filename: sccp.conf.annotated (/usr/local/asterisk-13-branch/etc/asterisk/sccp.conf.annotated)
;! Generator: sccp config generate
;! Creation Date: Sun Nov 1 01:27:41 2015
;!
;
; general section
;
[general]
;servername = Asterisk ; (REQUIRED) show this name on the device registration
;keepalive = 60 ; (REQUIRED) Phone keep alive message every 60 secs. Used to check the voicemail and keep an open connection between server and phone (nat).
; Don't set any lower than 60 seconds.
;debug = core ; (MULTI-ENTRY) console debug level or categories
; examples: debug = 11 | debug = mwi,event,core | debug = all | debug = none or 0
; possible categories:
; core, sccp, hint, rtp, device, line, action, channel, cli, config, feature, feature_button, softkey, indicate, pbx
; socket, mwi, event, adv_feature, conference, buttontemplate, speeddial, codec, realtime, lock, newcode, high, all, none
;context = default ; (REQUIRED) pbx dialplan context
;dateformat = M/D/Y ; (SIZE: 7) M-D-Y in any order. Use M/D/YA (for 12h format)
;bindaddr = 0.0.0.0 ; (REQUIRED) replace with the ip address of the asterisk server (RTP important param)
;port = 2000 ; listen on port 2000 (Skinny, default)
deny = 0.0.0.0/0.0.0.0
permit = internal ; (REQUIRED) (MULTI-ENTRY) Deny every address except for the only one allowed. example: '0.0.0.0/0.0.0.0'
; Accept class C 192.168.1.0 example '192.168.1.0/255.255.255.0'
; 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.
; using 'internal' will allow the 10.0.0.0, 172.16.0.0 and 192.168.0.0 networks
;localnet = internal ; (MULTI-ENTRY) All RFC 1918 addresses are local networks, example '192.168.1.0/255.255.255.0'
;externip = 0.0.0.0 ; 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.
;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
;recorddigittimeoutchar = no ; You can force the channel to dial with this char in the dialing state
;simulate_enbloc = yes ; Use simulated enbloc dialing to speedup connection when dialing while onhook (older phones)
;ringtype = outside ; Ringtype for incoming calls (default='outside')
;autoanswer_ring_time = 1 ; Ringing time in seconds for the autoanswer.
;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
;transfer_on_hangup = no ; 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.
;dnd_tone = 0x0 ; Use 0x2D, 0x31, 0x32, 0x33 to activate dnd incoming call indication when dnd silent is active
;callwaiting_tone = 0x2d ; Sets to 0 to disable the callwaiting tone
;callwaiting_interval = 0 ; Callwaiting ring interval in seconds. Set to 0 to disable the callwaiting ringing interval.
;musicclass = default ; Sets the default music on hold class
;language = en ; Default language setting
;callevents = yes ; Generate manager events when phone
; Performs events (e.g. hold)
;accountcode = skinny ; Accountcode to ease billing
;sccp_tos = 0x68 ; Sets the default sccp signaling packets Type of Service (TOS) (defaults to 0x68 = 01101000 = 104 = DSCP:011010 = AF31)
; Others possible values : [CS?, AF??, EF], [0x??], [lowdelay, throughput, reliability, mincost(solaris)], none
;sccp_cos = 4 ; sets the default sccp signaling packets Class of Service (COS).
;audio_tos = 0xB8 ; sets the default audio/rtp packets Type of Service (TOS) (defaults to 0xb8 = 10111000 = 184 = DSCP:101110 = EF)
;audio_cos = 6 ; sets the default audio/rtp packets Class of Service (COS).
;video_tos = 0x88 ; sets the default video/rtp packets Type of Service (TOS) (defaults to 0x88 = 10001000 = 136 = DSCP:100010 = AF41)
;video_cos = 5 ; sets the default video/rtp packets Class of Service (COS).
;echocancel = yes ; sets the phone echocancel for all devices
;silencesuppression = no ; sets the silence suppression for all devices
; we don't have to trust the phone ip address, but the ip address of the connection
;earlyrtp = progress ; valid options: none, offhook, immediate, dial, ringout and progress.
; The audio stream will be open in the progress and connected state by default. Immediate forces overlap dialing.
; (POSSIBLE VALUES: ["Immediate","OffHook","Dialing","Ringout","Progress","None"])
;dndFeature = on ; Turn on the dnd softkey for all devices. Valid values are 'off', 'on'.
;private = yes ; permit the private function softkey
;mwilamp = on ; Set the MWI lamp style when MWI active to on, off, wink, flash or blink
; (POSSIBLE VALUES: ["Off","On","Wink","Flash","Blink"])
;mwioncall = no ; Set the MWI on call.
;blindtransferindication = ring ; moh or ring. the blind transfer should ring the caller or just play music on hold
; (POSSIBLE VALUES: ["RING","MOH"])
;cfwdall = yes ; activate the callforward ALL stuff and softkeys
;cfwdbusy = yes ; activate the callforward BUSY stuff and softkeys
;cfwdnoanswer = yes ; activate the callforward NOANSWER stuff and softkeys
;nat = auto ; Global NAT support.
; (POSSIBLE VALUES: ["Auto","Off","(Auto)Off","On","(Auto)On"])
;directrtp = no ; This option allow devices to do direct RTP sessions.
;allowoverlap = no ; Enable overlap dialing support. If enabled, starts dialing immediately and sends remaining digits as DTMF/inband.
; Use with extreme caution as it is very dialplan and provider dependent.
callgroup = "" ; We are in caller groups 1,3,4. Valid for all lines
pickupgroup = "" ; We can do call pick-p for call group 1,3,4,5. Valid for all lines
;directed_pickup_modeanswer = yes ; Automatically Answer when using Directed Pickup.
;amaflags = default ; Sets the default AMA flag code stored in the CDR record
;callanswerorder = oldestfirst ; oldestfirst or lastestfirst
; (POSSIBLE VALUES: ["OldestFirst","LastFirst"])
regcontext = "" ; SCCP Lines will we added to this context in asterisk for Dundi lookup purposes.
; Do not set to an already created/used context. The context will be autocreated. You can share the sip/iax regcontext if you like.
;devicetable = sccpdevice ; datebasetable for devices
;linetable = sccpline ; datebasetable for lines
;meetme = yes ; enable/disable conferencing via meetme (on/off), make sure you have one of the meetme apps mentioned below activated in module.conf
; when switching meetme=on it will search for the first of these three possible meetme applications and set these defaults
; - {'MeetMe', 'qd'},
; - {'ConfBridge', 'Mac'},
; - {'Konference', 'MTV'}
;meetmeopts = qxd ; options to send the meetme application, defaults are dependent on meetme app see the list above
; 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
;jbenable = no ; Enables the use of a jitterbuffer on the receiving side of a sccp channel.
; An enabled jitterbuffer will be used only if the sending side can create and the receiving side can not accept jitter.
; The sccp channel can accept jitter, thus a jitterbuffer on the receive sccp side will beused only if it is forced and enabled.
;jbforce = no ; Forces the use of a jitterbuffer on the receive side of a sccp channel.
;jblog = no ; Enables jitterbuffer frame logging.
;jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
;jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs.
;jbimpl = fixed ; (SIZE: 11) Jitterbuffer implementation, used on the receiving side of a
; sccp channel. Two implementations are currently available
; - 'fixed' (with size always equals to jbmaxsize)
; - 'adaptive' (with variable size, actually the new jb of IAX2).
;hotline_enabled = yes ; 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 different circumstances, for example at a door, where you want people to be
; able to only call one number, or for unprovisioned phones to only be able to call the helpdesk to get their phone
; set up. If hotline_enabled = yes, any device which is not included in the configuration explicitly will be allowed
; to registered as a guest device. All such devices will register on a single shared line called 'hotline'.
;hotline_context = sccp ;
;hotline_extension = 111 ;
;hotline_label = hotline ;
;fallback = no ; Immediately fallback to primairy/master server when it becomes available (master/slave asterisk cluster) (TokenRequest)
; Possible values are: true/false/odd/even/script.
; active/passive cluster: true on active/false on passive
; active/active cluster: even on active1/off on active2
; more complex cluster: use script. It will be called with three arguments, namely mac-address, ip-address, devicetype.
; and it should return 'ACK' (without the quotes) to acknowledge the token, or a value for the number of seconds to backoff and try again.
; Value can be changed online via CLI/AMI command 'sccp set fallback true/false/odd/even/script'
;backoff_time = 60 ; Time to wait before re-asking to fallback to primairy server (Token Reject Backoff Time)
;server_priority = 1 ; Server Priority for fallback: 1=Primairy, 2=Secundary, 3=Tertiary etc
; For active-active (fallback=odd/even) use 1 for both
;
; device section
;
[default_device](!)
device = "" ; (SIZE: 15) device type
devicetype = "" ; (SIZE: 15) device type
description = "" ; device description
keepalive = "" ; set keepalive to 60
;tzoffset = 0 ; time zone offset
;disallow = all
;allow = ulaw ; (MULTI-ENTRY) Same as entry in [general] section
;allow = alaw
;transfer = yes ; enable or disable the transfer capability. It does remove the transfer softkey
;park = yes ; take a look to the compile how-to. Park stuff is not compiled by default.
;cfwdall = no ; activate the call forward stuff and soft keys
;cfwdbusy = no ; allow call forward when line is busy
;cfwdnoanswer = no ; allow call forward when line if not being answered
;dndFeature = yes ; allow usage do not disturb button
dnd = "" ; allow setting dnd action for this device. Valid values are 'off', 'reject' (busy signal), 'silent' (ringer = silent) or 'user' (not used at the moment). . The value 'on' has been made obsolete in favor of 'reject'
; (POSSIBLE VALUES: ["Off","Reject","Silent","UserDefined"])
;force_dtmfmode = auto ; auto, skinny or rfc2833. Some phone models with bad firmware do send dtmf in a messed up order and need to be forced to skinny mode.
; (POSSIBLE VALUES: ["AUTO","RFC2833","SKINNY"])
deny = ""
permit = "" ; (MULTI-ENTRY) Same as entry in [general] section
; This device can register only using this ip address
audio_tos = "" ; sets the audio/rtp packets Type of Service (TOS) (defaults to 0xb8 = 10111000 = 184 = DSCP:101110 = EF).
; Others possible values : 0x??, lowdelay, throughput, reliability, mincost(solaris), none.
audio_cos = "" ; sets the audio/rtp packets Class of Service (COS)
video_tos = "" ; sets the video/rtp packets Type of Service (TOS) (defaults to 0x88 = 10001000 = 136 = DSCP:100010 = AF41).
video_cos = "" ; sets the video/rtp packets Class of Service (COS).
nat = "" ; Device NAT support. Currently nat is automatically detected in most cases.
; (POSSIBLE VALUES: ["Auto","Off","(Auto)Off","On","(Auto)On"])
directrtp = "" ; This option allow devices to do direct RTP sessions.
earlyrtp = "" ; valid options: none, offhook, immediate, dial, ringout and progress.
; The audio stream will be open in the progress and connected state by default. Immediate forces overlap dialing.
; (POSSIBLE VALUES: ["Immediate","OffHook","Dialing","Ringout","Progress","None"])
private = "" ; permit the private function softkey for this device
privacy = "" ; permit the private function softkey for this device
mwilamp = "" ; Set the MWI lamp style when MWI active to on, off, wink, flash or blink
; (POSSIBLE VALUES: ["Off","On","Wink","Flash","Blink"])
mwioncall = "" ; Set the MWI on call.
meetme = "" ; enable/disable conferencing via app_meetme (on/off)
meetmeopts = "" ; options to send the app_meetme application (default 'qd' = quiet,dynamic pin)
; Other options (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 app_meetme documentation
;softkeyset = default ; use specified softkeyset with name 'default'
;useRedialMenu = no ; show redial phone book list instead of dialing the last number (adv_feature). Requires a Phone Service block in SEP....cnf.xml to work correct on Java phones (See conf/tftp/SEP example files)
;directed_pickup = yes ; enable/disable Pickup button to do directed pickup from a specific extension.
directed_pickup_context = "" ; context where direct pickup search for extensions. if not set current contect will be use.
;directed_pickup_modeanswer = yes ; on = asterisk way, the call has been answered when picked up.
monitor = "" ;
allowoverlap = "" ; Allow for Overlap dialing (Continue dialing after the first part of the number has already been send to the pstn)
setvar = "" ; (MULTI-ENTRY) extra variables to be set on line initialization multiple entries possible (for example the sip number to use when dialing outside)
; format setvar=param=value, for example setvar=sipno=12345678
permithost = "" ; (MULTI-ENTRY) permit/deny but by resolved hostname
addon = "" ; One of 7914, 7915, 7916
button = "" ; (MULTI-ENTRY) Buttons come in the following flavours (empty, line, speeddial, service, feature).
; Examples (read the documentation for more examples/combinations):
; - button = line,1234
; - button = line,1234,default
; - button = empty
; - button = line,98099@11:Phone1
; - button = line,98099@12:Phone2#ButtonLabel!silent ; append cidnum:'12' and cidname:'Phone2' to line-ci with label 'ButtonLabel', don't ring when dialed directly
; - button = line,98099@+12:Phone2@ButtonLabel!silent ; same as the previous line
; - button = line,98099@=12:Phone2!silent ; overwrite line-cid instead of appending
; - button = speeddial,Phone 2 Line 1, 98021, 98021@hints
; - button = feature,cfwdall,1234
; - button = feature,PDefault,ParkingLot,default ; feature, name, feature_type, parkinglotContext [,RetrieveSingle]
; - button = feature,PDefault,ParkingLot,default,RetrieveSingle ; feature, name, feature_type, parkinglotContext [,RetrieveSingle]
;allowRinginNotification = no ; allow ringin notification for hinted extensions. experimental configuration param that may be removed in further version
;conf_allow = yes ; Allow the use of conference
;conf_play_general_announce = yes ; Playback General Announcements (like: 'You are Entering/Leaving the conference')
;conf_play_part_announce = yes ; Playback Personal/Participant Announcements, (like: 'You have been muted / You have been kicked')
;conf_mute_on_entry = no ; Mute new participants from the start
;conf_music_on_hold_class = default ; 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.
;conf_show_conflist = yes ; Automatically show conference list to the moderator
backgroundImage = "" ; Set the Background Image after device registered. Image must be set as URI to a http served file.
ringtone = "" ; Set the Ring Tone after device registered. Ring Tone must be set as URI to a http served file.
imageversion = "" ; (SIZE: 31) ImageVersion to be loaded on the device.
;
; line section
;
[default_line](!)
id = "" ; (SIZE: 7) id
pin = "" ; (SIZE: 7) pin
description = "" ; description
context = "" ; pbx dialing context
defaultSubscriptionId_name = "" ; (SIZE: 79) Name used on a shared line when no name is specified on the line button for the device
defaultSubscriptionId_number = "" ; (SIZE: 79) Number used on a shared line when no name is specified on the line button for the device
mailbox = "" ; Mailbox to store messages in. Format 'mailbox@context' or 'mailbox' when you use 'default' context
vmnum = "" ; Number to dial to get to the users Mailbox
adhocNumber = "" ; Adhoc Number or Private-line automatic ring down (PLAR):
; Adhoc/PLAR circuits have statically configured endpoints and do not require the user dialing to connect calls.
; - The adhocNumber is dialed as soon as the Phone is taken off-hook or when the new-call button is pressed.
; - The number will not be dialed when choosing a line; so when you choose a line you can enter a number manually.
meetme = "" ; enable/disable conferencing via meetme, make sure you have one of the meetme apps mentioned below activated in module.conf.
; When switching meetme=on it will search for the first of these three possible meetme applications and set these defaults.
; Meetme=>'qd', ConfBridge=>'Mac', Konference=>'MTV'
meetmenum = "" ; This extension will receive meetme requests, SCCP_MEETME_ROOM channel variable will
; contain the room number dialed into simpleswitch (this parameter is going to be removed).
meetmeopts = "" ; options to send the meetme application, defaults are dependent on meetme app see the list above.
; 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 conferencing app for specific documentation
transfer = "" ; per line transfer capability
;incominglimit = 6 ; allow x number of incoming calls (call waiting)
echocancel = "" ; sets the phone echocancel for this line
silencesuppression = "" ; sets the silence suppression for this line
language = "" ; sets the language setting per line
musicclass = "" ; sets the music on hold class per line
accountcode = "" ; accountcode for this line to make billing per call possible
amaflags = "" ; sets the AMA flags stored in the CDR record for this line
callgroup = "" ; sets the caller groups this line is a member of
pickupgroup = "" ; sets the pickup groups this line is a member of (this phone can pickup calls from remote phones which are in this caller group
namedcallgroup = "" ; sets the named caller groups this line is a member of (ast111)
namedpickupgroup = "" ; sets the named pickup groups this line is a member of (this phone can pickup calls from remote phones which are in this caller group (ast111)
parkinglot = "" ; parkinglot assigned to this line
trnsfvm = "" ; extension to redirect the caller to for voice mail
secondary_dialtone_digits = "" ; digits to indicate an external line to user (secondary dialtone) (max 9 digits)
;secondary_dialtone_tone = 0x22 ; outside dialtone frequency
setvar = "" ; (MULTI-ENTRY) extra variables to be set on line initialization multiple entries possible (for example the sip number to use when dialing outside)
; format setvar=param=value, for example setvar=sipno=12345678
dnd = "" ; allow setting dnd action for this line. Valid values are 'off', 'reject' (busy signal), 'silent' (ringer = silent) or 'user' (not used at the moment). . The value 'on' has been made obsolete in favor of 'reject'
; (POSSIBLE VALUES: ["Off","Reject","Silent","UserDefined"])
regexten = "" ; SCCP Lines will we added to the regcontext with this number for Dundi look up purpose
; If regexten is not filled in the line name (categoryname between []) will be used
;
; softkey section
;
;[mysoftkeyset]
;type = softkeyset ; (SIZE: -1) This should be set to softkeyset
;onhook = redial,newcall,cfwdall,dnd,pickup,gpickup,private ; (SIZE: 15) displayed when we are on hook
;connected = hold,endcall,park,vidmode,select,cfwdall,cfwdbusy,idivert ; (SIZE: 15) displayed when we have a connected call
;onhold = resume,newcall,endcall,transfer,conflist,select,dirtrfr,idivert,meetme ; (SIZE: 15) displayed when we have a call on hold
;ringin = answer,endcall,transvm,idivert ; (SIZE: 15) displayed when we have an incoming call
;offhook = redial,endcall,private,cfwdall,cfwdbusy,pickup,gpickup,meetme,barge ; (SIZE: 15) displayed when the phone is taken off hook
;conntrans = hold,endcall,transfer,conf,park,select,dirtrfr,vidmode,meetme,cfwdall,cfwdbusy ; (SIZE: 15) displayed when we are connected and could transfer a call
;digitsfoll = back,endcall,dial ; (SIZE: 15) displayed when one or more digits have been entered, more are expected
;connconf = conflist,newcall,endcall,hold,vidmode ; (SIZE: 15) displayed when we are in a conference
;ringout = empty,endcall,transfer,cfwdall,idivert ; (SIZE: 15) displayed when We are calling someone
;offhookfeat = redial,endcall ; (SIZE: 15) displayed wenn we went offhook using a feature
;onhint = redial,newcall,pickup,gpickup,barge ; (SIZE: 15) displayed when a hint is activated
;onstealable = redial,newcall,cfwdall,pickup,gpickup,dnd,intrcpt ; (SIZE: 15) displayed when there is a call we could steal on one of the neighboring phones
;holdconf = resume,newcall,endcall,join ; (SIZE: 15) displayed when we are a conference moderator, have the conference on hold and have another active call
uriaction = "" ; (MULTI-ENTRY) (SIZE: 7) softkey uri action to replace default handling. Format: uriaction = softkeyname, uri[,uri...]
; . URI can be an embedded cisco action (like Key:Service, Play:1041.raw) or a URLIf uri is a url the following parameters will be added to it: devicename, linename, channelname, callid, linkedid, uniqueid, appid, transactionid