JUGGLER - downloads.globaliptel.comdownloads.globaliptel.com/nam/Phone-Encryption-JugglerVideoPhone... · JUGGLER SIP Phone Version 5.x Global IP Telecommunications, Ltd. behält

JUGGLER SIP Phone Version 5.x

Global IP Telecommunications, Ltd. behält sich das Recht vor, dieses Angebot ohne Vorankündigung zurückzuziehenoder zu modifizieren. Das Logo ist eingetragene Marke in den USA und/oder anderen Ländern. © 2005-2013 Global IP

Telecommunications Ltd. Alle Angaben, Produktbeschreibungen, Dokumentierungen ohne Gewähr.

Global IP Telecommunications, Ltd. reserves the right to withdraw or to modify this offer without prior notice. The logo isregistered trademark in the U.S.A. and/or other countries.

© 2005-2013 Global IP Telecommunications Ltd. No responsibility is taken for the correctness of the information containedin this document.

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© Global IP Telecommunications Ltd., 2005 - 2013

Inhaltsverzeichnis

SIP Softphone 3Handling of the phone Knocking menu | handling 3 Main navigation panel 5 Call line monitor 7 Context menu 10 Work area Contacts 11 Call history 15 MWI (Message Waiting Indicator) 17 Banned numbers 19 Audio Recording 20 Video 22 Shortcuts 24Settings Settings menu 25 SIP Proxy SIP Proxies 27 Voice box 30 Presence 32 Busy Lamp Field (BLF) 34 Localisation 36 Dialing rules (Outside line) 38 Advanced 39 Global settings Media Settings Audio 41 AEC 44 Network settings 45 Simple SIP Channel (SSC) 48 Voice box 50 License / Version 52 Advanced settings 54 Desktop Sharing 57 Dialing RulesCTI Interface (Snom, AAstra) Configuration 58 Dialing rules 60 Profile 61CTI Interface (Outlook) Overview 61 Call Monitor 64 Localisation (Country Code) 66 Uninstall Outlook CTI (AddIn) 68Security Security (SRTP/TLS) 68 Polymorphic "Peer-to-Peer" Encryption (PMC) 72TAPI

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SIP SoftphoneVersion Office

Knocking menu | handling

The Software Telephone is perfectly adapted to the ergonomic requirements of computer users.This applies especially to users in offices. One of the main features of the telephone is the abilityto remain off-screen in order to allow users to take advantage of the entire desktop for moreimportant tasks, rather than constantly having to move windows on the desktop due to ashortage in space.

One of the great features of the telephone is the so-called "Knocking Menu": The telephone userinterface (UI) shows up and disappears by knocking at the border of the desktop (the screen).Each of the four borders of the primary monitor is gesture-sensitive. Either the entire UI appearsby knocking at the bottom border two times or the respective part of the UI shows up byknocking two times at the let, the top or the right side of the border of the primary monitor. YOuare thus free to select which part of the UI of the telephone you want to see and to use.

The gesture is "two times quickly knocking at a border of the primary monitor".

Nummer Beschreibung 1 Call line monitor

Contains all currently active lines and all functions that are required tocontrol telephone calls as well as a "new call" area which serves as droptarget for phone book selections. The "new call" area also allows to enterphone numbers or SIP URIs manually.

2 Main navigation panelAll important global functions like "Mute", "Speakerphone", call recording,"Do not disturb", audio conference, automatic call acception, chat and thedisplay of the current dresence status are grouped here.

3 Work areaThis panel provides several tab sheets that contain important overviews

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and several main functions: Contact list, "busy lamp field", presence status,call history, recordings, video, etc.

4 All PanelsDouble-knocking here makes all three panels (the entire user interface)appear. The panels are hidden by subsequently double-knocking at thebottom border.

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Main navigation panel

Nummer Beschreibung 1 Connected to the SIP Server

Symbol flashes "green" upon connect to a SIP Server and red if an errorhas been occured.

2 Vokume control for microphone and speaker

3 Mute (microphone)

4 „Hands free" (Speaker output to “external ringtone”)

5 Audio Recorder (is activated during a conversation)

6 Opens the Video window

7 Auto Conference (AC) Combined with the Auto-Answer feature, these two features can control theway participants are added to a conference

8 Do not Disturb (DND)incoming calls will be sent to voicemail or callers will receive a “busy”indication. This feature differs between SIP service providers.

9 Auto Answer (AA)The phone immediately answers a call

10 ConferenceIf there are calls on more than one line (on hold or active), a simple click atthe "conference" button is sufficient to create a conference. All parties thathave previously been on hold or active are instantaneosly added to theconference. If you click at an unused line (a line that is idle) while being in aconference, e.g. to reply to an incoming call or to make a new call, allparticipants of the conference are set "on hold". By clicking at the"conference" button, the conference between all active parties and theparties that had been "on hold" is reestablished.

11 Chat / SMSOpens a neutral chat window where an outgoing SIP number anddestination number can be specified. Sent messages are, if supported bythe PBX / SIP provider, sent to mobile phones as short message andincoming SMS as Instant messages.

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12 Logg off an autoprovisioned account (depends on provider)This functions loggs off the registered SIP account. After logging off thephone opens as default a custom login window for entering username &password and restarts the autoprovisioning process.

13 Restart the phoneEnds the phone and restarts.

14 SSC - Simple SIP Channel (service dependent)Uses a HTTP connection through a SSC server to be able to make andreceive calls in difficult network environments (firewall), limited wirelesshotspot, etc.

15 Hardphone Mode (CTI) - Switch ButtonChanges the Software mode between Softphone and Hardphone mode.Details are described in "CTI interface".

16 Presence status und quick change buttonDisplays your current presence status and enables quick and easytransition to other presence states.

17 Short Keysallows own assignments of short keys (keyboard layout) for the mostimportant telephone functions.

18 Contact list with BLF (Busy Lamp Field) and PresenceOpens the contact overview including search criteriums for individual filters.The filling of the contact list can also be done automatically by serviceproviders through a provisioning system.

19 MWI (Message Waiting Indicator)Will be shown if you have missed calls or if you have voicemail messages.

20 SettingsOpens the telephone settings.

21 HelpOpens the help of the telephone.

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Call line monitorLine monitor | Drag & Drop

Each call is displayed as one line in the line monitor. This enables to initiate simple and specific actions foreach telephone line/call.

All advanced functions like "call transfer", "attended transfer" or "call deflection" (transfer without prior callacception) can easily be made by Drag & Drop.

Overview on the functionality of the line monitor:

Nummer Beschreibung 1 Anonymous calls

Activate this feature to make the next call "anonymously". The called partydoesn't receive a phone number. You can set a permanent "anonymous"flag in the SIP-Account settings.

2 New Call - Input field for dialling a phone numberPlease enter the phone number which you want to call in the edit control.With a click at the arrow you can select a number from the last 20 dialedphone numbers.

3 DialDials the entered phone number in the edit box.

4 RedialDials the last dialed number.

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5 Caller informationDisplays phone number or name specified by the PBX. Duration of thetelephone call and the status information to the current call are alsodisplayed.

6 HoldBy clicking at an active line during a phone conversation, the line is set tothe "on hold" state and a "Pause" symbol will be shown. If your PublicBranch Exchange (PBX) supports "music on hold", the party that has beenset on hold will hear that music. In order to unhold a line, simply click at theline and a "Speaking" symbol will be displayed (see reference markr 10).You'll then be able to continue your conversation with the remote party thatwas previously put "on hold".

7 EncryptionIf the remote party supports encryption, the "encryption symbol" isdisplayed (a little key icon). By clicking at this key icon, the telephone triesto initiate call encryption. If the attempt was successful, the icon turns togreen colour. It remains red if the attempt was not successful. If "Peer-2-Peer encryption" is active, you further receive a short authenticationmessage (a short HASH value). Only if both parties can read the samehash value, the telephone call cannot be tapped over the entire line.

8 CodecShows the currently used Audio and Video Codec.

9 Desktop Sharing If the remote party supports Desktop Sharing, you can see the DesktopSharing Icon on your call line. The icon depicts two computers located nextto each other. In order to launch a Desktop Sharing session, simply click atthe symbol. The remote party is subsequently and directly invited to a newDesktop Sharing session. The Desktop Sharing Module runs by default fortest purposes in Demo Mode for up to 5 minutes (currently). The DesktopSharing module requires a separate licence. Please contact us or yoursupplier.

10 Hold & Selektive Telephone ConferenceFor the "Hold" function, please see ( 6 ).

Selective conference: In case there are a number of lines currently activeand thus calls in progress and if some calls are "on hold", then it is possibleto add all calls to a conference at once by clicking at the "conferencebutton" in the main menu (top panel).

Above of this, you can select a number of lines specifically to initate aconference.This "selective conference" is initiated by removing the "on holdstate" for each single line that you want to participate in your conference.All the lines that are in conversation mode take part in the conference.Those that are "on hold" do not take part.

11 Hang Up | IgnoreEnds the active call.

An incoming call can be ignored for one time with this function. The callerwill receive the status "Busy" or receives a "dial tone" (depens on settings).

12 Take callAn incoming call will be connected.

13 BanThis feature blocks a specific caller ID. Incoming calls from banned partiesare automatically handled as if "Do Not Disturb" was activated. Bannedparties are listed in a "Banned Parties" list and can be unblocked by simply

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deleting the corresponding entry. The caller will receive the status "Busy"or receives a "dial tone" (depends on settings).

14 Contact areaShows a picture of the callee if the PBX sends such a photo. A click ontothe preview picture will enlarge the image by four times.

Drag & Drop Functions:

For Blind Transfer, Attended Transfer or Call Deflection, intuitive Drag & Drop is used.

Example:You have two parties that you want to connect with each other!This is easily performed by clicking at the left mouse button on a call line, keep the mouse button pressed unddrag the call onto the party that you want to forward the call to (corresponds with an attended transfer).

As soon as you keep the mouse button down, a "Drag & Drop" window is displayed. On top is the always theparty of origin displayed, from which the action was initiated. Below is always the target displayed. If a"technically" valid target is underneath, the "Drag & Drop" box goes green. If the target is invalid (e.g. whentrying to transfer to the same party of origin), then the "Drag & Drop" box goes red.

Attended Transfer: Is initiated by simply dragging the telephone line of the originating party onto the the tlelephone line of thesecond party.

Transfer (Blind transfer): Simply enter the phone number of your transfer target. Then drop the telephone line of the caller onto the editbox. If the targetted person is already in your list of contacts, you can alternatively drag the origin onto a phonenumber of the desired contact and drop it there. It is finally also possible to drag and drop a phone number ofone of your contacts on the line that is to be forwarded.

Call Deflection: In case on an inbound call that you don't want to answer, please don't accept the call. Simply proceed asexplained in "Transfer (Blind transfer)".

New call: Please enter the telephone number that you want to call in the edit box and press the "RETURN" key or click atthe "Dial" button ( 12 ). Alternatively you can drag and drop a phone number from the contact list onto the editbox or you can select an arbitrary text in any text application (editor, web browser, etc.) and hit the "PAUSE"key.

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Context menu

Context menu in the system tray

The context menu is accessible by clicking at the phone icon in thesystem tray. Alternatively you can access the context menu byclicking at the right mouse key when the mouse pointer is locatedwithin the visible area of the softphone.

Nummer Beschreibung 1 Select a default SIP account

If you have more than one SIP account active you will get the SIP accountstatus at the context menu. Green means everything works well, Redmeans that any error occured.

To choose a SIP account as your default account for outgoing callplease click at the SIP account. It will be marked with bold letters.

2 Show phoneThe phone is moved to the foreground by clicking here.

3 LanguageAll currently available languages are displayed with this option. If youchoose a language that differs from the current setting, the phone willimmediately redraw the user interface with the new language.

4 SettingsA click at this option opens the phone settings dialog.

5 SkinsYou will get a list with selectable skins.

6 Select monitorSelect the monitor where the phone shall be displayed. This menu is onlyavailable if you use more than one monitor.

7 Call historyCall logs: Outbound, Inbound, Missed and Dialled.

8 Launch on bootStarts the softphone when the operating system boots.

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9 Online StatusIf your SIP service supports the "Presence" function (RFC standard), thenyou can select your presence activity status here.

10 Desktop SharingAllows the manual invitation of any desktop sharing participant.

11 QuitTerminate this softphone application.

ContactsContacts

Nummer Beschreibung 1 Search contact (Filter)

Shows a contact which matches with the entered text string

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2 ResetResets the Search string and shows all entries

3 Title of a contact group

4 PresenseShows the activity status of a contact

5 Busy Lamp Field (BLF)Shows the classic call status of a call:

Terminated The party has no phone call

Incoming call (Early) The party receives an incoming call. Depending on the PBX you are able topick up the call by double clicking the shown call

Connected (Confirmed) The party has a phone call

6 Pick up a call (yellow)If the PBX supports "Pick Up" you can take a call by double clicking theincoming call in the overview.

7 Contact details - context menuClick the right mouse button on a contact to open the contact detail menu.Here you can click at Add, Edit oder Delete contact.

By double clicking a contact in the overview the contact will be callesautomatically. Contains the contact more than one phone number a PopUp menu will be opend where you can select the phone number whichsgall be dialed.

8 Contact Quick InformationBy selecting a contact, detail information of this contact will be shown inthis area.

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Funktionstaste Beschreibung 1 Phone numbers

Enter up to four phone numbers for each contact.

2 Contact detailsEnter contact details like name, address or eMail address

3 Activate Presence and Busy Lamp fieldThe Presence and Busy Lamp Field status will only be sent and received ifyou have activated this feature in the SIP account settings.

ATTENTION:

The Presence support must be activated in each SIP account to preventunwanted information exchange.

If you want to share your activities with other parties you have to activate thePresence/BLF features for each SIP account!

4 Group nameAllocate a contact to one of the meaningful group names. If you don't select a

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group a contact will be assigned to the group "Default".

5 Presense SIP-IdentifierTo enable the receiving of any Presence information, the monitored phoneextension must register at your PBX. Enter a valid phone number or SIP-name which shall monitored. If you have corresponding rights on your PBX,you can receives all status information of the entered extension.

Correct : 41Correct : happy.james

False : [email protected] : [email protected]

The SIP-Suffix (No.6) will be added automatically!

6 Presence state / Allocate a SIP accountAllocate the entered extension (No.5) to a SIP account. This is needed tosend queries to the right PBX where the entity is a regular member.

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Call historyCall history

No. Description 1 Show incoming calls

2 Show outgoing calls

3 Show missed calls

4 Show not reached calls

5 Show calls with voice messages (Softphone voice box)

6 Call the selected entry

7 Delete the selected entry

8 Delete the complete call history

9 Mark all entries of the call history as "read"

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10 Open the selected entry with Microsoft Outlook (if Outlook contact)

11 Create a new contact from the selected entry

12 Edit the selected entry if the entry is a local contact

13 Edit the phone number of the selected entry

14 Dials the phone number of the Edit field (No. 13)

15 List of Call history entries

16 Play voice message of the selected entry

17 Stop the playback of the voice message

18 Duration and Position of the voice message

19 Volume of the playback of a voice message

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MWI (Message Waiting Indicator)MWI (Message Waiting Indicator)

No. Description 1 Wähltaste

Select a SIP account and click the green phone button to call your PBXvoicebox. Attention: you need to configure the SIP settings previously.

Enter the phone number you have got from your Provider for calling yourvoicebox. After entering the phone number you can call your voiceboxdirectly from the MWI overview by double clicking an MWI entry.

You find this option in your SIP account settings ("Voicebox").

2 SIP accounts (phone lines)Overview of all configured and active SIP accounts.

3 New messagesCount of all new voicebox messages on your PBX

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4 Old messagesCount of all old and not deleted messages on your PBX

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Banned numbersBan – List of banned numbers

No. Description 1 Delete selected entry

Deletes the selected entry. By deleting a phone number the caller will besignaled again with an incoming call.

2 Delete listDeletes the complete "BANNED" list with all entries.

3 Banned callerIf you ban an incoming call, the corresponding phone number will be storedin the ban list. If the caller calls you, he will be blocked automatically andyou won't here any signal for an incoming call!

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Audio RecordingAudio Recording

No. Description 1 Delete recording

Deletes a highlighted recording.

2 Delete listDeletes the complete list.

3 Recording listOverview of all recorded calls including date, local party, remoteparty and call duration.

4 PlayClick to play the selected record entry.

5 StopStops playing the recorded call.

6 PositionPlay position of the record call with total duration.

7 Volume

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Volume output when playing a recorded call.

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Video

Nummer Beschreibung 1 Info window of the running video transmission (No. 6)

2 Video qualityClick at on of the "person icons" to change the Bitrate of the videotransmission. Sie auf eines der senkrecht angelegten Personensymboleund wählen die Videoqualität. All the more the icons are qualitativ higherthe bitrate increases. Attention: Quality depends on you available internetbandwidth. If the remote party receives a fragmented video then select alower bitrate.

3 Start / Stop video transmissionClick the Play symbol to start the video transmission. The icon will changeto a Stop symbol. Click again to stop the playback.

4 Automatic video transmissionActivate the automatic video transmission with a click (green Hook ifactivated)

5 Picture sizeSelect the transmitting picture size (SubQCIF, QCIF, CIF, 4CIF) Attention:you need more bandwidth with higher sizes. If the remote party receives a

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fragmented video then select a lower bitrate.

6 Activate / Deactivate the Video Transmission Details (No. 1)

7 Change Video device & Video manufacturer settingsClick to see and select all available Video devices. Select the wanteddevice to switch in realtime to the selected device. Furthermore you canclick at "Webcam vendor settings" to open settings of the manufacturers.

8 Video device nameShows the name of the selected webcam

9 Video Full screenBy double clicking the video window the video will be resized into afullscreen mode. Click again to minimize the Video again.

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Shortcuts

Nummer Beschreibung 1 Describes the function of a shortkey

2 Short keyChange to the desired short key input field which you want to change.Press the desired key combination with which you want to perform theoperation. Confirm the short key by clicking "Apply" or "OK.". Theconfiguartion represent only "valid" key combination which the systemoffers.

Short keys won't work during the short key configuration is opened. Fortesting please close the window!

Important: Please note that not "all" key combinations will always workregarding to other software products.

Shortcuts Beschreibung Alt+Ctrl+F9 Open / Close - Main navigationAlt+Ctrl+F10 Open / Close - Line monitorAlt+Ctrl+F11 Open / Close - worker areaAlt+Ctrl+Space Open / Close - all windows

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Ctrl+F1 HelpCtrl+F2 SettingsCtrl+F3 ContactsCtrl+F4 Call historyCtrl+F5 Hold / UnholdAlt+Ctrl+o Start / stop recording

Alt+Ctrl+a Take a callAlt+Ctrl+h Hung up a callAlt+Ctrl+r RedialAlt+Ctrl+m Mute

PAUSE Mark any phone number in any application and press PAUSEto start immediately the call process of the phone.

Alt+Ctrl+Pfeiltaste links Microphone volume downAlt+Ctrl+Pfeiltaste rechts Microphone volume upAlt+Ctrl+Pfeiltaste hoch Speaker volume upAlt+Ctrl+Pfeiltaste runter Speaker volume down

Protokoll handler commands for dialing by using LINKs in webpages (suported with commercial versions).

Shortcuts Beschreibung callto://IhreTelefonnummer starts the phone automatically and dials the numbersip://IhreTelefonnummer starts the phone automatically and dials the numbersip2://IhreTelefonnummer starts the phone automatically and dials the numbertel://IhreTelefonnummer starts the phone automatically and dials the numberhangup://IhreTelefonnummer hangs up the call

Settings menu

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No. Description 1 SIP Proxies

By selecting an entry from the 10 available slots you’ll access the settingsdialog for the corresponding slot. Up to 10 VoIP services can beconfigured.

Correctly setup accounts are marked with a green icon.

Accounts with erroneous settings are marked with a red box. In thesecases it is impossible to log on the SIP proxy of the respective VoIP serviceprovider.

Call „l + Proxy“ [l1] to initiate the call with the wanted Proxy, e.g.:

l20044123456789 à Anruf mit Proxy2 mit 0044123456789

Alternatively you can also choose an individual Prefix like „999“ andmanage proxies with the Dialing Rules.

2 Globale settingsCommon settings which affect to alle SIP accounts.

3 Language settingsChanges the phone localization

4 Media settingsConfiguration of audio devices, echo cancellation (AEC) and ringtones

5 Network settingsConfiguration of network-related parameters like STUN, SIP- and RTPports

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6 Voice boxConfiguration of your local voicebox

7 License InformationHere you find your license information and you can import / export thephone license including your configuration.

8 Advanced settingsDetailed settings of the phone like: call waiting, minimize etc.

9 Dialing rulesConfiguration if individual dialing rules

10 USB Headset devices (Attended Headset Support)Configuration of available USB End Devices like JABRA, Yealink,Plantronics CS 50 / CS 60 / CS 70 to connect them with the phone forusing headset buttons like: get call / hang up call

SIP Proxies

No. Description

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1 PreselectionA list of preconfigured SIP services world wide. Select a service topreconfigure all settings automaticalls (Proxy, realm, etc.)

2 Confimation button (Preselection)Click the confirmation button to confirm the selected preconfiguration.

3 ActiveSet to „yes“ in order to activate the SIP account

4 Verwende Peer-to-PeerActivating this function enables you to make calls without SIP proxy. Youonly need to provide the user name and the port for the communication totake place. The port is specified with the option "Global settings" ->"Network settings" -> "Use local SIP port". All participants must use thesame port.In order to enable for dialing complete SIP URIs please deactivate"Advanced settings" -> "Ignore SIP suffix". The phone can now be reached under the folowing URI: username@IPaddress

5 Proxy namePlease enter a meaningful name for the SIP Proxy (e.g. OurPBX orSipProvider1)

6 Domain / RealmDomain name of your SIP service provider. This parameter is supplied byyour SIP service provider!

Often "Domain / Realm" and "SIP-Server" are the same!

7 Proxy-ServerProxy server address of the SIP registrar server of your SIP provider. Canbe an IP address or a URL that is later parsed by a name server (e.g. sip.mysipprovider.com). A registrar server manages the registering of internettelephones.

8 SIP PortA logical port used to communicate with the SIP registrar server. Mostcommonly used: Port 5060.

9 Reregister (Timeout)Interval in seconds in which the phone shall update its registration with thePBX system. The specified default value is 300 seconds. The telephonesystem (PBX) can force its own interval up to 3600 seconds (common).

10 Outbound ProxyA proxy that receives requests from a client, even though it may not be theserver resolved by the Request-URI. In some cases, the outbound proxy isplaced alongside the firewall and is the only way to let SIP traffic pass fromthe internal network to the Internet.

Proxy server address (IP or named URI) of the server that handlesoutbound calls

11 Outbound Proxy PortA logical port used to communicate with the outbound server.

12 Display name

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The name that is displayed by the telephones of your remote parties. Thedisplay name is your caller ID. It can contain space characters.

13 User nameYour username/user ID of the account at your SIP telephony serviceprovider. This parameter is supplied by your SIP service provider.

14 PasswordYour password for the SIP account. This parameter is supplied by your SIPservice provider.

15 Auth User (Authentication User)Depending on your SIP telephony service provider, an additional username can be provided by your service provider. Please enter thisparameter in this field.

16 AnonyousSelect "yes" to make permanently "anonymous" phone calls with this proxy.This function must be supported from your SIP service to take effect.

17 DeleteDeletes the settings of this SIP account configuration.

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Voice box

No. Description 1 Activate voice box

Select the checkbox "Activate voice box" to enable the feature

2 Dauer der Rufannahme ab AnrufeingangEnter the time in seconds when the voicebox shall start.

3 Aktivate Online voice box)Activate this feature to use an online voice-box, external voicebox which isavailable by a landline number.

4 Forward to URLSpecify a SIP URI where the softphone transmits callsusing a "blindtransfer". Alternatively you can use this feature for simply forwarding aphone number.

5 Use local voice boxThe softclient uses the built-in voice box. Please select one pre-recordedannouncement scheme for each SIP account. Maximum recording durationcan be set to any value between one and two minutes. Standardannouncements are output in the language the softclient is set to.Individual announcements are valid globally (independent of any languagesetting).

6 Use announcement schemeSelect a scheme which shall hearable for the callee.

7 Recording duration

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Enter the maximum recording time in minutes.

8 Phone number to query voice boxEnter the phone number you have got from your Provider for calling yourvoicebox. After entering the phone number you can call your voiceboxdirectly from the MWI (Message Waiting Indicator) overview

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PresencePresence status

The presence status shows the activity which the phone owner has just in time. You find all presence states inthe contact overview o fthe phone.

No. Description 1 Activity / Presence status (Presence)

2 Busy Lamp Field (BLF)

At the Top navigation you can change your own presence state quickly by using the "quick selection" area asdescribed below:

Simply select a new status symbol to change your presence state globally.

Prepare a contact for Presencing

The Presencing feature is based on managed contacts. It is recommended to add corresponding“Presence contacts” on both your softphone installation as well as on the remote softphoneinstallation so that the two clients can identify matching contacts (or other compatible SIPdevices).

Enable/disable "Presencing" by selecting "Yes" in the Proxy menu "Presence".

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Example: User Alice has SIP account 40. Bod has SIP account 41. Alice’s SIP account 40 can beassigned the contact information of remote user “41” (Bob) and Bob’s SIP account 41 can beassigned a contact with user “40” (Alice). In the above screenshot the SIP Account 40 (Alice)gets presence information of user 41 (Bob) assigned.

Please fill in the contact data first. In the phone number area you can use any telephone number(SIP, Landline or Mobile) which you are normally using for your calls.

For some information the phone changes the status automatically, e.g. if you're "On the Phone"or Available. Open the phone context menu and choose "Automatic" like shown in the picturebelow.

More information about how to configure contacts for presence and BLF please go to the menu"contacts"

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Busy Lamp Field (BLF)Busy Lamp Field (BLF | Busy Lamp Field)

The Busy Lamp is a classic system telephone feature, to make several telephone states of a participantsavailable.You can find the Busy Lamp Field to each of your contacts in the contact overview.

Number Description 1 Activity/Presence status (Presence)

2 Busy Lamp Field (BLF)

There are three different Busy Lamp Field states:

Status Description

TerminatedThe party has no phone call

Incoming call (Early)The party receives an incoming call. Depending on the PBX you are able topick up the call by double clicking the shown call (PickUp).

Pick up a call (yellow)If the PBX supports "Pick Up" you can take a call by double clicking theincoming call in the overview.

Connected (Confirmed)The party has a phone call

Activate Presence and Busy Lamp fieldThe Presence and Busy Lamp Field status will only be sent and received when you have activated this featurein the SIP account settings.

ATTENTION:

The Presence support must be activated in each SIP account to prevent unwanted information exchange.

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If you want to share your activities with other parties you have to activate the Presence/BLF features for eachSIP account!

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LocalisationLocalisation (Country Code)

Please enter for each SIP account the correct "Country Code". The phone checks all existing phone numbersof your contacts. If any of them matches you will get shown the contact Name and Firstname in the Display andin the Call History. The phone number will be shown as Tiptext if you move the Mouse Icon over the phonenumber in the display.

Nummer Beschreibung 1 Country Code

Pleasy enter your country here in order to enable the telephone to correctlydetect contacts on inbound and outbound calls.

2 Dialing rule for national calls

3 Remove country code Only removes the country code that was entered in the edit box „Countrycode“ (1). The remaining digits are called. This selection is only useful orcountries that do not require leading digits for the area code, e.g. "0" inGermany (069).

4 Replace country code with Only replaces the country code that was entered in the edit box „Countrycode“ (1) by the string entered here.

5 Dialing rule for international callsReplaces the leading "+" by "00" for outbound international calls. Thisselection is applicable for VoIP services that strictly require telephonenumbers for international calls to begin with "00".

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6 Replace + or 00 by ...Replaces the leading "+" or "00" for outbound international calls by thestring entered here. This selection is applicable for VoIP services or incountries that require a prefix that differs from the common standard. E.g.it is necessary to make a call from the U.S. to Great Britain (+44) by dialing01144 in lieu of 0044.

IMPORTANT!

Please add the localization to your SIP account(s) by entering the country code for your country!

If your phone numbers in MS Outlook are formatted in the canonical format (for the U.S.: 001, forScwitzerland: 0041), these numbers cannot be identified without previously executed localization.

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Dialing rules (Outside line)Outside line

No. Description 1 Activate "Outside Line rule"

Activate / Deaktivate the "outside line" dialing rule.

2 Maximum lengthChoose the maximum length of your internal telephone numbers. If the callednumber is longer than the value of the maximum length, the phone dials the"outside line" number automatically as a prefix.

3 Outside Line numberEnter the number which shall be dialed as a Prefix to get an "outside line".

4 Drop outside-line with incoming callDrops the outside-line from the regular dialed phone number, e.g.:0-069-123456789 will be changed to 069-123456789

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Advanced

No. Description 1 Default values

Arranges the audio codecs to the default settings, optimized for broadbandaccess (high-speed DSL) or narrowband access (slow DSL).

2 Audio CodecsVoice encoders/decoders (codecs) are activated and deactivated byenabling/disabling the corresponding checkboxes.

Depending on your preferences, you can set the priority for each codec byclicking at the buttons depicting arrows. Otherwise will your phone transmitthe preferred codec order to the server and a suitable codec will be chosento establish your call. By clicking at the “Reset to defaults” button will thedefault order be restored, depending on your environment setting (narrowbandwidth or broadband).

3 Video CodecsVoice encoders/decoders (codecs) are activated and deactivated byenabling/disabling the corresponding checkboxes.

Depending on your preferences, you can set the priority for each codec byclicking at the buttons depicting arrows. Otherwise will your phone transmitthe preferred codec order to the server and a suitable codec will be chosento establish your call. By clicking at the “Reset to defaults” button will thedefault order be restored, depending on your environment setting (narrowbandwidth or broadband).

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4 PBX hinter NAT / lokale IPIf a SIP server is located in the same network like the phone, than youselect „YES" in the Pulldown menu. That ensures, that the domain nameor IP will be dissolved correctly.

PBX behind NAT/ local IP -> YES (activated)The phone sends the local IP address fo the computer to the PBX. ThePBX in the Local Area Network (LAN), sends data back to the receiced IP.

PBX behind NAT / local IP -> NO (deactivated)The phone sends the public IP (internet connection) to the PBX. To receivethe public IP a STUN server is needed (network settings). The PBX in theWAN (Internet) received and sends data packets to that public IP address.Here your Router nedds NAT support.

In a few cases it could be possible that it's needed to send local IPaddresses (really rarely).

5 Use DNSThis function is preconfigured and shouldn't be changed. In case that theSIP service transmits its configuration data through SVR of the DNSprotocol, this function can be used. In this case configuration is performedfully automatically. If the server does not support this feature, runtimeproblems are likely to occur.

6 Local call signalNormally a VoIP PBX plays the "ringing" audio signal with an outgoing call.Anyways, in a few cases the PBX doesn't play anything. Then you havethe possibility to change the settings manually.

The default setting is "AUTO" and should not be modified because thephone decides automatically if it receives "early media" or not.

Change to "YES" if you want to emulate a local ringtone. Attention: In thismode you won't here any "early media" like annoucements for phone costs,etc.

Select "NO" to force only "early media" which is send by the PBX.

7 DTMF (Dual Tone Multiple Frequency)Please select the dial tone setting that your SIP service is compatible with(default: Outband).

DTMF ist used for transmitting digits during a call (DTMF dialing, MFV).

Inband: DTMF-digits are transfered in the voice band (reliable over G.711).

Outband: DTMF-digits are transmitted outside the speech band bysignaling RFC 2833.

8 Video on DemandBy default, this function should be enabled and aims that the SIP servernegotiates VIDEO support directly with the SIP account registration. Someservices are requiring the renegotiation of VIDEO Codec support! In thatcase please set the value to "No".

IMPORTANT: An incorrect setting of 'Video on Demand' can occursregistration problems and errors during phone calls!

9 Individual ringtone (each SIP-Konto)Enable the custom ringtone for the selected SIP account to give a musicalscore for incoming calls. By activation all ringtrones delivered by the PBX

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are available (Alert-Info |*. WAV audio format).

10 Select an individual ringtoneClick at "..." to select an alternative URL or a local music file. Common*.wav files are supported. Is a file selected, it has always priority.

AudioAudio

The dialog box contains controls for setting up audio devices. Please choose your preferred audio devices forthe microphone (Input device), the speaker (Output device) and for playing back the ringtone.

The controls in the third line are a speciality. With these controls you can set the overall volume of thetelephone for your input as well as for your output WITHOUT affecting system or device settings.

No. Description 1 Input device

Select a device for voice input (microphone). Regarding to quality issuesan USB audio device (eg JABRA) is recommanded.

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2 Output deviceSelect a device for the voice audio output (eg Headset)

3 RingtoneSelect a speaker device for playing ringtones.

4 Volume of the Windows Audio MixerShows the original Windows OS Mixer., including text label which comesfrom the audio device manufacturer.

5 Silent | Mute - Windows Audio MixerShows the original Windows OS Mixer für selecting für "Silent | Mute"on/off.

6 WAVE Volume of the Windows Audio MixerShows the original Windows OS WAVE Mixer., including text label from theaudio device manufacturer.

7 Stumm / Lautstärke WAVE ausschalten des Windows Audio MixerShows the original Windows WAVE OS Mixer. ON / OFF "silent"

8 Volume - microphoneOwn phone volume setting which depends on the basic MIC input volumeof the windows OS.

Example: The Windows side MIC input level of the device is 50%.. adjustthe microphone input level up to 100%, then the real volume is 50%.

9 Volume - Speaker for voice audioOwn phone volume setting which depends on the basic volume of thewindows OS.

Beispiel: The Windows side speaker volume of the device is 50%.. adjustthe phone speaker volume up to 100%, then the real volume ootput is50%.

10 Mic BoostSome Audio devices don't have the classic MIC Boost support ( x 2). Toget a better (louder) microphone signal you need only to activate thatphone function.

11 Automatic device configuration (Zero touch)Detects new audio devices automatically and use them as the defaultdevice for the phone. If you deactivate (plug off) an audio device theprevious audio device will be activated again.

12 RingtonesYou can select from the built-in ringtones : „Standard“, „Feeling free!“ and„Fighting mermaid“. Output volume can be adjusted with the ringtonevolume control.

13 Speaker | Ringtone testClick at the speaker button to hear the selected ringtone.

14 PC SpeakerCheck "Activate PC Speaker" to get a simple ringtone on the integrated PCspeaker.

15 User defined ringtonesSelect a *.WAV oder *.OGG file as ringtone to play individual sound withincoming calls.

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The dialog root directory points to the "Ringtones" directory od the phone,which includes several ringtones (*.OGG).

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AECAEC

AEC stands for Acoustic Echo Cancellation. This option enables you to optimize sound quality when using aloudspeaker instead of a headset. Use the level control to adjust the delay. We recommend the use of a goodheadset to take advantage of the best audio quality.

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Network settings

No. Description 1 Use STUN Server

STUN is a protocol that allows the discovery of ports that are suitable forcommunication through NATs (Network Address Translators, RFC3489) sothat it’s possible to establish a voice connection.You don’t have to use aspecific STUN server (e.g. the one of your provider). Any public STUNserver will do.

2 STUN Server AdresseEnter any STUN Server address. It's not needed to use a "special" STUNserver from SIP service. Entering any valid STUN server address isenough.

3 Use local SIP-Port (UDP Ports)The phone determines all ports dynamically by default. By setting thecheckbox to the checked state you are forcing the phone to use the portnumbers defined by you. This feature is very useful when setting up afirewall so that Network Address translation is facilitated.

4 SIP PortEnter the whished SIP Port

5 Use local RTP Port (AUDIO und VIDEO) - (UDP Ports)The phone determines all ports dynamically by default. By setting thecheckbox to the checked state you are forcing the phone to use the portnumbers defined by you. Setup of a firewall is easy because only two RTPports (Audio and Video) and one SIP port needs to be reserved.

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6 Audio Port (UDP)Enter the whished UDP Port

7 Video Port (UDP)Enter the whished UDP Port

Information zu Quality of ServiceWith Windows Vista OS and later, the phone supports Quality of Service.Quality of Service. Quality of Service is needed for prioritisation datapackets. Each packet is signed with a value, so a switch or router forwardsdata packets depending on the attached information. Important: QoS mustalso suported by your hardware (router, switch).

Following QoS level are available:

BestEffort | 0x00

Background | 0x08 | Class1

ExcellentEffort | 0x28 | Class5

AudioVideo | 0x28 | Class5

Voice | 0x38 | Class7

Control | 0x38 | Class7

The QoS level, eg 0x38, is appended to any data data packet. This datapacket is used to determine with which priority a switch or router shalltransport the data packet through the network.

8 Quality of Service (QoS) - SIPSelect the prioritisation level.

9 Quality of Service (QoS) - AudioSelect the prioritisation level.

10 Quality of Service (QoS) - VideoSelect the prioritisation level.

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Simple SIP Channel (SSC)

No. Description 1 Activate SSC Gateway

Activates the SSC Gateway. Simple SIP Channel (SSC) opens a new andunique possibility to you to make VoIP calls from any network environmentto your PBX at home, to your office or with an arbitrary SIP VoIP servicelike e.g. 1und1, sipgate or A1. You neither need a VPN System nor anadministrator to make or receive calls from within blocked networks orVoIP blocking countries. A simple connection that allows you to browseweb pages in the internet is sufficient to make and receive VoIP calls.

The phone client detects automatically proxies and supports also automaticconfiguration scripts (*.PAC files).

2 SSC Gateway (Host) adressEnter a valid SSC Gateway address

3 SSC Port (TCP)Enter the TCP Port of the listening SSC Gateway. As default SSC offersPOrt 443 and Port 80 or POrt 9999. Administrators can select any TCPPort so ask them if you are not sure which Ports are in use.

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4 Use Proxy (HTTP)In some cases, the whole traffic (e.g. in companies) is transmitted throughso-called http proxies. Ask your administrator which proxy adress and portyou have to to set up if you want to have access for webpages in theinternet.

The phone client detects automatically proxies and supports also automaticconfiguration scripts (*.PAC files). Before you try to configure manuallycheck if you have Internet access with your Internet Explorer.

5 Proxy addressIf needed, enter the IP oder Domain address of your HTTP proxy

6 Proxy PortPort number of the HTTP proxy

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Voice box

Nummer Beschreibung 1 Announcement "System default"

The integrated voice box plays Announcements for each SIP account. TheAnnouncement includes a "Welcome message" and a "Conclusionmessage".

The phone comes with a standard Anouncement, available in German,English and French. The "System default" message cannot be deleted.

2 Own AnnouncementsThe announcement wizard enables you to create recorded announcementsthat are played back by the voice box of your softclient ("test"). OwnAnnoucements can be deleted at any time. Then the "System default"announcement will be used.

3 New Announcement Follow the notes during the intuitive recording wizard.

4 Edit an Announcement

5 Delete an Announcement

6 Start / Stopp - Announcement 1 (Knopf)By clicking "Start/Stop - Announcement 1" you can hear the selectedannouncement. The Announcement will played on the sound device whichis configures for Ringtones.

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7 Start / Stopp - Announcement 2 (Knopf)By clicking "Start/Stop - Announcement 2" you can hear the selectedannouncement. The Announcement will played on the sound device whichis configures for Ringtones.

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License / VersionLicense / Version

Nummer Beschreibung 1 License information

Common license information of your license like customer number, licensenumber etc.

2 Import licenseClick at the button to open a common dialog for selecting the sent licensekey file. Click "Open" to finalize the license import. The phone restartsincluding all new licensed functions.

3 Export licenseClick "Export license" make a license backup including all configured SIPaccounts and other phone settings.

4 Buy licenseOpens an online web shop page where you can order a telephone license

5 Version informationIncludes Version and Build number od the installed telephone

6 File versionFile version of single telephone files for supporting purposes

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Advanced settings

No. Description 1 Popup on incoming calls

If checkbox is set to the checked state, the phone will be placed on top ofall windows when a call comes in.

2 Popup on outgoing callsThe phone pops up with an outgoing call.

3 Minimize on hangupIf checkbox is set to the checked state, the phone will automatically beplaced in the background.

4 Create local FTMF tonesIf activated, DTMF tones are audible. (DTMF stands for Dual-tone multi-frequency signaling). The auible output means not that the signals aresend to the remote party. Here No. 5 must be activated too.

5 Send DTMFWhen clicking at the buttons 0, 1, .. 9, # and *, DTMF tones will betransmitted to the remote party if thisfunction has been activated by settingthe checkbox to the checked state

6 Launch on bootStarts the softphone when the operating system boots.

7 Generate Call Waiting signal The softphone generates a “call waiting” signal if checkbox is in the

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checked state. If checkbox is not in checked state, the caller will receive a"busy" signal if you're already communicating with the softclient.

8 Only numeric charactersThe phone dials only numeric number even if alphabetic characters areincluded.

Example:

"David0044123456789" --> the phone dials "0044123456789"

9 Remember "Auto Answer"The last state of the „Auto answer“ setting is saved and will be restoredon the next launch of the softphone.

10 Remember "Auto Conference"The last state of the “Auto conference” setting is saved and will berestored on the next launch of the softphone.

11 Remember Don Not Disturb (DND)The last state of the “Do not disturb” setting is saved and will berestored on the next launch of the softphone.

12 Remember MuteThe last state of the “Mute” setting is saved and will be restored on thenext launch of the softphone.

13 Dot not terminate during a callWhile in a telephone conversion the softclient will refuse to terminateexecution.

14 Show alertboxDisplays a popup window on an inbound call with details about the caller. Ifthe softclient does not own input focus, the popup window will not getfocus. During an icoming call you can thus continue to write an e-mail or atext in another application program

15 Ignore SIP-Suffix ('@hostname.com')Please deactivate this function in case you might receive SIP URIs thatinclude the server name (e.g. for TAPI calls or protocol handlers) or in caseyou want to make peer-to-peer calls.

16 Send "Busy" on "Ignore"Please activate this function in case the caller shall get a busy signal if youpress the "Ignore" icon.

17 Send "Busy" on "DND"Please activate this function in case you might send a "Busy" signal if youhave 'DND' activated.

18 Dial entered phone number after...Activate this function if you want that the phone dials the entered phonenumber automatically if there is no further activity in the display after Xseconds.

19 Diel entered phone number after...Enter time in seconds when the phone shall start dialing automatically theentered phone number.

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Desktop Sharing

No. Description 1 Activate Desktop Sharing

"Yes" activates Desktop Sharing, which allows you to share presentationswith the remote party directly from within your call. Desktop Sharing is aswell used frequently for remote maintenance. If the remote party supports Desktop Sharing, you can see the Desktop Sharing Icon on your call line.The icon depicts two computers located next to each other. In order tolaunch a Desktop Sharing session, simply click at the symbol. The remoteparty is subsequently and directly invited to a new Desktop Sharingsession.

Desktop Sharing sessions can as well be initiated manually. This might beneeded if the remote party does not have a telephone with DesktopSharing support at his disposal. Desktop Sharing sessions can belaunched via the context menu of the telephone. In this case does theremote party simply use the Desktop Sharing Client Module (FastViewer).

2 Desktop Sharing Licence (FastViewer)Here your valid Desktop Sharing Licence is entered. The Desktop SharingModule runs by default for test purposes in Demo Mode for up to 5 minutes(currently). The Desktop Sharing module requires a separate licence.Please contact us or your supplier.

3 Selection of Desktop Sharing Profile (FastViewer)The setting determines the fastViewer profile name. Profile names areself-explanatory.

4 Timeout for Desktop Sharing Sessions (seconds)

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The default value is 300 seconds. This is the time during which the remoteparty must react on a pending invitation. If the telephone does not read anacknowleding message to the invitation, the session is closed again.

ConfigurationSupported Devices

SNOM 300,320,360,370,820,870AAstra 57i

Die functionality depends on the hardware device of the manufacturer!

Switching between softwarephone and desktop phone mode

By default, the phone starts in the softphone mode. Click the button "HARD PHONE" to switch to the desktopphone mode (snom AASTRA). Click the button "SOFT PHONE" to get back into the the Softphone mode.

CTI Configuration (only available in special versions)

The CTI mode allows the control of the phone devices SNOM 300,320,360,370,820,870 andAASTRA 57i by using a graphical user interface. The functionality and the device status arehighly device-dependent. The symbolism of the graphical user interface is based on thesoftphone mode to provide you with a familiar environment.

The CTI mode supports TAPI. So table devices can be connected to the TAPI TSP driver from anyTAPI supporting CRM application (Address Managment, etc.). Details can be found at the chapter"TAPI".

snom firmware versionAll snom desktop phones ex firmware version 7.3.10a are supported. If not set by default pleaseactivate the function "CSTA". The feature is found in the snom Administration menu"SETUP"->"ADVANCED"-> tab "QoS/Security ". Activate the function "Allow CSTA control" in the

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TAB sheet "QoS/SECURITY".

AASTRA 57i firmware versionNo special firmware is needed.

Configuration "Device settings"

Enter the IP address of your desktop phone in the field "IP address". Possibly an authenticationis required for the phone control. In that case please check "Authentication Required" and enterthe device username and password into the fields "Username" and "password".

Configuration "Local settings"Stand alone phones send data via http to the CTI interface in order to transmit controlinformation. To enable the communication you need to configure an interface on the localmachine based on the machines local IP address and port number. Maybe you have to allow theusage of this Port in your Firewall.

Usually the local IP address is already present in the box labeled 'Local IP' and you can keep thesettings. If you are using a PC with more that one network card or if you are using multiple IPaddresses you might need to select the appropriate IP address. Use the arrow to select form thelist of available IP addresses.

The box labeled 'Local port' must be set to the local TCP port where the communication with thephone takes place. Please make sure to update your firewall settings in order to enable thecommunication.

Hit 'OK' to start testing the CTI interface with the current settings. On successfull callestablishment you will see the phone number, version number and supported CTI mode of thedevice. The CTI mode can be 'Basic' or 'CSTA' (extended functional mode).

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Dialing rulesOutside line (PBX)

Select your rules for the "outside line" of your PBX.

Function Description Active Activate / Deaktivate the "outside line" dialing rule.

Maximum length (internalcalling number)

Choose the maximum length of your internal telephone numbers. If the callednumber is longer than the value of the maximum length, the phone dials the"outside line" number automatically as a prefix.

Outside line Choose the number which shall be dialed as a Prefix to get an "outside line".

Drop outside-line withincoming call

Drops the outside-line from the regular dialed phone number, e.g.:0-069-123456789 will be changed to 069-123456789

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ProfileProfiles

You can create different profiles for any location.

Function Description New Profile Click on "Create Profile" to enter a new "profile name". Select the new profile and

click "OK" to enter the profile. Now, change your settings for the device. Whenclosing the application all data will be stored in the profile automatically.

Change Profile Select the desired profile and click "OK".

Delete Profile Select the profile which shall be deleted and click the "Delete Profile" button.

Overview

The following explanation is exemplary and explains the ADDIN functionality and handling. Allforms, screenshots and menus are relating to Outlook 2007.

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How to use the Ninja Addin for MS Outlook

The Outlook AddIn installs by default through an installer. Once installed it automatically connects with theNinja software telephone or alternatively with the CTI interface of a hardware device as soon as one suchapplication has started. There is no need for installing or configuring any other protocol or interface like TAPI(Telephony Advanced Programmers Interface). If you want to deactivate/reactivate the Ninja Outlook AddIn in the future, simply proceed to the "Com AddIn"menu item under "Extras" in MS Outlook. There you‘ll find the options "Add" and "Remove" of AddIns.

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Synchronisation of contact data

Upon the first launch of the Ninja Addin for MS Outlook, all contacts are synchronized. Synchronizationprogress is being displayed during the entire period. All future changes are automatically synchronized in orderto eliminate the need for a full synchronization. Synchronization with older versions of MS Outlook including the2003 release may take quite long due to technical reasons. Due to the more favourable software interface ofMS Outlook 2007 has synchronization speed encreased drastically. The reading of 18.000 contacts takes onlyapproximately 6 seconds on an average PC.

Exchange and Business Contact Manager (BCM) Folders

If isn't able to connect to Exchange or Business Contact Manager folders (power outage, server down), setOutlook into the "Offline" Mode. Otherwise Outlook tries to connect automatically, which can potentially lead todelays.

Supported Versions (Operating Systems: 32 Bit & 64 Bit)Outlook Folder Outlook 2010 (32 Bit)

Outlook 2007Outlook 2003Outlook XPOutlook 2002Outlook 2000

Exchange Folder Supported

Business ContactManager (BCM) Folder

Supported

IMPORTANT!

The usability and design can be different using Office 2010 depending on the changedMicrosoft Office program architectur.



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Call Monitor

The following explanation is exemplary and explains the ADDIN functionality and handling. Allforms, screenshots and menus are relating to Outlook 2007.

Function Description Accept incoming call Start of a telephone conversion with the offered call.

Show user interface ofsoftware telephone

The Ninja software telephone is brought to front.

Hide/Show Call Monitor Toggles between hiding/showing the Call Monitor.

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Function Description Usage Usage of the Addin is identical with the use of the standard Ninja user interface.

You can use the functionality as you’re already used to witgh Ninja.

Display of contactinformation

Contact information is automatically filtered through the Outlook database and isbeing displayed for the active line and for all calls – incoming or outbound. Thetelephone history further contains contact infos.

IMPORTANT!



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Localisation (Country Code)Localisation (Country Code)

Please enter for each SIP account the correct "Country Code". The phone checks all existing phone numbersof your contacts. If any of them matches you will get shown the contact Name and Firstname in the Display andin the Call History. The phone number will be shown as Tiptext if you move the Mouse Icon over the phonenumber in the display.

Nummer Beschreibung 1 Country Code

Pleasy enter your country here in order to enable the telephone to correctlydetect contacts on inbound and outbound calls.

2 Dialing rule for national calls

3 Remove country code Only removes the country code that was entered in the edit box „Countrycode“ (1). The remaining digits are called. This selection is only useful orcountries that do not require leading digits for the area code, e.g. "0" inGermany (069).

4 Replace country code with Only replaces the country code that was entered in the edit box „Countrycode“ (1) by the string entered here.

5 Dialing rule for international callsReplaces the leading "+" by "00" for outbound international calls. Thisselection is applicable for VoIP services that strictly require telephonenumbers for international calls to begin with "00".

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6 Replace + or 00 by ...Replaces the leading "+" or "00" for outbound international calls by thestring entered here. This selection is applicable for VoIP services or incountries that require a prefix that differs from the common standard. E.g.it is necessary to make a call from the U.S. to Great Britain (+44) by dialing01144 in lieu of 0044.

IMPORTANT!



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Uninstall Outlook CTI (AddIn)Uninstall Outlook CTI (AddIn)

Click the right mouse button on the Microsoft Outlook Navigation bar and choose the option "Customize"

In the following dialog select ""XPhone Outlook AddIn" and click "DELETE".

Security (SRTP/TLS)Security (SRTP/SSL)

Select "SRTP/TLS" as your wanted encryption method.

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Function Description Activate security Enable the encryption by choosing "Yes". After finishing all settings click "OK" that

the phone reconnect to the SIP server.

TLS Port This Port (TCP) is explicitly needed for secure phone connects. Ask youradministrator or provider to get the right port number.

Certificate Registering with SRTP/TLS to the SIP server is only possible with a validcertificate. You will get such a certificate from your administrator or provider.

Often the certificate files are provided as downloads. You can install the certificatewith a double click onto the file.

Meaning TLS Transport Layer Security (TLS) and its predecessor, Secure Sockets Layer (SSL),are cryptographic protocols that provide security and data integrity forcommunications over the Internet.

Meaning SRTP The Secure Real-time Transport Protocol (or SRTP) defines an encrypted profileof RTP (Real-time Transport Protocol).

Meaning AES The Advanced Encryption Standard (AES) is an encryption standard adopted bythe U.S. government and was announced by National Institute of Standards andTechnology (NIST) since 2000

AES is the first publicly accessible and open cipher approved by the NSA forsecret information.

Symbols of the encryption status

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Funktion Beschreibung Green key Encrypted connection to the SIP server has been established.

Red key Encrypted connection to the SIP ist NOT possible.

Green frame The connection on this line is encrypted.

Yellow frame The status of the line is determined.

Red frame The connection on this line is NOT encrypted.

IMPORTANT!

SRTP without TLS

We only support SRTP in combination with TLS. Only with TLS the secure key exchange can beguaranteed. Regarding to that it's not possible to use only SRTP without TLS.

Open SSL Components

This application uses Open SSL components for the secure data transfer.

The SSL components are under Copyright of "The OpenSSL Project".

Copyright (c) 1998-2008 The OpenSSL Project. All rights reserved.This product includes software developed by the OpenSSL Project for use in the OpenSSL Toolkit.(http://www.openssl.org/)

THIS SOFTWARE IS PROVIDED BY THE OpenSSL PROJECT ``AS IS'' AND ANY EXPRESSED ORIMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OFMERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENTSHALL THE OpenSSL PROJECT OR ITS CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT,INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITEDTO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; ORBUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER INCONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANYWAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCHDAMAGE.

The Open SSL components are including the Original SSLeay license.

Copyright (C) 1995-1998 Eric Young ([email protected])

This product includes cryptographic software written by Eric Young ([email protected])This product includes software written by Tim Hudson ([email protected])

THIS SOFTWARE IS PROVIDED BY ERIC YOUNG ``AS IS'' AND ANY EXPRESS OR IMPLIEDWARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITYAND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE AUTHOROR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, ORCONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTEGOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVERCAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THISSOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.

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Polymorphic "Peer-to-Peer" Encryption (PMC)Polymorphic "Peer-to-Peer" Encryption (PMC)

This enciphering method offers the following decisive advantages over SRTP/TLS:

High end encryption that protects your voice conversations very effectively from attacks by any third party.

Automatic attempt to estblish encrypted session as soon as the remote party picks up a call.

Works with almost any commercial PBX (Public Branch Exchange - e.g. 3CX, Asterisk, nfon, Auerswald,Hipath, etc.) as well as with any standard SIP Telephony Service Provider like sipgate, 1und1, mondotalk,TPad, etc. (The remote party needs to be connected to the same SIP Service Provider or with an affiliate(same telephony network).

Secure and fully automatic Diffie-Hellman Public Key Exchange (4096 bit)(http://en.wikipedia.org/wiki/Diffie%E2%80%93Hellman_key_exchange)

High Encryption using a Polymorphic Cipher(http://www.turbocrypt.com/eng/content/Backround-Info/Basic-Idea.html)

Visual check of the Short Authentication String in order to prevent possible attacks (Man-in-the-MiddleAttack) very effectively. If the hash (Short Authentication String) that is displayed by your phone differesfrom the one that your remote party reads alound, simply hang up and try to again.

Activation of "Peer-to-Peer" Encryption

Please select "Peer to Peer" as encryption method.

Then please select "Yes" at "Activate encryption".

Function Description

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Verschlüsselung aktiviert Activate encryption and subsequently click at "OK" in order to let the telephoneregister again with the SIP (telephony) server.

Automatischverschlüsseln

At the beginning of each and every call the telephone tries to establish encryption.

AES Advanced Encryption Standard (also known as Rijndael algorithm.

http://en.wikipedia.org/wiki/Advanced_Encryption_Standard

PMC Polymorphic Elcryption Algorithm. This cipher features longer keys in contrast toAES and the actual cipher depends on the key.

http://www.turbocrypt.com/eng/content/Backround-Info/Basic-Idea.html

IMPORTANT INFORMATION!

Checksum (Short Authentication String):

Please read this checksum to your conversational partner and make sure that the values match on bothsides. If these values are identical on both sides, the line is secure. If although the values differ, then a"Man-in-the Middle" attack has been mounted on your specific conversation! Please terminate the call andtry to establish another call. Please repeat this check!

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Index

- -

- A -AAstra 58activate answering maschine 30Advanced settings 54AEC 44Alertbox 54announcement scheme 30Announcement wizard 50Announcements wizard 50answering maschine 50, 30application folder 20Audio Codecs 39audio file 20Audio Recorder 20audio volume 41Auth User 27Auto Answer 54auto conference 54Available 32

- B -ban 19Ban phone numbers 19banned 19banned list 15Basic 58Busy Lamp Field (BLF) 34

- C -call history 15call history 15Call Monitor 64call signal 39call waiting 54called 15callto 24cam 22CIF 22Codecs 39configuration 58contacts 11, 36, 15context menu 10country code 36Create DTMF tones locally 54CSTA 58CTI Interface 58CTI Mode 58

- D -Default announcements 30Desktop Sharing 57Destination number 30, 11Dialing rules 38Display name 27DND 54Domain 39Domain 27

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DSL 22DTMF 54

- E -Export License data 52

- F -Firewalls 45Forward 30FPS 22Frames per second 22functional survey 5, 3

- G -Globale Shortcuts 24

- H -H263 22hang up 24

- I -Input- and Output devices 41IP 58IP-Adress 58

- K -Knocking-Menu | Handling 3

- L -LAN 45, 22Languages 30License information 52License key 52licenses 52Line monitor 7, 3local net 39Localisation 36Localisation 66

- M -Main navigation panel 3Management 25matching 36messages 15microphone 41minimize 54Minimize on hangup 54missed 15mute 54MWI 15, 17MWI (Message Waiting Indicator) 17

- N -Network Settings 45not reached 15

- O -Offline 32On the phone 32Outbound Proxy 27Outbound Proxy Port 27Outside line 38

Seite76


Outside line 38Overview 61

- P -Password 27PAUSE 24Peer-to-Peer 27phone number 15Pictures 22Play 20Polymorphe Peer to Peer Encryption (PMC) 72Popup on incoming call 54Presence 32Presence 32Presence Service 32Profiles 61Protocol handler 24Proxy 27Proxy Name 27Proxy Server 27

- Q -Quality 22

- R -Realm 27received 15Recording 50recording 20Recording 20recording 20remeber auto conference 54remember Auto Answer 54remember DND 54remember mute 54Resolution 22resolution 22Ringtone device 50RTP 45RTP Port 45

- S -send DTMF 54Send Video automaticaly 22Shortcuts 24Simple SIP Channel 48Simple SIP Channel (SSC) 48SIP 45, 27, 24SIP Account 32SIP Port 45, 27SIP Softphone 3sip2 24SNOM 58speaker 41STUN 45STUN Server 45System informationen 52System settings 41System settings 41

- T -table phone 58TCP 58tel 24

Seite77


- U -Uninstall Outlook CTI 68Use local answering maschine 30Use local IP 39Use online answering maschine 30User name 27

- V -Video 22Video (Settings) 22Video data 22Video device 22Video telephony 22voice box messages 17voicebox recorder wizard 50Volume 41

- W -Work areas 3

© Global IP Telecommunications Ltd, 2005 - 2013w w w .globaliptel.com

Documents

JUGGLER - downloads.globaliptel.comdownloads.globaliptel.com/nam/Phone-Encryption-JugglerVideoPhone... · JUGGLER SIP Phone Version 5.x Global IP Telecommunications, Ltd. behält