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Help page for Blink Pro is available here
Help page for Blink QT is available here
New releases are announced on
To report a problem, please use Blink Support Forum.
If you encounter connectivity problems (e.g. no incoming call), you can find the reason by going to Menu Window -> Logs ->SIP. If you need help for reading these logs please copy and paste them to the support forum.
If you experience a software crash, please send the complete crash report available in the crash reporter window (just copy and paste from it). You can find the previous crash reports saved in ~/Library/Logs/DiagnosticReports/ on MacOSX 10.6 or ~/Library/Logs/CrashReporter/ on MacOSX 10.5.
Non-critical programming errors are logged in the system log file /var/log/system.log. You can isolate the entries by opening Applications -> Utilities -> Console, selecting LOG FILES -> system.log and typing Blink in the search bar. Please provide us with this information when you report the problem you encountered.
Blink accounts and settings are stored in a flat-text configuration file. The location of the configuration file is ~/Library/Application Support/Blink/config. Do not edit this file manually. If you did and Blink does not start you must delete the file and start all over again.
If you send the configuration file to us for debugging, use a text editor to strip the passwords.
Please report your OS name and version and Blink version. You can find Blink version in menu Blink -> About.
To use Blink you need a SIP Account. At first start, Blink shows the Add Account window that helps you configure the initial SIP Account.
If you do not have a SIP Account, Blink can create one for you using the free service provided by SIP2SIP. For more information about the features of the SIP service go to http://sip2sip.info
If you already have a SIP Account, select this option then enter the SIP address in the form of user@domain and password. The SIP registrar of the domain is then automatically located. If the SIP provider does not have proper DNS records set for his domain, you can set a particular SIP Outbound Proxy in menu Preferences in the Advanced section of the SIP Account.
You may configure Blink to use multiple SIP Accounts. To configure a new SIP Account, go to menu Preferences->Accounts and click the + button. The Add Account window shows up and you can repeat the steps described above.
After you have configured an account, you can see the status of the registration in the account list of Main Window window. Solid color means that the account has successfully registered, gray color means that the registration has failed or is in progress. You can also see the status of the registration in the account list of Preferences window. Green color means that the account has successfully registered, Red color the registration has failed and Yellow means that he registration process is in progress.
Blink presents a Contacts driven interface as the main window. The design allows to locate a Contact then to start an action for it, for example to start an Audio Call.
The main interface can be collapsed into a compact mode that hides the Contact List. Click on the + Green button of the window to collapse and expand the Contacts List.
Hover the mouse over the GUI elements to see a short description of their function.
You can select the SIP Account used for outbound sessions in the Accounts drop down box presented on the top of the main interface.
Bonjour Account is a special type of account that is designed to announce itself and discover other neighbours on the local area network. This account does not require a server or SIP service in order to operate.
The Search bar is used for finding an existing contact or for entering a SIP address or telephone number.
To search for a Contact in Blink's Contacts List and the system Address Book, fill in the name in the Search Bar. You may select a Contact that matched the types text or use the typed text as the Contact for the next action.
Use the buttons presented at the bottom of the main window to start a session to the current selected Contact or press on Add Contact to add the typed address to the Contacts List.
Pressing enter after entering text in the search bar will start an audio call to that address.
If you have registered a SIP Account with Blink, you can search for SIP addresses of other Blink users in menu Tools -> Search for people....
Contacts can be organized in Groups. The Groups support ordering and can be collapsed or expanded when clicking on their left-side triangle.
Right click on a Contact to see all possible actions for each of them.
The group named Address Book cannot be deleted, it is dynamically populated with entries from the system Address Book. To reload changes made in the Address Book collapse and expand the group again.
For Address Book entries, to have a SIP Address recognized by Blink you must have entries in either Email or URL sections that start with the sip: prefix.
Bonjour Neighbours appear in the Contacts List as a separate Group. When selecting Bonjour as the default account, Bonjour Neighbours group is expanded and moved on top of the Contacts list. When changing back to a regular SIP Account, the groups in the Contacts list reshuffle to their previous positions.
Chat sessions are automatically accepted from Bonjour Neighbours.
Audio sessions can be automatically accepted based on the correspondent setting from Account menu.
You can edit a Contact by right clicking on it and select Edit menu option. Edit Contact window appears. To set the Icon for the Contact, click on the Icon area, a file selection dialog appears.
You can add a Contact to the Contacts List by clicking on the + button presented on the bottom left of the main interface. Add Contact window appears.
Use the buttons presented at the bottom of the main window to start a session to the current selected Contact.
Clicking on Return key or double-clicking a Contact will start the default session type depending on the preferred media configured for each Contact. The default action is to start an Audio Call.
Incoming session requests bring up the Alert Panel. Click on the Accept Button to accept a session request. There are two options for rejecting an incoming session request:
When the Answering Machine is enabled, a countdown appears in the Alert Panel. While the countdown counts you can still disable the Answering Machine from the Account menu.
The currently selected input and output audio devices are showed for incoming Audio Call requests.
Audio Calls are displayed in the lateral drawer attached to the main interface.
Blink has no traditional telephone dialpad, nor you will need one to interact with it. As computers have physical keyboards displaying a classic 12 key dial-pad is a poor design choice for a modern graphical user interface of a rich communications SIP client.
To start an Audio Call, select an existing contact or enter a SIP address or telephone number in the search bar.
Once text is entered in the search bar just press enter to start an audio call to that address.
Once a Contact is selected, click on the Green Handset located at the bottom of the main interface or right click and select Start Audio Session from the contextual menu. For each new call an entry is added to the Audio Calls drawer.
To switch between multiple calls just click on another call in the drawer. When switching to a particular session, the audio stream is connected to the input and output device and all other Audio Calls are put on hold.
To start Instant Messaging next to an existing Audio Call, right click on the session displayed in the Audio Drawer and select an Add Chat
To mute your microphone click on the bottom right microphone icon. To un-mute click it again.
Dial tones are used to interact with PBX and IVR systems or legacy PSTN gateways. To send DTMF tones, focus the Audio Call by clicking on it. The current call is already focused if you just started it. What you then type on the keyboard is automatically translated into DTMF tones. Alpha-numeric keys are translated into numeric DTMF keys as they are on a numeric telephone keypad.
To hangup click on the Red handset displayed for each Audio Session in the right hand side of the drawer or press the Escape key.
The shortcut key for hold and unhold of the selected session is Space.
To record click on the Black circle. While recording, the circle will toggle between red and black. Recorded sessions can be found in menu Audio -> Recordings.
History of previous sessions is available in the History menu. If you have registered a SIP Account with Blink, you can review the missed calls while Blink was off-line in Account-> Settings on SIP Server -> History menu.
To redial the last session type Command-R, to redial a previous session select it in the History menu.
iTunes automatically stops playing before any Audio Call starts and resumes playing after all Audio Calls have finished.
Audio Calls have status indication about the progress of establishment of the RTP stream and ICE negotiation, the negotiated codec and the presence of encryption for signaling and media.
The RTP end-points and negotiated ICE candidates (when ICE negotiation succeeded) are displayed when hovering the mouse over the Audio codec information area.
To dial a telephone number, just enter the number in the Search Bar followed by Return key.
When dialing numbers by selecting entries from the Address Book it is recommended that you store your numbers in international E.164 format (that is + sign followed by the country code then the subscriber number).
You can set in Advanced -> PSTN preferences how to replace the + sign with a prefix (the IDD prefix setting) recognized by your SIP service provider or PBX. For example if you set 001 as IDD prefix, when you dial +44XXX the number will be dialed as 00144XXX.
Optionally, telephone numbers can also be prefixed before dialed out, for example if a 9 is required by your PBX (the Prefix setting).
If you have created a SIP Account with Blink, you can call to PSTN numbers if you must have a positive credit. You can use a Credit Card to add Credit to your SIP Account in menu Tools -> Buy prepaid credit.... You may request the assignment of a PSTN Caller Id by sending an request to the support email address. Recognized number formats are + or 00 followed by the Country Code and then Subscriber Number.
If you use another SIP Account than the one provided by Blink, access to the PSTN is subject to the support provided by your SIP service provider. The number format depends as well on the conventions imposed by the SIP service provider in question.
Voicemail is a feature provided for SIP accounts created by Blink. It is provided by the network while Blink is offline, when the call forwarding to Voicemail is activated. To change the settings of your SIP2SIP voicemail box dial 1233. You can record your unavailable message and listen to your messages stored on the server. The voicemail delivered method (by email attachement or by dialing into the server) is set in menu Tools -> Settings on SIP server in the Voicemail section.
When the Answering Machine is enabled, Audio Call requests will be automatically answered with either the standard or the custom recorded message after the delay configured set in Preferences -> Settings -> Answering Machine setting.
You can enable the Answering Machine using the menu item Tools -> Enable Answering Machine. The incoming Audio Calls are then recorded and can be listened to later in Tools -> Recordings menu.
While the Answering Machine is handling an Audio Call, you can hear what the remote party is speaking while your microphone is muted. If you wish you can take over by clicking on the Green Handset icon in the Audio drawer. The recording stops and the session is connected to the microphone.
Drag and drop Audio Calls on top of each other to create a Conference or use the button presented at the botton at the Audio Drawer to start a Conference with all active Audio Calls.
Add new participants to a Conference by dragging Contacts from the Contacts list onto an Audio Call.
Each participant can be muted individually by pressing on the microphone symbol that appears on each conferenced session.
To end an Audio Conference drag the session out of the conference area or press the Conference button again.
There can be only one active Audio Conference at any given time.
Chat Sessions are displayed into the Chat window. For each recipient there is a new Tab created at the bottom of the window. The Tabs are matched based on the SIP Address of the recipient and the Chat Aliases configured for the Contacts.
The lateral drawer display the active participants present in the chat session or multi-party conference.
Chat Sessions are encrypted using TLS and handled like the Audio Sessions, a Chat Session is established only to the device where the user accepts the session. Chat sessions can carry arbitrarily large amounts of text information.
You can start a Chat Session by right-clicking on a Contact and chose Start Chat Session.
Messages typed before the session is accepted by the remote party are queued and delivered once the session has been established. If the session fails to establish, the undelivered messages will be resent at the next successful attempt.
Message marked with a red color background have failed to be delivered.
To an established Chat Session you can add Audio or start a Desktop Sharing session by clicking on the buttons presented on the toolbar.
To review previous chat conversations go to menu History -> Chat Conversations.
To insert an empty line use Shift-Enter combination.
The rendering of Emoticons can be disabled in each chat window, this can be useful to transfer programming code snippets without interpreting the special characters as emoticons.
Short Messages is the way to interact with legacy SIP IM clients that do not support session based Chat Sessions using MSRP protocol.
Short Messages end-up on all SIP devices registered for the remote SIP Account, including hardware phones that accept such messages but do not display them, the recipient has no control where to receive them. Short Messages have no end-to-end delivery report and cannot be encrypted the way Chat Sessions are.
A Short Message cannot exceed 1300 characters on IP networks and 160 on PSTN mobile networks.
You can send a Short Message by right-clicking on a Contact and chose Send SMS.
Previous exchanged messages can be reviewed in the menu Window -> Chat History.
Outgoing messages are by default synchronized with all Blink instances registered under the same SIP Account. To disable this functionality, go to Preferences -> Settings -> Chat and un-check the SMS replication checkbox.
To send a file you can use drag and drop from Finder or other application like iPhoto onto a Contact, to a Chat Window or by right clicking on a Contact and then selecting Send File... option.
File Transfers are displayed in a separate window. Each file has a progress bar attached with estimated transfer speed and remaining time.
Desktop Sharing is implemented using VNC protocol over an MSRP connection.
The standard Screen Sharing client provided by MacOSX is used while on the server side Vine VNC server built-in Blink is used. During the Desktop Sharing session, the connection window displaying Connecting to localhost remains visible but can be safely ignored.
The party sharing the desktop must have a solid broadband connection, the bandwidth required for a good experience is minimum 500 Kbit/s.
Make sure MSRP and Notifications debugging are disabled during Desktop Sharing, these features overload the CPU.
For Audio Calls you can enable the use of ICE negotiation in Advanced -> NAT Traversal -> Use ICE. ICE requires the presence of STUN and TURN servers, they must be provided by the SIP service provider. The ICE negotiation details are displayed in RTP Logs window. While the ICE negotiation progresses, its status is also displayed in the Audio Call information, for example the gathering of ICE the candidates and the probing of the remote candidates.
Once the audio stream is established, the audio information area is updated with the negotiated codec type and the sample rate. The active RTP end-points and the ICE candidates types are displayed while hovering the mouse over the audio status information area.
To traverse a NAT for MSRP related sessions (Instant Messaging, File Transfer and Desktop Sharing) a MSRP relay configured for the called party domain is needed. The SIP service provider that provides the SIP Account must support this feature. If you have direct IP connectivity and no MSRP relay available, your client needs a TLS certificate loaded otherwise the media stream will fail.
The TLS certificate in PEM format and its unencrypted private key must be saved in the same file, the file must be specified in the Advanced -> TLS section of the SIP Account. If you are not familiar with the generation of a TLS certificate, Go to menu Preferences -> Accounts, click on the + sign, select Create a SIP Account in the Add Account window. Blink will then create a new SIP Account under sip2sip.info domain and a TLS certificate will be created. You can re-use that certificate for other SIP Accounts.
Growl notifications are raised for missed sessions and the Dock Icon badge is updated with the number of missed sessions.
When sessions are accepted elsewhere and the SIP server supports the Reason header, the sessions will be logged in the incoming section rather than the missed section.
Logs window helps troubleshoot network traffic for various protocols used by Blink. For example, one can follow in the self scrolling interface the time sequence of all DNS lookups and SIP messages exchanged with the other end-points.
To see what Blink is doing under the hood go to the menu Window -> Logs. It is advisable not to leave MSRP Trace and Notifications Logs turned on unless you really need them for debugging purposes, they are very demanding in terms of CPU load during File Transfer and Desktop Sharing sessions.
Blink is integrated with the SIP server functionality. One can change settings directly on the SIP server like Call Forwarding, Do Not Disturb, list of last missed sessions and other settings.
The settings are displayed in Server Settings window. The window is actually a web interface that loads the URL configured in the Advanced -> Server -> Settings URL for each SIP Account. The remote web server can support digest authentication, Blink sends the SIP Account credentials when prompted by the server.
Blink auto-detects audio devices there are plugged in and out on the fly and prompts the user to switch to a new added device. To selected a particular audio device go to menu Audio.
The best experience is obtained by using a wired headset that introduces no additional processing or possibility of acoustic echo.
Echo may appear when using Blink in Speaker Phone mode (no headset attached). This manifests itself for the remote party when his/her voice heard in the speaker is finding its way back into the microphone.
The built-in Acoustic Echo Cancellation works to some degree when using the computer built-in speaker and microphone. You can improve the performance of the built-in Echo Cancellation by using a speaker that is further away from the microphone. For instance on certain laptops the speaker and microphone are located too close to each other and the echo is not cancelled out. You can try placing an external speaker for solving this particular situation.
To avoid the acoustic echo completely, you must use a headset or purchase a dedicated hardware audio device specialized in the Echo Cancellation function.
To change Blink configuration go to menu Preferences. The Preferences window appears.
Each SIP Account has Advanced options available, click on the small triangle presented under the SIP Address to access them.
Blink preferences are stored in ~/Library/Application Support/Blink folder. You can reset Blink by deleting this folder.
Blink SIP accounts and settings are stored in ~/Library/Application Support/Blink/config file.
GUI preferences are stored in ~/Library/Preferences/com.agprojects.Blink.plist file. (e.g. size, location of the windows and other MacOSX related settings).
If you ever want to move your Blink configuration to another login account you must edit the content of your config file and replace the absolute file paths accordingly. You can edit this file with any text editor as long as you do not break the return lines and formatting of the file.