AddressCardForm Twinkle - Address Card &Remark: Infix name of contact. First name of contact. &First name: You may place any remark about the contact here. &Phone: &Infix name: Phone number or SIP address of contact. Last name of contact. &Last name: &OK Alt+O &Cancel Alt+C You must fill in a name. You must fill in a phone number or SIP address. AddressTableModel Name Phone Remark AuthenticationForm Twinkle - Authentication user No need to translate The user for which authentication is requested. profile No need to translate The user profile of the user for which authentication is requested. User profile: User: &Password: Your password for authentication. Your SIP authentication name. Quite often this is the same as your SIP user name. It can be a different name though. &User name: &OK &Cancel Login required for realm: realm No need to translate The realm for which you need to authenticate. BuddyForm Twinkle - Buddy Address book Select an address from the address book. &Phone: Name of your buddy. &Show availability Alt+S Check this option if you want to see the availability of your buddy. This will only work if your provider offers a presence agent. &Name: SIP address your buddy. &OK Alt+O &Cancel Alt+C You must fill in a name. Invalid phone. Failed to save buddy list: %1 BuddyList Availability unknown offline online request failed request rejected not published failed to publish Click right to add a buddy. CoreAudio Failed to open sound card Failed to create a UDP socket (RTP) on port %1 Failed to create audio receiver thread. Failed to create audio transmitter thread. CoreCallHistory local user remote user failure unknown in out DeregisterForm Twinkle - Deregister deregister all devices &OK &Cancel DiamondcardProfileForm Fill in your account ID. Fill in your PIN code. A user profile with name %1 already exists. DtmfForm Twinkle - DTMF Keypad 2 3 Over decadic A. Normally not needed. 4 5 6 Over decadic B. Normally not needed. 7 8 9 Over decadic C. Normally not needed. Star (*) 0 Pound (#) Over decadic D. Normally not needed. 1 &Close Alt+C GUI Failed to create a %1 socket (SIP) on port %2 Override lock file and start anyway? The following profiles are both for user %1 You can only run multiple profiles for different users. If these are users for different domains, then enable the following option in your user profile (SIP protocol) Use domain name to create a unique contact header Cannot find a network interface. Twinkle will use 127.0.0.1 as the local IP address. When you connect to the network you have to restart Twinkle to use the correct IP address. Line %1: incoming call for %2 Call transferred by %1 Line %1: far end cancelled call. Line %1: far end released call. Line %1: SDP answer from far end not supported. Line %1: SDP answer from far end missing. Line %1: Unsupported content type in answer from far end. Line %1: no ACK received, call will be terminated. Line %1: no PRACK received, call will be terminated. Line %1: PRACK failed. Line %1: failed to cancel call. Line %1: far end answered call. Line %1: call failed. The call can be redirected to: Line %1: call released. Line %1: call established. Response on terminal capability request: %1 %2 Terminal capabilities of %1 Accepted body types: unknown Accepted encodings: Accepted languages: Allowed requests: Supported extensions: none End point type: Line %1: call retrieve failed. %1, registration failed: %2 %3 %1, registration succeeded (expires = %2 seconds) %1, registration failed: STUN failure %1, de-registration succeeded: %2 %3 %1, de-registration failed: %2 %3 %1, fetching registrations failed: %2 %3 : you are not registered : you have the following registrations : fetching registrations... Line %1: redirecting request to Redirecting request to: %1 Line %1: DTMF detected: invalid DTMF telephone event (%1) Line %1: send DTMF %2 Line %1: far end does not support DTMF telephone events. Line %1: received notification. Event: %1 State: %1 Reason: %1 Progress: %1 %2 Line %1: call transfer failed. Line %1: call successfully transferred. Line %1: call transfer still in progress. No further notifications will be received. Line %1: transferring call to %2 Transfer requested by %1 Line %1: Call transfer failed. Retrieving original call. %1, STUN request failed: %2 %3 %1, STUN request failed. Redirecting call User profile: User: Do you allow the call to be redirected to the following destination? If you don't want to be asked this anymore, then you must change the settings in the SIP protocol section of the user profile. Redirecting request Do you allow the %1 request to be redirected to the following destination? Transferring call Request to transfer call received from: Request to transfer call received. Do you allow the call to be transferred to the following destination? Info: Warning: Critical: Firewall / NAT discovery... Abort Line %1 Click the padlock to confirm a correct SAS. The remote user on line %1 disabled the encryption. Line %1: SAS confirmed. Line %1: SAS confirmation reset. %1, voice mail status failure. %1, voice mail status rejected. %1, voice mailbox does not exist. %1, voice mail status terminated. Accepted by network Line %1: call rejected. Line %1: call redirected. Failed to start conference. Failed to save message attachment: %1 Transferred by: %1 Cannot open web browser: %1 Configure your web browser in the system settings. GetAddressForm Twinkle - Select address &KAddressBook This list of addresses is taken from KAddressBook. Contacts for which you did not provide a phone number are not shown here. To add, delete or modify address information you have to use KAddressBook. &Show only SIP addresses Alt+S Check this option when you only want to see contacts with SIP addresses, i.e. starting with "sip:". &Reload Alt+R Reload the list of addresses from KAddressbook. &Local address book Contacts in the local address book of Twinkle. &Add Alt+A Add a new contact to the local address book. &Delete Alt+D Delete a contact from the local address book. &Edit Alt+E Edit a contact from the local address book. &OK Alt+O &Cancel Alt+C You seem not to have any contacts with a phone number in KAddressBook, KDE's address book application. Twinkle retrieves all contacts with a phone number from KAddressBook. To manage your contacts you have to use KAddressBook.As an alternative you may use Twinkle's local address book. Are you sure you want to delete contact '%1' from the local address book? Delete contact GetProfileNameForm Twinkle - Profile name &OK &Cancel Enter a name for your profile: The name of your profile A profile contains your user settings, e.g. your user name and password. You have to give each profile a name. If you have multiple SIP accounts, you can create multiple profiles. When you startup Twinkle it will show you the list of profile names from which you can select the profile you want to run. To remember your profiles easily you could use your SIP user name as a profile name, e.g. example@example.com Cannot find .twinkle directory in your home directory. Profile already exists. Rename profile '%1' to: HistoryForm Twinkle - Call History Time In/Out From/To Subject Status Call details Details of the selected call record. View &Incoming calls Alt+I Check this option to show incoming calls. &Outgoing calls Alt+O Check this option to show outgoing calls. &Answered calls Alt+A Check this option to show answered calls. &Missed calls Alt+M Check this option to show missed calls. Current &user profiles only Alt+U Check this option to show only calls associated with this user profile. C&lear Alt+L Clear the complete call history. Note: this will clear all records, also records not shown depending on the checked view options. Clo&se Alt+S Close this window. &Call Alt+C Call selected address. Call... Delete Call start: Call answer: Call end: Call duration: Direction: From: To: Reply to: Referred by: Subject: Released by: Status: Far end device: User profile: conversation Re: Number of calls: ### Total call duration: IncomingCallPopup %1 calling InviteForm Twinkle - Call &To: Optionally you can provide a subject here. This might be shown to the callee. Address book Select an address from the address book. The address that you want to call. This can be a full SIP address like sip:example@example.com or just the user part or telephone number of the full address. When you do not specify a full address, then Twinkle will complete the address by using the domain value of your user profile. The user that will make the call. &Subject: &From: &Hide identity Alt+H With this option you request your SIP provider to hide your identity from the called party. This will only hide your identity, e.g. your SIP address, telephone number. It does not hide your IP address. Warning: not all providers support identity hiding. &OK &Cancel Not all SIP providers support identity hiding. Make sure your SIP provider supports it if you really need it. F10 LogViewForm Twinkle - Log Contents of the current log file (~/.twinkle/twinkle.log) &Close Alt+C C&lear Alt+L Clear the log window. This does not clear the log file itself. MessageForm Twinkle - Instant message &To: The user that will send the message. The address of the user that you want to send a message. This can be a full SIP address like sip:example@example.com or just the user part or telephone number of the full address. When you do not specify a full address, then Twinkle will complete the address by using the domain value of your user profile. Address book Select an address from the address book. &User profile: Conversation Type your message here and then press "send" to send it. &Send Alt+S Send the message. Delivery failure Delivery notification Send file... Send file image size is scaled down in preview Open with %1... Open with... Save attachment as... File already exists. Do you want to overwrite this file? Failed to save attachment. %1 is typing a message. F10 Size MessageFormView sending message MphoneForm Twinkle Buddy list You can create a separate buddy list for each user profile. You can only see availability of your buddies and publish your own availability if your provider offers a presence server. &Call: Label in front of combobox to enter address The address that you want to call. This can be a full SIP address like sip:example@example.com or just the user part or telephone number of the full address. When you do not specify a full address, then Twinkle will complete the address by using the domain value of your user profile. Address book Select an address from the address book. Dial Dial the address. &User: The user that will make the call. Auto answer indication. Call redirect indication. Do not disturb indication. Message waiting indication. Missed call indication. Registration status. Display Line status Line &1: Alt+1 Click to switch to line 1. From: To: Subject: idle No need to translate Transferring call sas No need to translate Short authentication string g711a/g711a No need to translate Audio codec 0:00:00 Call duration sip:from No need to translate sip:to No need to translate subject No need to translate photo No need to translate Line &2: Alt+2 Click to switch to line 2. &File &Edit C&all Activate line &Message &Registration &Services &View &Help Quit &Quit Ctrl+Q About Twinkle &About Twinkle Call someone F5 Answer incoming call F6 Release call Esc Reject incoming call F8 Put a call on hold, or retrieve a held call Redirect incoming call without answering Open keypad to enter digits for voice menu's Register &Register Deregister &Deregister Deregister this device Show registrations &Show registrations Terminal capabilities Request terminal capabilities from someone Do not disturb &Do not disturb Call redirection Call &redirection... Repeat last call F12 About Qt About &Qt User profile &User profile... Join two calls in a 3-way conference Mute a call Transfer call System settings &System settings... Deregister all Deregister &all Deregister all your registered devices Auto answer &Auto answer Log &Log... Call history Call &history... F9 Change user ... &Change user ... Activate or de-activate users What's This? What's &This? Shift+F1 Line 1 Line 2 &Display Voice mail &Voice mail Access voice mail F11 Msg Instant &message... Instant message &Buddy list &Call... &Edit... &Delete O&ffline &Online &Change availability &Add buddy... idle dialing attempting call, please wait incoming call establishing call, please wait established established (waiting for media) releasing call, please wait unknown state Voice is encrypted Click to confirm SAS. Click to clear SAS verification. Transfer consultation User: Call: Hide identity Registration status: Registered Failed Not registered Click to show registrations. No users are registered. %1 new, 1 old message %1 new, %2 old messages 1 new message %1 new messages 1 old message %1 old messages Messages waiting No messages Voice mail status: Failure Unknown Click to access voice mail. Do not disturb active for: Redirection active for: Auto answer active for: Click to activate/deactivate Click to activate Do not disturb is not active. Redirection is not active. Auto answer is not active. Click to see call history for details. You have no missed calls. You missed 1 call. You missed %1 calls. Starting user profiles... The following profiles are both for user %1 You can only run multiple profiles for different users. You have changed the SIP UDP port. This setting will only become active when you restart Twinkle. not provisioned You must provision your voice mail address in your user profile, before you can access it. The line is busy. Cannot access voice mail. The voice mail address %1 is an invalid address. Please provision a valid address in your user profile. Failed to save buddy list: %1 F10 Diamondcard Manual &Manual Sign up &Sign up... Recharge... Balance history... Call history... Admin center... Recharge Balance history Admin center Call &Answer Answer &Bye Bye &Reject Reject &Hold Hold R&edirect... Redirect &Dtmf... Dtmf &Terminal capabilities... &Redial Redial &Conference Conf &Mute Mute Trans&fer... Xfer NumberConversionForm Twinkle - Number conversion &Match expression: &Replace: Perl style format string for the replacement number. Perl style regular expression matching the number format you want to modify. &OK Alt+O &Cancel Alt+C Match expression may not be empty. Replace value may not be empty. Invalid regular expression. RedirectForm Twinkle - Redirect Redirect incoming call to You can specify up to 3 destinations to which you want to redirect the call. If the first destination does not answer the call, the second destination will be tried and so on. &3rd choice destination: &2nd choice destination: &1st choice destination: Address book Select an address from the address book. &OK &Cancel F10 F12 F11 SelectNicForm Twinkle - Select NIC Select the network interface/IP address that you want to use: You have multiple IP addresses. Here you must select which IP address should be used. This IP address will be used inside the SIP messages. Set as default &IP Alt+I Make the selected IP address the default IP address. The next time you start Twinkle, this IP address will be automatically selected. Set as default &NIC Alt+N Make the selected network interface the default interface. The next time you start Twinkle, this interface will be automatically selected. &OK Alt+O If you want to remove or change the default at a later time, you can do that via the system settings. SelectProfileForm Twinkle - Select user profile Select user profile(s) to run: Tick the check boxes of the user profiles that you want to run and press run. Create a new profile with the profile editor. &Wizard Alt+W Create a new profile with the wizard. &Edit Alt+E Edit the highlighted profile. &Delete Alt+D Delete the highlighted profile. Ren&ame Alt+A Rename the highlighted profile. &Set as default Alt+S Make the selected profiles the default profiles. The next time you start Twinkle, these profiles will be automatically run. &Run Alt+R Run Twinkle with the selected profiles. S&ystem settings Alt+Y Edit the system settings. &Cancel Alt+C <html>Before you can use Twinkle, you must create a user profile. Click OK to create a profile.</html> &Profile editor <html>Next you may adjust the system settings. You can change these settings always at a later time. Click OK to view and adjust the system settings.</html> You did not select any user profile to run. Please select a profile. Are you sure you want to delete profile '%1'? Delete profile Failed to delete profile. Failed to rename profile. If you want to remove or change the default at a later time, you can do that via the system settings. Cannot find .twinkle directory in your home directory. Create profile Ed&itor Alt+I Dia&mondcard Alt+M Modify profile Startup profile &Diamondcard Create a profile for a Diamondcard account. With a Diamondcard account you can make worldwide calls to regular and cell phones and send SMS messages. <html>You can use the profile editor to create a profile. With the profile editor you can change many settings to tune the SIP protocol, RTP and many other things. Alternatively you can use the wizard to quickly setup a user profile. The wizard asks you only a few essential settings. If you create a user profile with the wizard you can still edit the full profile with the profile editor at a later time. You can create a Diamondcard account to make worldwide calls to regular and cell phones and send SMS messages. Choose what method you wish to use.</html> SelectUserForm Twinkle - Select user &Cancel Alt+C &Select all Alt+S &OK Alt+O C&lear all Alt+L purpose No need to translate Register Select users that you want to register. Deregister Select users that you want to deregister. Deregister all devices Select users for which you want to deregister all devices. Do not disturb Select users for which you want to enable 'do not disturb'. Auto answer Select users for which you want to enable 'auto answer'. SendFileForm Twinkle - Send File Select file to send. &File: &Subject: &OK Alt+O &Cancel Alt+C File does not exist. Send file... SrvRedirectForm Twinkle - Call Redirection User: There are 3 redirect services: Unconditional: redirect all calls Busy: redirect a call if both lines are busy No answer: redirect a call when the no-answer timer expires &Unconditional &Redirect all calls Alt+R Activate the unconditional redirection service. Redirect to You can specify up to 3 destinations to which you want to redirect the call. If the first destination does not answer the call, the second destination will be tried and so on. &3rd choice destination: &2nd choice destination: &1st choice destination: Address book Select an address from the address book. &Busy &Redirect calls when I am busy Activate the redirection when busy service. &No answer &Redirect calls when I do not answer Activate the redirection on no answer service. &OK Alt+O Accept and save all changes. &Cancel Alt+C Undo your changes and close the window. You have entered an invalid destination. F10 F11 F12 SysSettingsForm Twinkle - System Settings General Audio Ring tones Address book Network Log Select a category for which you want to see or modify the settings. &OK Alt+O Accept and save your changes. &Cancel Alt+C Undo all your changes and close the window. Sound Card Select the sound card for playing the ring tone for incoming calls. Select the sound card to which your microphone is connected. Select the sound card for the speaker function during a call. &Speaker: &Ring tone: Other device: &Microphone: &Validate devices before usage Alt+V Twinkle validates the audio devices before usage to avoid an established call without an audio channel. On startup of Twinkle a warning is given if an audio device is inaccessible. If before making a call, the microphone or speaker appears to be invalid, a warning is given and no call can be made. If before answering a call, the microphone or speaker appears to be invalid, a warning is given and the call will not be answered. Advanced OSS &fragment size: 16 32 64 128 256 The ALSA play period size influences the real time behaviour of your soundcard for playing sound. If your sound frequently drops while using ALSA, you might try a different value here. ALSA &play period size: &ALSA capture period size: The OSS fragment size influences the real time behaviour of your soundcard. If your sound frequently drops while using OSS, you might try a different value here. The ALSA capture period size influences the real time behaviour of your soundcard for capturing sound. If the other side of your call complains about frequently dropping sound, you might try a different value here. &Max log size: The maximum size of a log file in MB. When the log file exceeds this size, a backup of the log file is created and the current log file is zapped. Only one backup log file will be kept. MB Log &debug reports Alt+D Indicates if reports marked as "debug" will be logged. Log &SIP reports Alt+S Indicates if SIP messages will be logged. Log S&TUN reports Alt+T Indicates if STUN messages will be logged. Log m&emory reports Alt+E Indicates if reports concerning memory management will be logged. System tray Create &system tray icon on startup Enable this option if you want a system tray icon for Twinkle. The system tray icon is created when you start Twinkle. &Hide in system tray when closing main window Alt+H Enable this option if you want Twinkle to hide in the system tray when you close the main window. Startup S&tartup hidden in system tray Next time you start Twinkle it will immediately hide in the system tray. This works best when you also select a default user profile. If you always use the same profile(s), then you can mark these profiles as default here. The next time you start Twinkle, you will not be asked to select which profiles to run. The default profiles will automatically run. Services Call &waiting Alt+W With call waiting an incoming call is accepted when only one line is busy. When you disable call waiting an incoming call will be rejected when one line is busy. Hang up &both lines when ending a 3-way conference call. Alt+B Hang up both lines when you press bye to end a 3-way conference call. When this option is disabled, only the active line will be hung up and you can continue talking with the party on the other line. &Maximum calls in call history: The maximum number of calls that will be kept in the call history. &Auto show main window on incoming call after Alt+A When the main window is hidden, it will be automatically shown on an incoming call after the number of specified seconds. Number of seconds after which the main window should be shown. secs Maximum allowed size (0-65535) in bytes of an incoming SIP message over UDP. &SIP port: &RTP port: Max. SIP message size (&TCP): The UDP/TCP port used for sending and receiving SIP messages. Max. SIP message size (&UDP): Maximum allowed size (0-4294967295) in bytes of an incoming SIP message over TCP. The UDP port used for sending and receiving RTP for the first line. The UDP port for the second line is 2 higher. E.g. if port 8000 is used for the first line, then the second line uses port 8002. When you use call transfer then the next even port (eg. 8004) is also used. Ring tone &Play ring tone on incoming call Alt+P Indicates if a ring tone should be played when a call comes in. &Default ring tone Play the default ring tone when a call comes in. C&ustom ring tone Alt+U Play a custom ring tone when a call comes in. Specify the file name of a .wav file that you want to be played as ring tone. Select ring tone file. Ring back tone P&lay ring back tone when network does not play ring back tone Alt+L Play ring back tone while you are waiting for the far-end to answer your call. Depending on your SIP provider the network might provide ring back tone or an announcement. D&efault ring back tone Play the default ring back tone. Cu&stom ring back tone Play a custom ring back tone. Specify the file name of a .wav file that you want to be played as ring back tone. Select ring back tone file. &Lookup name for incoming call On an incoming call, Twinkle will try to find the name belonging to the incoming SIP address in your address book. This name will be displayed. Ove&rride received display name Alt+R The caller may have provided a display name already. Tick this box if you want to override that name with the name you have in your address book. Lookup &photo for incoming call Lookup the photo of a caller in your address book and display it on an incoming call. Ring tones Description of .wav files in file dialog Choose ring tone Ring back tones Description of .wav files in file dialog Choose ring back tone W&eb browser command: Command to start your web browser. If you leave this field empty Twinkle will try to figure out your default web browser. 512 1024 Tip: for crackling sound with PulseAudio, set play period size to maximum. Enable in-call OSD SysTrayPopup Answer Reject Incoming Call TermCapForm Twinkle - Terminal Capabilities &From: Get terminal capabilities of &To: The address that you want to query for capabilities (OPTION request). This can be a full SIP address like sip:example@example.com or just the user part or telephone number of the full address. When you do not specify a full address, then Twinkle will complete the address by using the domain value of your user profile. Address book Select an address from the address book. &OK &Cancel F10 TransferForm Twinkle - Transfer Transfer call to &To: The address of the person you want to transfer the call to. This can be a full SIP address like sip:example@example.com or just the user part or telephone number of the full address. When you do not specify a full address, then Twinkle will complete the address by using the domain value of your user profile. Address book Select an address from the address book. &Blind transfer Alt+B Transfer the call to a third party without contacting that third party yourself. T&ransfer with consultation Alt+R Before transferring the call to a third party, first consult the party yourself. Transfer to other &line Alt+L Connect the remote party on the active line with the remote party on the other line. &OK Alt+O &Cancel F10 TwinkleCore Anonymous Warning: Failed to create log file %1 . Cannot open file for reading: %1 File system error while reading file %1 . Cannot open file for writing: %1 File system error while writing file %1 . Excessive number of socket errors. Built with support for: Contributions: This software contains the following software from 3rd parties: * GSM codec from Jutta Degener and Carsten Bormann, University of Berlin * G.711/G.726 codecs from Sun Microsystems (public domain) * iLBC implementation from RFC 3951 (www.ilbcfreeware.org) * Parts of the STUN project at http://sourceforge.net/projects/stun * Parts of libsrv at http://libsrv.sourceforge.net/ For RTP the following dynamic libraries are linked: Translated to english by <your name> Directory %1 does not exist. Cannot open file %1 . %1 is not set to your home directory. Directory %1 (%2) does not exist. Cannot create directory %1 . %1 is already running. Lock file %2 already exists. Cannot create %1 . Syntax error in file %1 . Failed to backup %1 to %2 unknown name (device is busy) Default device Cannot access the ring tone device (%1). Cannot access the speaker (%1). Cannot access the microphone (%1). Cannot receive incoming TCP connections. Call transfer - %1 Sound card cannot be set to full duplex. Cannot set buffer size on sound card. Sound card cannot be set to %1 channels. Cannot set sound card to 16 bits recording. Cannot set sound card to 16 bits playing. Cannot set sound card sample rate to %1 Opening ALSA driver failed Cannot open ALSA driver for PCM playback Cannot open ALSA driver for PCM capture Cannot resolve STUN server: %1 You are behind a symmetric NAT. STUN will not work. Configure a public IP address in the user profile and create the following static bindings (UDP) in your NAT. public IP: %1 --> private IP: %2 (SIP signaling) public IP: %1-%2 --> private IP: %3-%4 (RTP/RTCP) Cannot reach the STUN server: %1 If you are behind a firewall then you need to open the following UDP ports. Port %1 (SIP signaling) Ports %1-%2 (RTP/RTCP) NAT type discovery via STUN failed. Failed to create file %1 Failed to write data to file %1 Failed to send message. Cannot lock %1 . UserProfileForm Twinkle - User Profile User profile: Select which profile you want to edit. User SIP server Voice mail Instant message Presence RTP audio SIP protocol Transport/NAT Address format Timers Ring tones Scripts Security Select a category for which you want to see or modify the settings. &OK Alt+O Accept and save your changes. &Cancel Alt+C Undo all your changes and close the window. SIP account &User name*: The SIP user name given to you by your provider. It is the user part in your SIP address, username@domain.com This could be a telephone number. This field is mandatory. The domain part of your SIP address, username@domain.com. Instead of a real domain this could also be the hostname or IP address of your SIP proxy. If you want direct IP phone to IP phone communications then you fill in the hostname or IP address of your computer. This field is mandatory. You may fill in the name of your organization. When you make a call, this might be shown to the called party. This is just your full name, e.g. John Doe. It is used as a display name. When you make a call, this display name might be shown to the called party. &Your name: SIP authentication &Realm: Authentication &name: &Password: The realm for authentication. This value must be provided by your SIP provider. If you leave this field empty, then Twinkle will try the user name and password for any realm that it will be challenged with. Your SIP authentication name. Quite often this is the same as your SIP user name. It can be a different name though. Your password for authentication. Registrar &Registrar: The hostname, domain name or IP address of your registrar. If you use an outbound proxy that is the same as your registrar, then you may leave this field empty and only fill in the address of the outbound proxy. The registration expiry time that Twinkle will request. seconds Re&gister at startup Alt+G Indicates if Twinkle should automatically register when you run this user profile. You should disable this when you want to do direct IP phone to IP phone communication without a SIP proxy. Add q-value to registration The q-value indicates the priority of your registered device. If besides Twinkle you register other SIP devices for this account, then the network may use these values to determine which device to try first when delivering a call. The q-value is a value between 0.000 and 1.000. A higher value means a higher priority. Outbound Proxy &Use outbound proxy Alt+U Indicates if Twinkle should use an outbound proxy. If an outbound proxy is used then all SIP requests are sent to this proxy. Without an outbound proxy, Twinkle will try to resolve the SIP address that you type for a call invitation for example to an IP address and send the SIP request there. Outbound &proxy: &Send in-dialog requests to proxy Alt+S SIP requests within a SIP dialog are normally sent to the address in the contact-headers exchanged during call setup. If you tick this box, that address is ignored and in-dialog request are also sent to the outbound proxy. &Don't send a request to proxy if its destination can be resolved locally. Alt+D When you tick this option Twinkle will first try to resolve a SIP address to an IP address itself. If it can, then the SIP request will be sent there. Only when it cannot resolve the address, it will send the SIP request to the proxy (note that an in-dialog request will only be sent to the proxy in this case when you also ticked the previous option.) The hostname, domain name or IP address of your outbound proxy. Co&decs Codecs Available codecs: G.711 A-law G.711 u-law GSM speex-nb (8 kHz) speex-wb (16 kHz) speex-uwb (32 kHz) List of available codecs. Move a codec from the list of available codecs to the list of active codecs. Move a codec from the list of active codecs to the list of available codecs. Active codecs: List of active codecs. These are the codecs that will be used for media negotiation during call setup. The order of the codecs is the order of preference of use. Move a codec upwards in the list of active codecs, i.e. increase its preference of use. Move a codec downwards in the list of active codecs, i.e. decrease its preference of use. &G.711/G.726 payload size: The preferred payload size for the G.711 and G.726 codecs. ms &Follow codec preference from far end on incoming calls Alt+F For incoming calls, follow the preference from the far-end (SDP offer). Pick the first codec from the SDP offer that is also in the list of active codecs. If you disable this option, then the first codec from the active codecs that is also in the SDP offer is picked. Follow codec &preference from far end on outgoing calls Alt+P For outgoing calls, follow the preference from the far-end (SDP answer). Pick the first codec from the SDP answer that is also in the list of active codecs. If you disable this option, then the first codec from the active codecs that is also in the SDP answer is picked. &iLBC iLBC i&LBC payload type: iLBC &payload size (ms): The dynamic type value (96 or higher) to be used for iLBC. 20 30 The preferred payload size for iLBC. &Speex Speex Perceptual &enhancement Alt+E Perceptual enhancement is a part of the decoder which, when turned on, tries to reduce (the perception of) the noise produced by the coding/decoding process. In most cases, perceptual enhancement make the sound further from the original objectively (if you use SNR), but in the end it still sounds better (subjective improvement). &Ultra wide band payload type: Alt+V &Wide band payload type: Alt+B Variable bit-rate (VBR) allows a codec to change its bit-rate dynamically to adapt to the "difficulty" of the audio being encoded. In the example of Speex, sounds like vowels and high-energy transients require a higher bit-rate to achieve good quality, while fricatives (e.g. s,f sounds) can be coded adequately with less bits. For this reason, VBR can achieve a lower bit-rate for the same quality, or a better quality for a certain bit-rate. Despite its advantages, VBR has two main drawbacks: first, by only specifying quality, there's no guarantee about the final average bit-rate. Second, for some real-time applications like voice over IP (VoIP), what counts is the maximum bit-rate, which must be low enough for the communication channel. The dynamic type value (96 or higher) to be used for speex wide band. Co&mplexity: Discontinuous transmission is an addition to VAD/VBR operation, that allows one to stop transmitting completely when the background noise is stationary. The dynamic type value (96 or higher) to be used for speex narrow band. With Speex, it is possible to vary the complexity allowed for the encoder. This is done by controlling how the search is performed with an integer ranging from 1 to 10 in a way that's similar to the -1 to -9 options to gzip and bzip2 compression utilities. For normal use, the noise level at complexity 1 is between 1 and 2 dB higher than at complexity 10, but the CPU requirements for complexity 10 is about 5 times higher than for complexity 1. In practice, the best trade-off is between complexity 2 and 4, though higher settings are often useful when encoding non-speech sounds like DTMF tones. &Narrow band payload type: G.726 G.726 &40 kbps payload type: The dynamic type value (96 or higher) to be used for G.726 40 kbps. The dynamic type value (96 or higher) to be used for G.726 32 kbps. G.726 &24 kbps payload type: The dynamic type value (96 or higher) to be used for G.726 24 kbps. G.726 &32 kbps payload type: The dynamic type value (96 or higher) to be used for G.726 16 kbps. G.726 &16 kbps payload type: Codeword &packing order: RFC 3551 ATM AAL2 There are 2 standards to pack the G.726 codewords into an RTP packet. RFC 3551 is the default packing method. Some SIP devices use ATM AAL2 however. If you experience bad quality using G.726 with RFC 3551 packing, then try ATM AAL2 packing. DT&MF DTMF The dynamic type value (96 or higher) to be used for DTMF events (RFC 2833). DTMF vo&lume: The power level of the DTMF tone in dB. The pause after a DTMF tone. DTMF &duration: DTMF payload &type: DTMF &pause: dB Duration of a DTMF tone. DTMF t&ransport: Auto RFC 2833 Inband Out-of-band (SIP INFO) <h2>RFC 2833</h2> Send DTMF tones as RFC 2833 telephone events. <h2>Inband</h2> Send DTMF inband. <h2>Auto</h2> If the far end of your call supports RFC 2833, then a DTMF tone will be send as RFC 2833 telephone event, otherwise it will be sent inband. <h2>Out-of-band (SIP INFO)</h2> Send DTMF out-of-band via a SIP INFO request. General Protocol options RFC 2543 RFC 3264 Indicates if RFC 2543 (set media IP address in SDP to 0.0.0.0) or RFC 3264 (use direction attributes in SDP) is used to put a call on-hold. Allow m&issing Contact header in 200 OK on REGISTER Alt+I A 200 OK response on a REGISTER request must contain a Contact header. Some registrars however, do not include a Contact header or include a wrong Contact header. This option allows for such a deviation from the specs. &Max-Forwards header is mandatory Alt+M According to RFC 3261 the Max-Forwards header is mandatory. But many implementations do not send this header. If you tick this box, Twinkle will reject a SIP request if Max-Forwards is missing. Put &registration expiry time in contact header Alt+R In a REGISTER message the expiry time for registration can be put in the Contact header or in the Expires header. If you tick this box it will be put in the Contact header, otherwise it goes in the Expires header. &Use compact header names Indicates if compact header names should be used for headers that have a compact form. Allow SDP change during call setup A SIP UAS may send SDP in a 1XX response for early media, e.g. ringing tone. When the call is answered the SIP UAS should send the same SDP in the 200 OK response according to RFC 3261. Once SDP has been received, SDP in subsequent responses should be discarded. By allowing SDP to change during call setup, Twinkle will not discard SDP in subsequent responses and modify the media stream if the SDP is changed. When the SDP in a response is changed, it must have a new version number in the o= line. Use domain &name to create a unique contact header value Alt+N Twinkle creates a unique contact header value by combining the SIP user name and domain: <tt> user_domain@local_ip</tt> This way 2 user profiles, having the same user name but different domain names, have unique contact addresses and hence can be activated simultaneously. Some proxies do not handle a contact header value like this. You can disable this option to get a contact header value like this: <tt> user@local_ip</tt> This format is what most SIP phones use. &Encode Via, Route, Record-Route as list The Via, Route and Record-Route headers can be encoded as a list of comma separated values or as multiple occurrences of the same header. Redirection &Allow redirection Alt+A Indicates if Twinkle should redirect a request if a 3XX response is received. Ask user &permission to redirect Indicates if Twinkle should ask the user before redirecting a request when a 3XX response is received. The number of redirect addresses that Twinkle tries at a maximum before it gives up redirecting a request. This prevents a request from getting redirected forever. SIP extensions disabled supported required preferred Indicates if the 100rel extension (PRACK) is supported: disabled: 100rel extension is disabled supported: 100rel is supported (it is added in the supported header of an outgoing INVITE). A far-end can now require a PRACK on a 1xx response. required: 100rel is required (it is put in the require header of an outgoing INVITE). If an incoming INVITE indicates that it supports 100rel, then Twinkle will require a PRACK when sending a 1xx response. A call will fail when the far-end does not support 100rel. preferred: Similar to required, but if a call fails because the far-end indicates it does not support 100rel (420 response) then the call will be re-attempted without the 100rel requirement. &100 rel (PRACK): Replaces REFER Call transfer (REFER) Alt+T Indicates if Twinkle should transfer a call if a REFER request is received. As&k user permission to transfer Alt+K Indicates if Twinkle should ask the user before transferring a call when a REFER request is received. Hold call &with referrer while setting up call to transfer target Alt+W Indicates if Twinkle should put the current call on hold when a REFER request to transfer a call is received. Ho&ld call with referee before sending REFER Alt+L Indicates if Twinkle should put the current call on hold when you transfer a call. Auto re&fresh subscription to refer event while call transfer is not finished While a call is being transferred, the referee sends NOTIFY messages to the referrer about the progress of the transfer. These messages are only sent for a short interval which length is determined by the referee. If you tick this box, the referrer will automatically send a SUBSCRIBE to lengthen this interval if it is about to expire and the transfer has not yet been completed. Attended refer to AoR (Address of Record) Privacy Privacy options &Send P-Preferred-Identity header when hiding user identity Include a P-Preferred-Identity header with your identity in an INVITE request for a call with identity hiding. SIP transport UDP TCP Transport mode for SIP. In auto mode, the size of a message determines which transport protocol is used. Messages larger than the UDP threshold are sent via TCP. Smaller messages are sent via UDP. T&ransport protocol: UDP t&hreshold: Messages larger than the threshold are sent via TCP. Smaller messages are sent via UDP. NAT traversal &NAT traversal not needed Choose this option when there is no NAT device between you and your SIP proxy or when your SIP provider offers hosted NAT traversal. &Use statically configured public IP address inside SIP messages Indicates if Twinkle should use the public IP address specified in the next field inside SIP message, i.e. in SIP headers and SDP body instead of the IP address of your network interface. When you choose this option you have to create static address mappings in your NAT device as well. You have to map the RTP ports on the public IP address to the same ports on the private IP address of your PC. Choose this option when your SIP provider offers a STUN server for NAT traversal. The hostname, domain name or IP address of the STUN server. &Public IP address: The public IP address of your NAT. Telephone numbers Only &display user part of URI for telephone number If a URI indicates a telephone number, then only display the user part. E.g. if a call comes in from sip:123456@twinklephone.com then display only "123456" to the user. A URI indicates a telephone number if it contains the "user=phone" parameter or when it has a numerical user part and you ticked the next option. &URI with numerical user part is a telephone number If you tick this option, then Twinkle considers a SIP address that has a user part that consists of digits, *, #, + and special symbols only as a telephone number. In an outgoing message, Twinkle will add the "user=phone" parameter to such a URI. &Remove special symbols from numerical dial strings Telephone numbers are often written with special symbols like dashes and brackets to make them readable to humans. When you dial such a number the special symbols must not be dialed. To allow you to simply copy/paste such a number into Twinkle, Twinkle can remove these symbols when you hit the dial button. &Special symbols: The special symbols that may be part of a telephone number for nice formatting, but must be removed when dialing. Number conversion Match expression Replace Move the selected number conversion rule upwards in the list. Move the selected number conversion rule downwards in the list. &Add Add a number conversion rule. Re&move Remove the selected number conversion rule. &Edit Edit the selected number conversion rule. Type a telephone number here an press the Test button to see how it is converted by the list of number conversion rules. &Test Test how a number is converted by the number conversion rules. When an incoming call is received, this timer is started. If the user answers the call, the timer is stopped. If the timer expires before the user answers the call, then Twinkle will reject the call with a "480 User Not Responding". NAT &keep alive: &No answer: Select ring back tone file. Select ring tone file. Ring &back tone: Specify the file name of a .wav file that you want to be played as ring back tone for this user. This ring back tone overrides the ring back tone settings in the system settings. Specify the file name of a .wav file that you want to be played as ring tone for this user. This ring tone overrides the ring tone settings in the system settings. &Ring tone: This script is called when you release a call. <h2>Environment variables</h2> The values of all SIP headers of the outgoing SIP BYE request are passed in environment variables to your script. TWINKLE_TRIGGER=local_release. SIPREQUEST_METHOD=BYE. SIPREQUEST_URI contains the request-URI of the BYE. The name of the user profile will be passed in TWINKLE_USER_PROFILE. Select script file. This script is called when an incoming call fails. <h2>Environment variables</h2> The values of all SIP headers of the outgoing SIP failure response are passed in environment variables to your script. TWINKLE_TRIGGER=in_call_failed. SIPSTATUS_CODE contains the status code of the failure response. SIPSTATUS_REASON contains the reason phrase. The name of the user profile will be passed in TWINKLE_USER_PROFILE. This script is called when the remote party releases a call. <h2>Environment variables</h2> The values of all SIP headers of the incoming SIP BYE request are passed in environment variables to your script. TWINKLE_TRIGGER=remote_release. SIPREQUEST_METHOD=BYE. SIPREQUEST_URI contains the request-URI of the BYE. The name of the user profile will be passed in TWINKLE_USER_PROFILE. You can customize the way Twinkle handles incoming calls. Twinkle can call a script when a call comes in. Based on the output of the script Twinkle accepts, rejects or redirects the call. When accepting the call, the ring tone can be customized by the script as well. The script can be any executable program. Note: Twinkle pauses while your script runs. It is recommended that your script does not take more than 200 ms. When you need more time, you can send the parameters followed by end and keep on running. Twinkle will continue when it receives the end parameter. With your script you can customize call handling by outputting one or more of the following parameters to stdout. Each parameter should be on a separate line. <blockquote> <tt> action=[ continue | reject | dnd | redirect | autoanswer ] reason=<string> contact=<address to redirect to> caller_name=<name of caller to display> ringtone=<file name of .wav file> display_msg=<message to show on display> end </tt> </blockquote> <h2>Parameters</h2> <h3>action</h3> continue - continue call handling as usual reject - reject call dnd - deny call with do not disturb indication redirect - redirect call to address specified by contact autoanswer - automatically answer a call When the script does not write an action to stdout, then the default action is continue. reason: With the reason parameter you can set the reason string for reject or dnd. This might be shown to the far-end user. caller_name: This parameter will override the display name of the caller. ringtone: The ringtone parameter specifies the .wav file that will be played as ring tone when action is continue. <h2>Environment variables</h2> The values of all SIP headers in the incoming INVITE message are passed in environment variables to your script. The variable names are formatted as SIP_<HEADER_NAME> E.g. SIP_FROM contains the value of the from header. TWINKLE_TRIGGER=in_call. SIPREQUEST_METHOD=INVITE. The request-URI of the INVITE will be passed in SIPREQUEST_URI. The name of the user profile will be passed in TWINKLE_USER_PROFILE. This script is called when the remote party answers your call. <h2>Environment variables</h2> The values of all SIP headers of the incoming 200 OK are passed in environment variables to your script. TWINKLE_TRIGGER=out_call_answered. SIPSTATUS_CODE=200. SIPSTATUS_REASON contains the reason phrase. The name of the user profile will be passed in TWINKLE_USER_PROFILE. This script is called when you answer an incoming call. <h2>Environment variables</h2> The values of all SIP headers of the outgoing 200 OK are passed in environment variables to your script. TWINKLE_TRIGGER=in_call_answered. SIPSTATUS_CODE=200. SIPSTATUS_REASON contains the reason phrase. The name of the user profile will be passed in TWINKLE_USER_PROFILE. Call released locall&y: This script is called when an outgoing call fails. <h2>Environment variables</h2> The values of all SIP headers of the incoming SIP failure response are passed in environment variables to your script. TWINKLE_TRIGGER=out_call_failed. SIPSTATUS_CODE contains the status code of the failure response. SIPSTATUS_REASON contains the reason phrase. The name of the user profile will be passed in TWINKLE_USER_PROFILE. This script is called when you make a call. <h2>Environment variables</h2> The values of all SIP headers of the outgoing INVITE are passed in environment variables to your script. TWINKLE_TRIGGER=out_call. SIPREQUEST_METHOD=INVITE. SIPREQUEST_URI contains the request-URI of the INVITE. The name of the user profile will be passed in TWINKLE_USER_PROFILE. Outgoing call a&nswered: Incoming call &failed: &Incoming call: Call released &remotely: Incoming call &answered: O&utgoing call: Out&going call failed: &Enable ZRTP/SRTP encryption ZRTP settings O&nly encrypt audio if remote party indicated ZRTP support in SDP A SIP endpoint supporting ZRTP may indicate ZRTP support during call setup in its signalling. Enabling this option will cause Twinkle only to encrypt calls when the remote party indicates ZRTP support. &Indicate ZRTP support in SDP Twinkle will indicate ZRTP support during call setup in its signalling. &Popup warning when remote party disables encryption during call A remote party of an encrypted call may send a ZRTP go-clear command to stop encryption. When Twinkle receives this command it will popup a warning if this option is enabled. &Voice mail address: The SIP address or telephone number to access your voice mail. &MWI type: Subscription &duration: Mailbox &user name: The hostname, domain name or IP address of your voice mailbox server. Your user name for accessing your voice mailbox. Mailbox &server: Via outbound &proxy Check this option if Twinkle should send SIP messages to the mailbox server via the outbound proxy. &Maximum number of sessions: When you have this number of instant message sessions open, new incoming message sessions will be rejected. Your presence &Publish availability at startup Publish your availability at startup. Publication &refresh interval (sec): Refresh rate of presence publications. Buddy presence &Subscription refresh interval (sec): Refresh rate of presence subscriptions. Dynamic payload type %1 is used more than once. You must fill in a user name for your SIP account. You must fill in a domain name for your SIP account. This could be the hostname or IP address of your PC if you want direct PC to PC dialing. Invalid domain. Invalid user name. Invalid value for registrar. Invalid value for outbound proxy. You must fill in a mailbox user name. You must fill in a mailbox server Invalid mailbox server. Invalid mailbox user name. Value for public IP address missing. Invalid value for STUN server. Ring tones Description of .wav files in file dialog Choose ring tone Ring back tones Description of .wav files in file dialog All files Choose incoming call script Choose incoming call answered script Choose incoming call failed script Choose outgoing call script Choose outgoing call answered script Choose outgoing call failed script Choose local release script Choose remote release script %1 converts to %2 P&ersistent TCP connection Keep the TCP connection established during registration open such that the SIP proxy can reuse this connection to send incoming requests. Application ping packets are sent to test if the connection is still alive. &Send composing indications when typing a message. Twinkle sends a composing indication when you type a message. This way the recipient can see that you are typing. AKA AM&F: A&KA OP: Authentication management field for AKAv1-MD5 authentication. Operator variant key for AKAv1-MD5 authentication. Prepr&ocessing Preprocessing (improves quality at remote end) &Automatic gain control Automatic gain control (AGC) is a feature that deals with the fact that the recording volume may vary by a large amount between different setups. The AGC provides a way to adjust a signal to a reference volume. This is useful because it removes the need for manual adjustment of the microphone gain. A secondary advantage is that by setting the microphone gain to a conservative (low) level, it is easier to avoid clipping. Automatic gain control &level: Automatic gain control level represents percentual value of automatic gain setting of a microphone. Recommended value is about 25%. &Voice activity detection When enabled, voice activity detection detects whether the input signal represents a speech or a silence/background noise. &Noise reduction The noise reduction can be used to reduce the amount of background noise present in the input signal. This provides higher quality speech. Acoustic &Echo Cancellation In any VoIP communication, if a speech from the remote end is played in the local loudspeaker, then it propagates in the room and is captured by the microphone. If the audio captured from the microphone is sent directly to the remote end, then the remote user hears an echo of his voice. An acoustic echo cancellation is designed to remove the acoustic echo before it is sent to the remote end. It is important to understand that the echo canceller is meant to improve the quality on the remote end. Variable &bit-rate Discontinuous &Transmission &Quality: Speex is a lossy codec, which means that it achieves compression at the expense of fidelity of the input speech signal. Unlike some other speech codecs, it is possible to control the tradeoff made between quality and bit-rate. The Speex encoding process is controlled most of the time by a quality parameter that ranges from 0 to 10. bytes Use tel-URI for telephone &number Expand a dialed telephone number to a tel-URI instead of a sip-URI. Accept call &transfer request (incoming REFER) Allow call transfer while consultation in progress When you perform an attended call transfer, you normally transfer the call after you established a consultation call. If you enable this option you can transfer the call while the consultation call is still in progress. This is a non-standard implementation and may not work with all SIP devices. Enable NAT &keep alive Send UDP NAT keep alive packets. If you have enabled STUN or NAT keep alive, then Twinkle will send keep alive packets at this interval rate to keep the address bindings in your NAT device alive. Do&main*: Organi&zation: E&xpiry: Call Hold &variant: &Max redirections: Indicates if the Replaces-extension is supported. An attended call transfer should use the contact URI as a refer target. A contact URI may not be globally routable however. Alternatively the AoR (Address of Record) may be used. A disadvantage is that the AoR may route to multiple endpoints in case of forking whereas the contact URI routes to a single endpoint. Include a P-Asserted-Identity header with your identity in an INVITE request for a call with identity hiding. &Send P-Asserted-Identity header when hiding user identity Use STUN (does not wor&k for incoming TCP) STUN ser&ver: Often the format of the telephone numbers you need to dial is different from the format of the telephone numbers stored in your address book, e.g. your numbers start with a +-symbol followed by a country code, but your provider expects '00' instead of the '+', or you are at the office and all your numbers need to be prefixed with a '9' to access an outside line. Here you can specify number format conversion using Perl style regular expressions and format strings. For each number you dial, Twinkle will try to find a match in the list of match expressions. For the first match it finds, the number will be replaced with the format string. If no match is found, the number stays unchanged. The number conversion rules are also applied to incoming calls, so the numbers are displayed in the format you want. <h3>Example 1</h3> Assume your country code is 31 and you have stored all numbers in your address book in full international number format, e.g. +318712345678. For dialling numbers in your own country you want to strip of the '+31' and replace it by a '0'. For dialling numbers abroad you just want to replace the '+' by '00'. The following rules will do the trick: <blockquote> <tt> Match expression = \+31([0-9]*) , Replace = 0$1 Match expression = \+([0-9]*) , Replace = 00$1 </tt> </blockquote> <h3>Example 2</h3> You are at work and all telephone numbers starting with a 0 should be prefixed with a 9 for an outside line. <blockquote> <tt> Match expression = 0[0-9]* , Replace = 9$& </tt> </blockquote> When ZRTP/SRTP is enabled, then Twinkle will try to encrypt the audio of each call you originate or receive. Encryption will only succeed if the remote party has ZRTP/SRTP support enabled. If the remote party does not support ZRTP/SRTP, then the audio channel will stay unencrypted. <H2>Message waiting indication type</H2> If your provider offers the message waiting indication service, then Twinkle can show you when new voice mail messages are waiting. Ask your provider which type of message waiting indication is offered. <H3>Unsolicited</H3> Asterisk provides unsolicited message waiting indication. <H3>Solicited</H3> Solicited message waiting indication as specified by RFC 3842. Unsolicited Solicited Solicited MWI For solicited MWI, an endpoint subscribes to the message status for a limited duration. Just before the duration expires, the endpoint should refresh the subscription. WizardForm Twinkle - Wizard The hostname, domain name or IP address of the STUN server. S&TUN server: The SIP user name given to you by your provider. It is the user part in your SIP address, username@domain.com This could be a telephone number. This field is mandatory. &Domain*: Choose your SIP service provider. If your SIP service provider is not in the list, then select Other and fill in the settings you received from your provider. If you select one of the predefined SIP service providers then you only have to fill in your name, user name, authentication name and password. &Authentication name: &Your name: Your SIP authentication name. Quite often this is the same as your SIP user name. It can be a different name though. The domain part of your SIP address, username@domain.com. Instead of a real domain this could also be the hostname or IP address of your SIP proxy. If you want direct IP phone to IP phone communications then you fill in the hostname or IP address of your computer. This field is mandatory. This is just your full name, e.g. John Doe. It is used as a display name. When you make a call, this display name might be shown to the called party. SIP pro&xy: The hostname, domain name or IP address of your SIP proxy. If this is the same value as your domain, you may leave this field empty. &SIP service provider: &Password: &User name*: Your password for authentication. &OK Alt+O &Cancel Alt+C None (direct IP to IP calls) Other User profile wizard: You must fill in a user name for your SIP account. You must fill in a domain name for your SIP account. This could be the hostname or IP address of your PC if you want direct PC to PC dialing. Invalid value for SIP proxy. Invalid value for STUN server. YesNoDialog &Yes &No incoming_call Answer Reject