UserProfileForm 0 0 783 594 Twinkle - User Profile User profile: false 7 0 0 0 Select which profile you want to edit. 1 7 0 0 150 0 0 Select a category for which you want to see or modify the settings. User :/icons/images/penguin.png SIP server :/icons/images/package_network.png Voice mail :/icons/images/mwi_none.png Instant message :/icons/images/message32.png Presence :/icons/images/presence.png RTP audio :/icons/images/kmix.png SIP protocol :/icons/images/package_system.png Transport/NAT :/icons/images/yast_babelfish.png Address format :/icons/images/yast_PhoneTTOffhook.png Timers :/icons/images/clock.png Ring tones :/icons/images/knotify.png Scripts :/icons/images/edit.png Security :/icons/images/encrypted32.png 441 20 QSizePolicy::Expanding Qt::Horizontal &OK Alt+O true Accept and save your changes. &Cancel Alt+C Undo all your changes and close the window. 7 5 0 0 Box 0 21 QFrame::Box User 10 false SIP account &User name*: usernameLineEdit false &Domain*: domainLineEdit false Or&ganization: organizationLineEdit false 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: displayLineEdit false SIP authentication &Realm: authRealmLineEdit false Authentication &name: authNameLineEdit false 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. AKA AM&F: authAkaAmfLineEdit false A&KA OP: authAkaOpLineEdit false QLineEdit::Password Your password for authentication. &Password: authPasswordLineEdit false Authentication management field for AKAv1-MD5 authentication. Operator variant key for AKAv1-MD5 authentication. 20 110 QSizePolicy::Expanding Qt::Vertical 1 21 QFrame::Box SIP server 10 false Registrar &Registrar: registrarLineEdit false 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. &Expiry: expirySpinBox false 90 0 999999 100 The registration expiry time that Twinkle will request. seconds false 260 20 QSizePolicy::Expanding Qt::Horizontal 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. 210 20 QSizePolicy::Expanding Qt::Horizontal 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. true Outbound &proxy: proxyLineEdit false &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.) true The hostname, domain name or IP address of your 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. 20 100 QSizePolicy::Expanding Qt::Vertical 2 21 QFrame::Box RTP audio 10 false Co&decs &G.711/G.726 payload size: ptimeSpinBox false 1 0 0 0 46 0 32767 32767 50 10 10 The preferred payload size for the G.711 and G.726 codecs. ms false 121 20 QSizePolicy::Expanding Qt::Horizontal &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. 20 16 QSizePolicy::Expanding Qt::Vertical Codecs Available codecs: false Q3ScrollView::Manual Q3ScrollView::AlwaysOff List of available codecs. G.711 A-law G.711 u-law GSM speex-nb (8 kHz) speex-wb (16 kHz) speex-uwb (32 kHz) 20 20 QSizePolicy::Expanding Qt::Vertical :/icons/images/1rightarrow.png Move a codec from the list of available codecs to the list of active codecs. :/icons/images/1leftarrow.png Move a codec from the list of active codecs to the list of available codecs. 20 21 QSizePolicy::Expanding Qt::Vertical Active codecs: false 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. 20 21 QSizePolicy::Expanding Qt::Vertical :/icons/images/1uparrow.png Move a codec upwards in the list of active codecs, i.e. increase its preference of use. :/icons/images/1downarrow.png Move a codec downwards in the list of active codecs, i.e. decrease its preference of use. 20 31 QSizePolicy::Expanding Qt::Vertical Prepr&ocessing Preprocessing (improves quality at remote end) &Automatic gain control Alt+A 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. true Automatic gain control &level: spxDspAgcLevelSpinBox false true 100 1 Automatic gain control level represents percentual value of automatic gain setting of a microphone. Recommended value is about 25%. &Voice activity detection Alt+V When enabled, voice activity detection detects whether the input signal represents a speech or a silence/background noise. &Noise reduction Alt+N 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 Alt+E 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. 31 20 QSizePolicy::Expanding Qt::Horizontal 20 121 QSizePolicy::Expanding Qt::Vertical &iLBC iLBC i&LBC payload type: ilbcPayloadSpinBox false iLBC &payload size (ms): ilbcPayloadSizeComboBox false 127 96 The dynamic type value (96 or higher) to be used for iLBC. The preferred payload size for iLBC. 20 30 71 20 QSizePolicy::Expanding Qt::Horizontal 20 81 QSizePolicy::Expanding Qt::Vertical &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: spxUwbPayloadSpinBox false &Wide band payload type: spxWbPayloadSpinBox false Variable &bit-rate 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. 127 96 The dynamic type value (96 or higher) to be used for speex wide band. Discontinuous &Transmission Alt+T Discontinuous transmission is an addition to VAD/VBR operation, that allows to stop transmitting completely when the background noise is stationary. 127 96 The dynamic type value (96 or higher) to be used for speex wide band. 127 96 The dynamic type value (96 or higher) to be used for speex narrow band. &Quality: spxQualitySpinBox false 10 0 Speex is a lossy codec, which means that it achives 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. Co&mplexity: spxComplexitySpinBox false 10 1 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: spxNbPayloadSpinBox false 31 20 QSizePolicy::Expanding Qt::Horizontal 20 121 QSizePolicy::Expanding Qt::Vertical G.726 G.726 G.726 &40 kbps payload type: g72640PayloadSpinBox false 127 96 The dynamic type value (96 or higher) to be used for G.726 40 kbps. 127 0 The dynamic type value (96 or higher) to be used for G.726 32 kbps. G.726 &24 kbps payload type: g72624PayloadSpinBox false 127 96 The dynamic type value (96 or higher) to be used for G.726 24 kbps. G.726 &32 kbps payload type: g72632PayloadSpinBox false 127 96 The dynamic type value (96 or higher) to be used for G.726 16 kbps. G.726 &16 kbps payload type: g72616PayloadSpinBox false 231 20 QSizePolicy::Expanding Qt::Horizontal Codeword &packing order: g726PackComboBox false 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. RFC 3551 ATM AAL2 141 20 QSizePolicy::Expanding Qt::Horizontal 20 150 QSizePolicy::Expanding Qt::Vertical DT&MF DTMF 280 20 QSizePolicy::Expanding Qt::Horizontal 1 0 0 0 49 0 32767 32767 127 96 The dynamic type value (96 or higher) to be used for DTMF events (RFC 2833). ms false DTMF vo&lume: dtmfVolumeSpinBox false 0 -63 10 -10 The power level of the DTMF tone in dB. 1 0 0 0 49 0 32767 32767 100 20 10 The pause after a DTMF tone. DTMF &duration: dtmfDurationSpinBox false ms false DTMF payload &type: dtmfPayloadTypeSpinBox false DTMF &pause: dtmfPauseSpinBox false dB false 1 0 0 0 49 0 32767 32767 500 40 10 Duration of a DTMF tone. DTMF t&ransport: dtmfTransportComboBox false <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. Auto RFC 2833 Inband Out-of-band (SIP INFO) 161 20 QSizePolicy::Expanding Qt::Horizontal 20 120 QSizePolicy::Expanding Qt::Vertical 3 21 QFrame::Box SIP protocol 10 false General 20 16 QSizePolicy::Expanding Qt::Vertical Protocol options Call &Hold variant: holdVariantComboBox false 1 0 0 0 110 0 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. RFC 2543 RFC 3264 70 20 QSizePolicy::Expanding Qt::Horizontal 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 Alt+U 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 Alt+E 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 Alt+P Indicates if Twinkle should ask the user before redirecting a request when a 3XX response is received. Max re&directions: maxRedirectSpinBox false 1 0 0 0 46 0 5 1 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. 80 20 QSizePolicy::Expanding Qt::Horizontal SIP extensions 1 0 0 0 120 0 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. disabled supported required preferred &100 rel (PRACK): ext100relComboBox false Replaces Indicates if the Replaces-extenstion is supported. REFER Call transfer (REFER) Accept call &transfer request (incoming 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 Alt+F 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) 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 endoint. 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. 20 200 QSizePolicy::Expanding Qt::Vertical Privacy Privacy options &Send P-Preferred-Identity header when hiding user identity Alt+S Include a P-Preferred-Identity header with your identity in an INVITE request for a call with identity hiding. 20 331 QSizePolicy::Expanding Qt::Vertical 4 21 QFrame::Box Transport/NAT 10 false SIP transport 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. Auto UDP TCP 151 20 QSizePolicy::Expanding Qt::Horizontal T&ransport protocol: sipTransportComboBox false UDP t&hreshold: udpThresholdSpinBox false bytes 65535 100 1300 Messages larger than the threshold are sent via TCP. Smaller messages are sent via UDP. 81 20 QSizePolicy::Expanding Qt::Horizontal NAT traversal &NAT traversal not needed Alt+N 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 Alt+U 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. &Public IP address: 21 publicIPLineEdit false The public IP address of your NAT. Use &STUN (does not work for incoming TCP) Alt+S Choose this option when your SIP provider offers a STUN server for NAT traversal. S&TUN server: 21 stunServerLineEdit false The hostname, domain name or IP address of the STUN server. P&ersistent TCP connection Alt+E 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. Enable NAT &keep alive Alt+K Send UDP NAT keep alive packets. 20 80 QSizePolicy::Expanding Qt::Vertical 5 21 QFrame::Box Address format 10 false Telephone numbers Only &display user part of URI for telephone number Alt+D 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 Alt+U 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 Alt+R 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. Use tel-URI for telephone &number Alt+N Expand a dialed telephone number to a tel-URI instead of a sip-URI. &Special symbols: specialLineEdit false The special symbols that may be part of a telephone number for nice formatting, but must be removed when dialing. Number conversion true Q3ListView::AllColumns Often the format of the telphone 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> Match expression true true Replace true true 20 21 QSizePolicy::Expanding Qt::Vertical :/icons/images/1uparrow.png Move the selected number conversion rule upwards in the list. :/icons/images/1downarrow.png Move the selected number conversion rule downwards in the list. 20 31 QSizePolicy::Expanding Qt::Vertical &Add Alt+A Add a number conversion rule. Re&move Alt+M Remove the selected number conversion rule. &Edit Alt+E Edit the selected number conversion rule. 291 20 QSizePolicy::Expanding Qt::Horizontal Type a telephone number here an press the Test button to see how it is converted by the list of number conversion rules. &Test Alt+T Test how a number is converted by the number conversion rules. 20 20 QSizePolicy::Expanding Qt::Vertical 6 21 QFrame::Box Timers 10 false seconds false 0 0 0 0 55 0 55 32767 900 10 10 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. 0 0 0 0 55 0 55 32767 600 10 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: tmrNatKeepaliveSpinBox false &No answer: tmrNoanswerSpinBox false 270 20 QSizePolicy::Expanding Qt::Horizontal 20 450 QSizePolicy::Expanding Qt::Vertical 7 21 QFrame::Box Ring tones 10 false Qt::TabFocus :/icons/images/fileopen.png Select ring back tone file. Qt::TabFocus :/icons/images/fileopen.png Select ring tone file. Ring &back tone: ringbackLineEdit false 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: ringtoneLineEdit false 20 391 QSizePolicy::Expanding Qt::Vertical 8 21 QFrame::Box Scripts 10 false 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. Qt::TabFocus :/icons/images/fileopen.png Select script file. Qt::TabFocus :/icons/images/fileopen.png Select script file. Qt::TabFocus :/icons/images/fileopen.png 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. Qt::TabFocus :/icons/images/fileopen.png Select script file. 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. Qt::TabFocus :/icons/images/fileopen.png Select script file. You can customize the way Twinkle handles incoming calls. Twinkle can call a script when a call comes in. Based on the ouput 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 outputing 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: inCallFailedLineEdit false Qt::TabFocus :/icons/images/fileopen.png Select script file. 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: inCallAnsweredLineEdit false Incoming call &failed: inCallFailedLineEdit false &Incoming call: incomingCallScriptLineEdit false Qt::TabFocus :/icons/images/fileopen.png Select script file. Call released &remotely: inCallFailedLineEdit false Incoming call &answered: inCallAnsweredLineEdit false Qt::TabFocus :/icons/images/fileopen.png Select script file. O&utgoing call: incomingCallScriptLineEdit false Out&going call failed: inCallFailedLineEdit false 20 190 QSizePolicy::Expanding Qt::Vertical 9 21 QFrame::Box Security 10 false &Enable ZRTP/SRTP encryption Alt+E 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 unecrypted. ZRTP settings O&nly encrypt audio if remote party indicated ZRTP support in SDP Alt+N 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 Alt+I Twinkle will indicate ZRTP support during call setup in its signalling. &Popup warning when remote party disables encryption during call Alt+P 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. 20 241 QSizePolicy::Expanding Qt::Vertical 10 21 QFrame::Box Voice mail 10 false &Voice mail address: vmAddressLineEdit false The SIP address or telephone number to access your voice mail. <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>Unsollicited</H3> Asterisk provides unsollicited message waiting indication. <H3>Sollicited</H3> Sollicited message waiting indication as specified by RFC 3842. Unsollicited Sollicited 221 20 QSizePolicy::Expanding Qt::Horizontal &MWI type: mwiTypeComboBox false Sollicited MWI 120 20 QSizePolicy::Expanding Qt::Horizontal Subscription &duration: mwiDurationSpinBox false Mailbox &user name: mwiUserLineEdit false The hostname, domain name or IP address of your voice mailbox server. 90 0 999999 100 For sollicited MWI, an endpoint subscribes to the message status for a limited duration. Just before the duration expires, the endpoint should refresh the subscription. seconds false 190 20 QSizePolicy::Expanding Qt::Horizontal Your user name for accessing your voice mailbox. Mailbox &server: mwiServerLineEdit false Via outbound &proxy Alt+P Check this option if Twinkle should send SIP messages to the mailbox server via the outbound proxy. 20 211 QSizePolicy::Expanding Qt::Vertical 11 21 QFrame::Box Instant message 10 false &Maximum number of sessions: imMaxSessionsSpinBox false 65535 When you have this number of instant message sessions open, new incoming message sessions will be rejected. 201 20 QSizePolicy::Expanding Qt::Horizontal &Send composing indications when typing a message. Alt+S Twinkle sends a composing indication when you type a message. This way the recipient can see that you are typing. 20 350 QSizePolicy::Expanding Qt::Vertical 12 21 QFrame::Box Presence 10 false Your presence &Publish availability at startup Alt+P Publish your availability at startup. Publication &refresh interval (sec): presPublishTimeSpinBox false 999999 100 Refresh rate of presence publications. 231 20 QSizePolicy::Expanding Qt::Horizontal Buddy presence &Subscription refresh interval (sec): presSubscribeTimeSpinBox false 999999 100 Refresh rate of presence subscriptions. 191 20 QSizePolicy::Expanding Qt::Horizontal 20 281 QSizePolicy::Expanding Qt::Vertical displayLineEdit usernameLineEdit domainLineEdit organizationLineEdit authRealmLineEdit authNameLineEdit authPasswordLineEdit authAkaOpLineEdit authAkaAmfLineEdit registrarLineEdit expirySpinBox regAtStartupCheckBox regAddQvalueCheckBox regQvalueLineEdit useProxyCheckBox proxyLineEdit allRequestsCheckBox proxyNonResolvableCheckBox vmAddressLineEdit mwiTypeComboBox mwiUserLineEdit mwiServerLineEdit mwiViaProxyCheckBox mwiDurationSpinBox imMaxSessionsSpinBox isComposingCheckBox presPublishCheckBox presPublishTimeSpinBox presSubscribeTimeSpinBox rtpAudioTabWidget availCodecListBox addCodecPushButton rmvCodecPushButton activeCodecListBox upCodecPushButton downCodecPushButton ptimeSpinBox inFarEndCodecPrefCheckBox outFarEndCodecPrefCheckBox spxDspAgcCheckBox spxDspAgcLevelSpinBox spxDspVadCheckBox spxDspNrdCheckBox spxDspAecCheckBox ilbcPayloadSpinBox ilbcPayloadSizeComboBox spxVbrCheckBox spxDtxCheckBox spxPenhCheckBox spxQualitySpinBox spxComplexitySpinBox spxNbPayloadSpinBox spxWbPayloadSpinBox spxUwbPayloadSpinBox g72616PayloadSpinBox g72624PayloadSpinBox g72632PayloadSpinBox g72640PayloadSpinBox g726PackComboBox dtmfTransportComboBox dtmfPayloadTypeSpinBox dtmfDurationSpinBox dtmfPauseSpinBox dtmfVolumeSpinBox sipProtoclTabWidget holdVariantComboBox maxForwardsCheckBox missingContactCheckBox regTimeCheckBox compactHeadersCheckBox multiValuesListCheckBox useDomainInContactCheckBox allowSdpChangeCheckBox allowRedirectionCheckBox askUserRedirectCheckBox maxRedirectSpinBox ext100relComboBox extReplacesCheckBox allowReferCheckBox askUserReferCheckBox refereeHoldCheckBox referrerHoldCheckBox refreshReferSubCheckBox referAorCheckBox pPreferredIdCheckBox sipTransportComboBox udpThresholdSpinBox natNoneRadioButton natStaticRadioButton publicIPLineEdit natStunRadioButton stunServerLineEdit persistentTcpCheckBox displayTelUserCheckBox numericalUserIsTelCheckBox removeSpecialCheckBox specialLineEdit useTelUriCheckBox conversionListView upConversionPushButton downConversionPushButton addConversionPushButton removePushButton editConversionPushButton testConversionLineEdit testConversionPushButton tmrNoanswerSpinBox tmrNatKeepaliveSpinBox ringtoneLineEdit ringbackLineEdit openRingtoneToolButton openRingbackToolButton incomingCallScriptLineEdit openIncomingCallScriptToolButton inCallAnsweredLineEdit openInCallAnsweredToolButton inCallFailedLineEdit openInCallFailedToolButton outCallLineEdit openOutCallToolButton outCallAnsweredLineEdit openOutCallAnsweredToolButton outCallFailedLineEdit openOutCallFailedToolButton localReleaseLineEdit openLocalReleaseToolButton remoteReleaseLineEdit openRemoteReleaseToolButton zrtpEnabledCheckBox zrtpSendIfSupportedCheckBox zrtpSdpCheckBox zrtpGoClearWarningCheckBox okPushButton cancelPushButton profileComboBox categoryListBox user.h q3valuelist.h map list categoryListBox highlighted(int) UserProfileForm showCategory(int) cancelPushButton clicked() UserProfileForm reject() okPushButton clicked() UserProfileForm validate() useProxyCheckBox toggled(bool) proxyTextLabel setEnabled(bool) useProxyCheckBox toggled(bool) proxyLineEdit setEnabled(bool) useProxyCheckBox toggled(bool) allRequestsCheckBox setEnabled(bool) allowRedirectionCheckBox toggled(bool) askUserRedirectCheckBox setEnabled(bool) allowRedirectionCheckBox toggled(bool) maxRedirectTextLabel setEnabled(bool) allowRedirectionCheckBox toggled(bool) maxRedirectSpinBox setEnabled(bool) useProxyCheckBox toggled(bool) proxyNonResolvableCheckBox setEnabled(bool) natStaticRadioButton toggled(bool) publicIPTextLabel setEnabled(bool) natStaticRadioButton toggled(bool) publicIPLineEdit setEnabled(bool) natStunRadioButton toggled(bool) stunServerTextLabel setEnabled(bool) natStunRadioButton toggled(bool) stunServerLineEdit setEnabled(bool) allowReferCheckBox toggled(bool) askUserReferCheckBox setEnabled(bool) allowReferCheckBox toggled(bool) refereeHoldCheckBox setEnabled(bool) profileComboBox activated(QString) UserProfileForm changeProfile(QString) openRingtoneToolButton clicked() UserProfileForm chooseRingtone() openRingbackToolButton clicked() UserProfileForm chooseRingback() openIncomingCallScriptToolButton clicked() UserProfileForm chooseIncomingCallScript() addCodecPushButton clicked() UserProfileForm addCodec() rmvCodecPushButton clicked() UserProfileForm removeCodec() upCodecPushButton clicked() UserProfileForm upCodec() downCodecPushButton clicked() UserProfileForm downCodec() availCodecListBox doubleClicked(Q3ListBoxItem*) UserProfileForm addCodec() activeCodecListBox doubleClicked(Q3ListBoxItem*) UserProfileForm removeCodec() openInCallAnsweredToolButton clicked() UserProfileForm chooseInCallAnsweredScript() openInCallFailedToolButton clicked() UserProfileForm chooseInCallFailedScript() openLocalReleaseToolButton clicked() UserProfileForm chooseLocalReleaseScript() openOutCallAnsweredToolButton clicked() UserProfileForm chooseOutCallAnsweredScript() openOutCallFailedToolButton clicked() UserProfileForm chooseOutCallFailedScript() openOutCallToolButton clicked() UserProfileForm chooseOutgoingCallScript() openRemoteReleaseToolButton clicked() UserProfileForm chooseRemoteReleaseScript() upConversionPushButton clicked() UserProfileForm upConversion() downConversionPushButton clicked() UserProfileForm downConversion() addConversionPushButton clicked() UserProfileForm addConversion() editConversionPushButton clicked() UserProfileForm editConversion() removePushButton clicked() UserProfileForm removeConversion() testConversionPushButton clicked() UserProfileForm testConversion() zrtpEnabledCheckBox toggled(bool) zrtpSettingsGroupBox setEnabled(bool) mwiTypeComboBox activated(int) UserProfileForm changeMWIType(int) regAddQvalueCheckBox toggled(bool) regQvalueLineEdit setEnabled(bool) sipTransportComboBox activated(int) UserProfileForm changeSipTransportProtocol(int) spxDspAgcCheckBox toggled(bool) spxDspAgcLevelTextLabel setEnabled(bool) spxDspAgcCheckBox toggled(bool) spxDspAgcLevelSpinBox setEnabled(bool) natStunRadioButton toggled(bool) natKeepaliveCheckBox setDisabled(bool)