UserProfileForm 0 0 777 592 Twinkle - User Profile User profile: false 0 0 Select which profile you want to edit. 0 0 150 0 Select a category for which you want to see or modify the settings. 0 32 32 User :/icons/images/penguin.png:/icons/images/penguin.png SIP server :/icons/images/package_network.png:/icons/images/package_network.png Voice mail :/icons/images/mwi_none.png:/icons/images/mwi_none.png Instant message :/icons/images/message32.png:/icons/images/message32.png Presence :/icons/images/presence.png:/icons/images/presence.png RTP audio :/icons/images/kmix.png:/icons/images/kmix.png SIP protocol :/icons/images/package_system.png:/icons/images/package_system.png Transport/NAT :/icons/images/yast_babelfish.png:/icons/images/yast_babelfish.png Address format :/icons/images/yast_PhoneTTOffhook.png:/icons/images/yast_PhoneTTOffhook.png Timers :/icons/images/clock.png:/icons/images/clock.png Ring tones :/icons/images/knotify.png:/icons/images/knotify.png Scripts :/icons/images/edit.png:/icons/images/edit.png Security :/icons/images/encrypted32.png:/icons/images/encrypted32.png Qt::Horizontal QSizePolicy::Expanding 441 20 Accept and save your changes. &OK Alt+O true Undo all your changes and close the window. &Cancel Alt+C 0 0 QFrame::StyledPanel 0 21 QFrame::StyledPanel User false 10 SIP account &User name*: false usernameLineEdit Do&main*: false domainLineEdit Organi&zation: false organizationLineEdit 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: false displayLineEdit SIP authentication &Realm: false authRealmLineEdit Authentication &name: false authNameLineEdit 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: false authAkaAmfLineEdit A&KA OP: false authAkaOpLineEdit Your password for authentication. QLineEdit::Password &Password: false authPasswordLineEdit Authentication management field for AKAv1-MD5 authentication. Operator variant key for AKAv1-MD5 authentication. Qt::Vertical QSizePolicy::Expanding 20 110 21 QFrame::StyledPanel SIP server false 10 Registrar &Registrar: false registrarLineEdit 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. E&xpiry: false expirySpinBox 90 0 The registration expiry time that Twinkle will request. 999999 100 seconds false Qt::Horizontal QSizePolicy::Expanding 260 20 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. Re&gister at startup Alt+G 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. Add q-value to registration The q-value is a value between 0.000 and 1.000. A higher value means a higher priority. Qt::Horizontal QSizePolicy::Expanding 210 20 Outbound Proxy 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. &Use outbound proxy Alt+U true Outbound &proxy: false proxyLineEdit 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.) &Don't send a request to proxy if its destination can be resolved locally. Alt+D true The hostname, domain name or IP address of your outbound proxy. 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. &Send in-dialog requests to proxy Alt+S Qt::Vertical QSizePolicy::Expanding 20 100 21 QFrame::StyledPanel RTP audio false 10 Co&decs &G.711/G.726 payload size: false ptimeSpinBox 0 0 46 0 32767 32767 The preferred payload size for the G.711 and G.726 codecs. 10 80 10 ms false Qt::Horizontal QSizePolicy::Expanding 121 20 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 incoming calls Alt+F 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. Follow codec &preference from far end on outgoing calls Alt+P Qt::Vertical QSizePolicy::Expanding 20 16 Codecs Available codecs: false 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) Qt::Vertical QSizePolicy::Expanding 20 20 Move a codec from the list of available codecs to the list of active codecs. :/icons/images/1rightarrow.png:/icons/images/1rightarrow.png Move a codec from the list of active codecs to the list of available codecs. :/icons/images/1leftarrow.png:/icons/images/1leftarrow.png Qt::Vertical QSizePolicy::Expanding 20 21 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. Qt::Vertical QSizePolicy::Expanding 20 21 Move a codec upwards in the list of active codecs, i.e. increase its preference of use. :/icons/images/1uparrow.png:/icons/images/1uparrow.png Move a codec downwards in the list of active codecs, i.e. decrease its preference of use. :/icons/images/1downarrow.png:/icons/images/1downarrow.png Qt::Vertical QSizePolicy::Expanding 20 31 Prepr&ocessing Preprocessing (improves quality at remote end) 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 Alt+A true Automatic gain control &level: false spxDspAgcLevelSpinBox true Automatic gain control level represents percentual value of automatic gain setting of a microphone. Recommended value is about 25%. 1 100 When enabled, voice activity detection detects whether the input signal represents a speech or a silence/background noise. &Voice activity detection Alt+V The noise reduction can be used to reduce the amount of background noise present in the input signal. This provides higher quality speech. &Noise reduction Alt+N 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. Acoustic &Echo Cancellation Alt+E Qt::Horizontal QSizePolicy::Expanding 31 20 Qt::Vertical QSizePolicy::Expanding 20 121 &iLBC iLBC i&LBC payload type: false ilbcPayloadSpinBox iLBC &payload size (ms): false ilbcPayloadSizeComboBox The dynamic type value (96 or higher) to be used for iLBC. 96 127 The preferred payload size for iLBC. 20 30 Qt::Horizontal QSizePolicy::Expanding 71 20 Qt::Vertical QSizePolicy::Expanding 20 81 &Speex Speex 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). Perceptual &enhancement Alt+E &Ultra wide band payload type: false spxUwbPayloadSpinBox &Wide band payload type: false spxWbPayloadSpinBox 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. Variable &bit-rate Alt+B The dynamic type value (96 or higher) to be used for speex wide band. 96 127 Discontinuous transmission is an addition to VAD/VBR operation, that allows one to stop transmitting completely when the background noise is stationary. Discontinuous &Transmission Alt+T The dynamic type value (96 or higher) to be used for speex wide band. 96 127 The dynamic type value (96 or higher) to be used for speex narrow band. 96 127 &Quality: false spxQualitySpinBox 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. 0 10 Co&mplexity: false spxComplexitySpinBox 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. 1 10 &Narrow band payload type: false spxNbPayloadSpinBox Qt::Horizontal QSizePolicy::Expanding 31 20 Qt::Vertical QSizePolicy::Expanding 20 121 G.726 G.726 G.726 &40 kbps payload type: false g72640PayloadSpinBox The dynamic type value (96 or higher) to be used for G.726 40 kbps. 96 127 The dynamic type value (96 or higher) to be used for G.726 32 kbps. 0 127 G.726 &24 kbps payload type: false g72624PayloadSpinBox The dynamic type value (96 or higher) to be used for G.726 24 kbps. 96 127 G.726 &32 kbps payload type: false g72632PayloadSpinBox The dynamic type value (96 or higher) to be used for G.726 16 kbps. 96 127 G.726 &16 kbps payload type: false g72616PayloadSpinBox Qt::Horizontal QSizePolicy::Expanding 231 20 Codeword &packing order: false g726PackComboBox 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 Qt::Horizontal QSizePolicy::Expanding 141 20 Qt::Vertical QSizePolicy::Expanding 20 150 DT&MF DTMF Qt::Horizontal QSizePolicy::Expanding 280 20 0 0 49 0 32767 32767 The dynamic type value (96 or higher) to be used for DTMF events (RFC 2833). 96 127 ms false DTMF vo&lume: false dtmfVolumeSpinBox The power level of the DTMF tone in dB. -63 0 10 -10 0 0 49 0 32767 32767 The pause after a DTMF tone. 20 100 10 DTMF &duration: false dtmfDurationSpinBox ms false DTMF payload &type: false dtmfPayloadTypeSpinBox DTMF &pause: false dtmfPauseSpinBox dB false 0 0 49 0 32767 32767 Duration of a DTMF tone. 40 500 10 DTMF t&ransport: false dtmfTransportComboBox <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) Qt::Horizontal QSizePolicy::Expanding 161 20 Qt::Vertical QSizePolicy::Expanding 20 120 21 QFrame::StyledPanel SIP protocol false 10 General Qt::Vertical QSizePolicy::Expanding 20 16 Protocol options Call Hold &variant: false holdVariantComboBox 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 Qt::Horizontal QSizePolicy::Expanding 70 20 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. Allow m&issing Contact header in 200 OK on REGISTER Alt+I 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. &Max-Forwards header is mandatory Alt+M 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. Put &registration expiry time in contact header Alt+R Indicates if compact header names should be used for headers that have a compact form. &Use compact header names Alt+U 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. Allow SDP change during call setup 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. Use domain &name to create a unique contact header value Alt+N 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. &Encode Via, Route, Record-Route as list Alt+E Redirection Indicates if Twinkle should redirect a request if a 3XX response is received. &Allow redirection Alt+A Indicates if Twinkle should ask the user before redirecting a request when a 3XX response is received. Ask user &permission to redirect Alt+P &Max redirections: false maxRedirectSpinBox 0 0 46 0 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. 1 5 Qt::Horizontal QSizePolicy::Expanding 80 20 SIP extensions 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): false ext100relComboBox Indicates if the Replaces-extension is supported. Replaces REFER Call transfer (REFER) Indicates if Twinkle should transfer a call if a REFER request is received. Accept call &transfer request (incoming REFER) Alt+T Indicates if Twinkle should ask the user before transferring a call when a REFER request is received. As&k user permission to transfer Alt+K Indicates if Twinkle should put the current call on hold when a REFER request to transfer a call 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 you transfer a call. Ho&ld call with referee before sending REFER Alt+L 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. Auto re&fresh subscription to refer event while call transfer is not finished Alt+F 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. Attended refer to AoR (Address of Record) 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. Allow call transfer while consultation in progress Qt::Vertical QSizePolicy::Expanding 20 200 Privacy Privacy options Include a P-Preferred-Identity header with your identity in an INVITE request for a call with identity hiding. &Send P-Preferred-Identity header when hiding user identity Alt+S 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 Alt+A Qt::Vertical QSizePolicy::Expanding 20 331 21 QFrame::StyledPanel Transport/NAT false 10 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 Qt::Horizontal QSizePolicy::Expanding 151 20 T&ransport protocol: false sipTransportComboBox UDP t&hreshold: false udpThresholdSpinBox Messages larger than the threshold are sent via TCP. Smaller messages are sent via UDP. bytes 65535 100 1300 Qt::Horizontal QSizePolicy::Expanding 81 20 NAT traversal Choose this option when there is no NAT device between you and your SIP proxy or when your SIP provider offers hosted NAT traversal. &NAT traversal not needed Alt+N 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. &Use statically configured public IP address inside SIP messages Alt+U &Public IP address: false 21 publicIPLineEdit The public IP address of your NAT. Choose this option when your SIP provider offers a STUN server for NAT traversal. Use STUN (does not wor&k for incoming TCP) Alt+S STUN ser&ver: false 21 stunServerLineEdit The hostname, domain name or IP address of the STUN server. 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. P&ersistent TCP connection Alt+E Send UDP NAT keep alive packets. Enable NAT &keep alive Alt+K Qt::Vertical QSizePolicy::Expanding 20 80 21 QFrame::StyledPanel Address format false 10 Telephone numbers 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. Only &display user part of URI for telephone number Alt+D 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. &URI with numerical user part is a telephone number Alt+U 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. &Remove special symbols from numerical dial strings Alt+R Expand a dialed telephone number to a tel-URI instead of a sip-URI. Use tel-URI for telephone &number Alt+N &Special symbols: false specialLineEdit The special symbols that may be part of a telephone number for nice formatting, but must be removed when dialing. Number conversion 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> QAbstractItemView::SingleSelection QAbstractItemView::SelectRows true false false Match expression Replace Qt::Vertical QSizePolicy::Expanding 20 21 Move the selected number conversion rule upwards in the list. :/icons/images/1uparrow.png:/icons/images/1uparrow.png Move the selected number conversion rule downwards in the list. :/icons/images/1downarrow.png:/icons/images/1downarrow.png Qt::Vertical QSizePolicy::Expanding 20 31 Add a number conversion rule. &Add Alt+A Remove the selected number conversion rule. Re&move Alt+M Edit the selected number conversion rule. &Edit Alt+E Qt::Horizontal QSizePolicy::Expanding 291 20 Type a telephone number here an press the Test button to see how it is converted by the list of number conversion rules. Test how a number is converted by the number conversion rules. &Test Alt+T Qt::Vertical QSizePolicy::Expanding 20 20 21 QFrame::StyledPanel Timers false 10 seconds false 0 0 55 0 55 32767 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. 10 900 10 0 0 55 0 55 32767 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". 600 10 NAT &keep alive: false tmrNatKeepaliveSpinBox &No answer: false tmrNoanswerSpinBox Qt::Horizontal QSizePolicy::Expanding 270 20 Qt::Vertical QSizePolicy::Expanding 20 450 21 QFrame::StyledPanel Ring tones false 10 Qt::TabFocus Select ring back tone file. :/icons/images/fileopen.png:/icons/images/fileopen.png Qt::TabFocus Select ring tone file. :/icons/images/fileopen.png:/icons/images/fileopen.png Ring &back tone: false ringbackLineEdit 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: false ringtoneLineEdit Qt::Vertical QSizePolicy::Expanding 20 391 21 QFrame::StyledPanel Scripts false 10 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 Select script file. :/icons/images/fileopen.png:/icons/images/fileopen.png Qt::TabFocus Select script file. :/icons/images/fileopen.png:/icons/images/fileopen.png Qt::TabFocus Select script file. :/icons/images/fileopen.png:/icons/images/fileopen.png 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 Select script file. :/icons/images/fileopen.png:/icons/images/fileopen.png 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 Select script file. :/icons/images/fileopen.png:/icons/images/fileopen.png 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: false inCallFailedLineEdit Qt::TabFocus Select script file. :/icons/images/fileopen.png:/icons/images/fileopen.png 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: false inCallAnsweredLineEdit Incoming call &failed: false inCallFailedLineEdit &Incoming call: false incomingCallScriptLineEdit Qt::TabFocus Select script file. :/icons/images/fileopen.png:/icons/images/fileopen.png Call released &remotely: false inCallFailedLineEdit Incoming call &answered: false inCallAnsweredLineEdit Qt::TabFocus Select script file. :/icons/images/fileopen.png:/icons/images/fileopen.png O&utgoing call: false incomingCallScriptLineEdit Out&going call failed: false inCallFailedLineEdit Qt::Vertical QSizePolicy::Expanding 20 190 21 QFrame::StyledPanel Security false 10 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. &Enable ZRTP/SRTP encryption Alt+E ZRTP settings 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. O&nly encrypt audio if remote party indicated ZRTP support in SDP Alt+N Twinkle will indicate ZRTP support during call setup in its signalling. &Indicate ZRTP support in SDP Alt+I 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. &Popup warning when remote party disables encryption during call Alt+P Qt::Vertical QSizePolicy::Expanding 20 241 21 QFrame::StyledPanel Voice mail false 10 &Voice mail address: false vmAddressLineEdit 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>Unsolicited</H3> Asterisk provides unsolicited message waiting indication. <H3>Solicited</H3> Solicited message waiting indication as specified by RFC 3842. Unsolicited Solicited Qt::Horizontal QSizePolicy::Expanding 221 20 &MWI type: false mwiTypeComboBox Solicited MWI Qt::Horizontal QSizePolicy::Expanding 120 20 Subscription &duration: false mwiDurationSpinBox Mailbox &user name: false mwiUserLineEdit The hostname, domain name or IP address of your voice mailbox server. 90 0 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. 999999 100 seconds false Qt::Horizontal QSizePolicy::Expanding 190 20 Your user name for accessing your voice mailbox. Mailbox &server: false mwiServerLineEdit Check this option if Twinkle should send SIP messages to the mailbox server via the outbound proxy. Via outbound &proxy Alt+P Qt::Vertical QSizePolicy::Expanding 20 211 21 QFrame::StyledPanel Instant message false 10 &Maximum number of sessions: false imMaxSessionsSpinBox When you have this number of instant message sessions open, new incoming message sessions will be rejected. 65535 Qt::Horizontal QSizePolicy::Expanding 201 20 Twinkle sends a composing indication when you type a message. This way the recipient can see that you are typing. &Send composing indications when typing a message. Alt+S Qt::Vertical QSizePolicy::Expanding 20 350 21 QFrame::StyledPanel Presence false 10 Your presence Publish your availability at startup. &Publish availability at startup Alt+P Publication &refresh interval (sec): false presPublishTimeSpinBox Refresh rate of presence publications. 999999 100 Qt::Horizontal QSizePolicy::Expanding 231 20 Buddy presence &Subscription refresh interval (sec): false presSubscribeTimeSpinBox Refresh rate of presence subscriptions. 999999 100 Qt::Horizontal QSizePolicy::Expanding 191 20 Qt::Vertical QSizePolicy::Expanding 20 281 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 pAssertedIdCheckBox sipTransportComboBox udpThresholdSpinBox 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 map list categoryListBox currentRowChanged(int) UserProfileForm showCategory(int) 31 63 20 20 cancelPushButton clicked() UserProfileForm reject() 126 576 20 20 okPushButton clicked() UserProfileForm validate() 39 576 20 20 useProxyCheckBox toggled(bool) proxyTextLabel setEnabled(bool) 240 329 240 357 useProxyCheckBox toggled(bool) proxyLineEdit setEnabled(bool) 240 329 353 357 useProxyCheckBox toggled(bool) allRequestsCheckBox setEnabled(bool) 240 329 240 387 allowRedirectionCheckBox toggled(bool) askUserRedirectCheckBox setEnabled(bool) 242 456 242 484 allowRedirectionCheckBox toggled(bool) maxRedirectTextLabel setEnabled(bool) 242 456 242 512 allowRedirectionCheckBox toggled(bool) maxRedirectSpinBox setEnabled(bool) 242 456 358 512 useProxyCheckBox toggled(bool) proxyNonResolvableCheckBox setEnabled(bool) 240 329 240 415 allowReferCheckBox toggled(bool) askUserReferCheckBox setEnabled(bool) 242 178 242 174 allowReferCheckBox toggled(bool) refereeHoldCheckBox setEnabled(bool) 242 178 242 170 profileComboBox activated(QString) UserProfileForm changeProfile(QString) 116 32 20 20 openRingtoneToolButton clicked() UserProfileForm chooseRingtone() 281 60 20 20 openRingbackToolButton clicked() UserProfileForm chooseRingback() 281 64 20 20 openIncomingCallScriptToolButton clicked() UserProfileForm chooseIncomingCallScript() 281 60 20 20 addCodecPushButton clicked() UserProfileForm addCodec() 451 266 20 20 rmvCodecPushButton clicked() UserProfileForm removeCodec() 451 300 20 20 upCodecPushButton clicked() UserProfileForm upCodec() 702 266 20 20 downCodecPushButton clicked() UserProfileForm downCodec() 702 300 20 20 availCodecListBox itemDoubleClicked(QListWidgetItem*) UserProfileForm addCodec() 243 211 20 20 activeCodecListBox itemDoubleClicked(QListWidgetItem*) UserProfileForm removeCodec() 493 211 20 20 openInCallAnsweredToolButton clicked() UserProfileForm chooseInCallAnsweredScript() 281 62 20 20 openInCallFailedToolButton clicked() UserProfileForm chooseInCallFailedScript() 281 63 20 20 openLocalReleaseToolButton clicked() UserProfileForm chooseLocalReleaseScript() 281 68 20 20 openOutCallAnsweredToolButton clicked() UserProfileForm chooseOutCallAnsweredScript() 281 66 20 20 openOutCallFailedToolButton clicked() UserProfileForm chooseOutCallFailedScript() 281 67 20 20 openOutCallToolButton clicked() UserProfileForm chooseOutgoingCallScript() 281 64 20 20 openRemoteReleaseToolButton clicked() UserProfileForm chooseRemoteReleaseScript() 281 70 20 20 upConversionPushButton clicked() UserProfileForm upConversion() 266 101 20 20 downConversionPushButton clicked() UserProfileForm downConversion() 266 103 20 20 addConversionPushButton clicked() UserProfileForm addConversion() 228 89 20 20 editConversionPushButton clicked() UserProfileForm editConversion() 260 89 20 20 removePushButton clicked() UserProfileForm removeConversion() 244 89 20 20 testConversionPushButton clicked() UserProfileForm testConversion() 267 80 20 20 zrtpEnabledCheckBox toggled(bool) zrtpSettingsGroupBox setEnabled(bool) 225 58 225 62 mwiTypeComboBox activated(int) UserProfileForm changeMWIType(int) 269 64 20 20 regAddQvalueCheckBox toggled(bool) regQvalueLineEdit setEnabled(bool) 241 251 447 250 sipTransportComboBox activated(int) UserProfileForm changeSipTransportProtocol(int) 368 158 20 20 spxDspAgcCheckBox toggled(bool) spxDspAgcLevelTextLabel setEnabled(bool) 242 179 258 179 spxDspAgcCheckBox toggled(bool) spxDspAgcLevelSpinBox setEnabled(bool) 242 179 274 179 natStunRadioButton toggled(bool) stunServerLineEdit setEnabled(bool) 418 351 431 374 natStaticRadioButton toggled(bool) publicIPLineEdit setEnabled(bool) 334 285 437 312