UserProfileForm
0
0
771
582
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
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SIP server
:/icons/images/package_network.png:/icons/images/package_network.png
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Voice mail
:/icons/images/mwi_none.png:/icons/images/mwi_none.png
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Instant message
:/icons/images/message32.png:/icons/images/message32.png
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Presence
:/icons/images/presence.png:/icons/images/presence.png
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RTP audio
:/icons/images/kmix.png:/icons/images/kmix.png
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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
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Timers
:/icons/images/clock.png:/icons/images/clock.png
-
Ring tones
:/icons/images/knotify.png:/icons/images/knotify.png
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Scripts
:/icons/images/edit.png:/icons/images/edit.png
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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
-
&Domain*:
false
domainLineEdit
-
Or&ganization:
false
organizationLineEdit
-
The SIP user name given to you by your provider. It is the user part in your SIP address, <b>username</b>@domain.com This could be a telephone number.
<br><br>
This field is mandatory.
-
The domain part of your SIP address, username@<b>domain.com</b>. Instead of a real domain this could also be the hostname or IP address of your <b>SIP proxy</b>. If you want direct IP phone to IP phone communications then you fill in the hostname or IP address of your computer.
<br><br>
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.
-
&Expiry:
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
50
10
-
ms
false
-
Qt::Horizontal
QSizePolicy::Expanding
121
20
-
<p>
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.
<p>
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
-
<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.
<p>
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
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Move a codec from the list of active codecs to the list of available codecs.
:/icons/images/1leftarrow.png:/icons/images/1leftarrow.png
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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 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 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.
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>
<p>Send DTMF tones as RFC 2833 telephone events.</p>
<h2>Inband</h2>
<p>Send DTMF inband.</p>
<h2>Auto</h2>
<p>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.
</p>
<h2>Out-of-band (SIP INFO)</h2>
<p>
Send DTMF out-of-band via a SIP INFO request.
</p>
-
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 ®istration 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
-
<p>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.</p>
<p>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.</p>
Allow SDP change during call setup
-
<p>
Twinkle creates a unique contact header value by combining the SIP user name and domain:
</p>
<p>
<tt> user_domain@local_ip</tt>
</p>
<p>
This way 2 user profiles, having the same user name but different domain names, have unique contact addresses and hence can be activated simultaneously.
</p>
<p>
Some proxies do not handle a contact header value like this. You can disable this option to get a contact header value like this:
</p>
<p>
<tt> user@local_ip</tt>
</p>
<p>
This format is what most SIP phones use.
</p>
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 re&directions:
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:<br><br>
<b>disabled</b>: 100rel extension is disabled
<br><br>
<b>supported</b>: 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.
<br><br>
<b>required</b>: 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.
<br><br>
<b>preferred</b>: 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-extenstion 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 endoint.
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
-
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.<br><br>
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 work for incoming TCP)
Alt+S
-
-
S&TUN server:
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
-
-
<p>
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.
</p>
<p>
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.
</p>
<p>
The number conversion rules are also applied to incoming calls, so the numbers are displayed in the format you want.
</p>
<h3>Example 1</h3>
<p>
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'.
</p>
<p>
The following rules will do the trick:
</p>
<blockquote>
<tt>
Match expression = \+31([0-9]*) , Replace = 0$1<br>
Match expression = \+([0-9]*) , Replace = 00$1</br>
</tt>
</blockquote>
<h3>Example 2</h3>
<p>
You are at work and all telephone numbers starting with a 0 should be prefixed with a 9 for an outside line.
</p>
<blockquote>
<tt>
Match expression = 0[0-9]* , Replace = 9$&<br>
</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
-
<p>
Specify the file name of a .wav file that you want to be played as ring back tone for this user.
</p>
<p>
This ring back tone overrides the ring back tone settings in the system settings.
</p>
-
<p>
Specify the file name of a .wav file that you want to be played as ring tone for this user.
</p>
<p>
This ring tone overrides the ring tone settings in the system settings.
</p>
-
&Ring tone:
false
ringtoneLineEdit
-
Qt::Vertical
QSizePolicy::Expanding
20
391
-
21
QFrame::StyledPanel
Scripts
false
10
-
-
<p>
This script is called when you release a call.
</p>
<h2>Environment variables</h2>
<p>
The values of all SIP headers of the outgoing SIP BYE request are passed in environment variables to your script.
</p>
<p>
<b>TWINKLE_TRIGGER=local_release</b>. <b>SIPREQUEST_METHOD=BYE</b>. <b>SIPREQUEST_URI</b> contains the request-URI of the BYE. The name of the user profile will be passed in <b>TWINKLE_USER_PROFILE</b>.
-
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
-
<p>
This script is called when an incoming call fails.
</p>
<h2>Environment variables</h2>
<p>
The values of all SIP headers of the outgoing SIP failure response are passed in environment variables to your script.
</p>
<p>
<b>TWINKLE_TRIGGER=in_call_failed</b>. <b>SIPSTATUS_CODE</b> contains the status code of the failure response. <b>SIPSTATUS_REASON</b> contains the reason phrase. The name of the user profile will be passed in <b>TWINKLE_USER_PROFILE</b>.
-
Qt::TabFocus
Select script file.
:/icons/images/fileopen.png:/icons/images/fileopen.png
-
<p>
This script is called when the remote party releases a call.
</p>
<h2>Environment variables</h2>
<p>
The values of all SIP headers of the incoming SIP BYE request are passed in environment variables to your script.
</p>
<p>
<b>TWINKLE_TRIGGER=remote_release</b>. <b>SIPREQUEST_METHOD=BYE</b>. <b>SIPREQUEST_URI</b> contains the request-URI of the BYE. The name of the user profile will be passed in <b>TWINKLE_USER_PROFILE</b>.
-
Qt::TabFocus
Select script file.
:/icons/images/fileopen.png:/icons/images/fileopen.png
-
<p>
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.
</p>
<p>
<b>Note:</b> 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 <b>end</b> and keep on running. Twinkle will continue when it receives the <b>end</b> parameter.
</p>
<p>
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.
</p>
<p>
<blockquote>
<tt>
action=[ continue | reject | dnd | redirect | autoanswer ]<br>
reason=<string><br>
contact=<address to redirect to><br>
caller_name=<name of caller to display><br>
ringtone=<file name of .wav file><br>
display_msg=<message to show on display><br>
end<br>
</tt>
</blockquote>
</p>
<h2>Parameters</h2>
<h3>action</h3>
<p>
<b>continue</b> - continue call handling as usual<br>
<b>reject</b> - reject call<br>
<b>dnd</b> - deny call with do not disturb indication<br>
<b>redirect</b> - redirect call to address specified by <b>contact</b><br>
<b>autoanswer</b> - automatically answer a call<br>
</p>
<p>
When the script does not write an action to stdout, then the default action is continue.
</p>
<p>
<b>reason: </b>
With the reason parameter you can set the reason string for reject or dnd. This might be shown to the far-end user.
</p>
<p>
<b>caller_name: </b>
This parameter will override the display name of the caller.
</p>
<p>
<b>ringtone: </b>
The ringtone parameter specifies the .wav file that will be played as ring tone when action is continue.
</p>
<h2>Environment variables</h2>
<p>
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 <b>SIP_<HEADER_NAME></b> E.g. SIP_FROM contains the value of the from header.
</p>
<p>
TWINKLE_TRIGGER=in_call. SIPREQUEST_METHOD=INVITE. The request-URI of the INVITE will be passed in <b>SIPREQUEST_URI</b>. The name of the user profile will be passed in <b>TWINKLE_USER_PROFILE</b>.
-
<p>
This script is called when the remote party answers your call.
</p>
<h2>Environment variables</h2>
<p>
The values of all SIP headers of the incoming 200 OK are passed in environment variables to your script.
</p>
<p>
<b>TWINKLE_TRIGGER=out_call_answered</b>. <b>SIPSTATUS_CODE=200</b>. <b>SIPSTATUS_REASON</b> contains the reason phrase. The name of the user profile will be passed in <b>TWINKLE_USER_PROFILE</b>.
-
<p>
This script is called when you answer an incoming call.
</p>
<h2>Environment variables</h2>
<p>
The values of all SIP headers of the outgoing 200 OK are passed in environment variables to your script.
</p>
<p>
<b>TWINKLE_TRIGGER=in_call_answered</b>. <b>SIPSTATUS_CODE=200</b>. <b>SIPSTATUS_REASON</b> contains the reason phrase. The name of the user profile will be passed in <b>TWINKLE_USER_PROFILE</b>.
-
Call released locall&y:
false
inCallFailedLineEdit
-
Qt::TabFocus
Select script file.
:/icons/images/fileopen.png:/icons/images/fileopen.png
-
<p>
This script is called when an outgoing call fails.
</p>
<h2>Environment variables</h2>
<p>
The values of all SIP headers of the incoming SIP failure response are passed in environment variables to your script.
</p>
<p>
<b>TWINKLE_TRIGGER=out_call_failed</b>. <b>SIPSTATUS_CODE</b> contains the status code of the failure response. <b>SIPSTATUS_REASON</b> contains the reason phrase. The name of the user profile will be passed in <b>TWINKLE_USER_PROFILE</b>.
-
<p>
This script is called when you make a call.
</p>
<h2>Environment variables</h2>
<p>
The values of all SIP headers of the outgoing INVITE are passed in environment variables to your script.
</p>
<p>
<b>TWINKLE_TRIGGER=out_call</b>. <b>SIPREQUEST_METHOD=INVITE</b>. <b>SIPREQUEST_URI</b> contains the request-URI of the INVITE. The name of the user profile will be passed in <b>TWINKLE_USER_PROFILE</b>.
-
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 unecrypted.
&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>
<p>
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.
</p>
<H3>Unsollicited</H3>
<p>
Asterisk provides unsollicited message waiting indication.
</p>
<H3>Sollicited</H3>
<p>
Sollicited message waiting indication as specified by RFC 3842.
</p>
-
Unsollicited
-
Sollicited
-
Qt::Horizontal
QSizePolicy::Expanding
221
20
-
&MWI type:
false
mwiTypeComboBox
-
Sollicited 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 sollicited 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
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
q3valuelist.h
map
list
categoryListBox
currentRowChanged(int)
UserProfileForm
showCategory(int)
20
20
20
20
cancelPushButton
clicked()
UserProfileForm
reject()
20
20
20
20
okPushButton
clicked()
UserProfileForm
validate()
20
20
20
20
useProxyCheckBox
toggled(bool)
proxyTextLabel
setEnabled(bool)
20
20
20
20
useProxyCheckBox
toggled(bool)
proxyLineEdit
setEnabled(bool)
20
20
20
20
useProxyCheckBox
toggled(bool)
allRequestsCheckBox
setEnabled(bool)
20
20
20
20
allowRedirectionCheckBox
toggled(bool)
askUserRedirectCheckBox
setEnabled(bool)
20
20
20
20
allowRedirectionCheckBox
toggled(bool)
maxRedirectTextLabel
setEnabled(bool)
20
20
20
20
allowRedirectionCheckBox
toggled(bool)
maxRedirectSpinBox
setEnabled(bool)
20
20
20
20
useProxyCheckBox
toggled(bool)
proxyNonResolvableCheckBox
setEnabled(bool)
20
20
20
20
allowReferCheckBox
toggled(bool)
askUserReferCheckBox
setEnabled(bool)
20
20
20
20
allowReferCheckBox
toggled(bool)
refereeHoldCheckBox
setEnabled(bool)
20
20
20
20
profileComboBox
activated(QString)
UserProfileForm
changeProfile(QString)
20
20
20
20
openRingtoneToolButton
clicked()
UserProfileForm
chooseRingtone()
20
20
20
20
openRingbackToolButton
clicked()
UserProfileForm
chooseRingback()
20
20
20
20
openIncomingCallScriptToolButton
clicked()
UserProfileForm
chooseIncomingCallScript()
20
20
20
20
addCodecPushButton
clicked()
UserProfileForm
addCodec()
20
20
20
20
rmvCodecPushButton
clicked()
UserProfileForm
removeCodec()
20
20
20
20
upCodecPushButton
clicked()
UserProfileForm
upCodec()
20
20
20
20
downCodecPushButton
clicked()
UserProfileForm
downCodec()
20
20
20
20
availCodecListBox
itemDoubleClicked(QListWidgetItem*)
UserProfileForm
addCodec()
20
20
20
20
activeCodecListBox
itemDoubleClicked(QListWidgetItem*)
UserProfileForm
removeCodec()
20
20
20
20
openInCallAnsweredToolButton
clicked()
UserProfileForm
chooseInCallAnsweredScript()
20
20
20
20
openInCallFailedToolButton
clicked()
UserProfileForm
chooseInCallFailedScript()
20
20
20
20
openLocalReleaseToolButton
clicked()
UserProfileForm
chooseLocalReleaseScript()
20
20
20
20
openOutCallAnsweredToolButton
clicked()
UserProfileForm
chooseOutCallAnsweredScript()
20
20
20
20
openOutCallFailedToolButton
clicked()
UserProfileForm
chooseOutCallFailedScript()
20
20
20
20
openOutCallToolButton
clicked()
UserProfileForm
chooseOutgoingCallScript()
20
20
20
20
openRemoteReleaseToolButton
clicked()
UserProfileForm
chooseRemoteReleaseScript()
20
20
20
20
upConversionPushButton
clicked()
UserProfileForm
upConversion()
20
20
20
20
downConversionPushButton
clicked()
UserProfileForm
downConversion()
20
20
20
20
addConversionPushButton
clicked()
UserProfileForm
addConversion()
20
20
20
20
editConversionPushButton
clicked()
UserProfileForm
editConversion()
20
20
20
20
removePushButton
clicked()
UserProfileForm
removeConversion()
20
20
20
20
testConversionPushButton
clicked()
UserProfileForm
testConversion()
20
20
20
20
zrtpEnabledCheckBox
toggled(bool)
zrtpSettingsGroupBox
setEnabled(bool)
20
20
20
20
mwiTypeComboBox
activated(int)
UserProfileForm
changeMWIType(int)
20
20
20
20
regAddQvalueCheckBox
toggled(bool)
regQvalueLineEdit
setEnabled(bool)
20
20
20
20
sipTransportComboBox
activated(int)
UserProfileForm
changeSipTransportProtocol(int)
20
20
20
20
spxDspAgcCheckBox
toggled(bool)
spxDspAgcLevelTextLabel
setEnabled(bool)
20
20
20
20
spxDspAgcCheckBox
toggled(bool)
spxDspAgcLevelSpinBox
setEnabled(bool)
20
20
20
20