kpsi.1

KphoneSI (kpsi) is a SIP (Session Initiation Protocol) user agent for Linux, with which you can in

.TH kpsi 1
.SH NAME
kpsi (KPhone SI) - A  SIP User Agent
.SH SYNOPSIS
.B kpsi [-u <user>] [-a <host>] [-p <port>] [-t]
.SH Options
.TP
\fB-u <user>\fR  \fIuser identification in database\fR
.br
<user> is not the user part of the SIP-URI (SIP-Phone number), but the user's ID in the database ($HOME/.qt/kphonesirc)
.TP
\fB-a <host>\fR \fIdissipate-host\fR
.br
<host> the interface used by the phone. If this parameter is not set, KPhoneIS scans the interface table. Th user is ask to make the choice.
.TP
\fB-p <port>\fR \fIport number you listen on\fR
.br
<port> The listen-to port. It is used to address the UA from outside . The default is 5060. If the parameter is not set, KPhoneIS finds an even 
port >= 5060 and sets it.
.TP
\fB-t\fR \fIUA point to point \fR
.br
In this case the dissipate-host equals the host part of sip URI of the UA). -t is one possible method to start point to point sessions without involving a proxy. The registration is switched off and the SIP-URI takes the form 
sip:port@dissipate-host:port.
.br
.SH DESCRIPTION
When starting kpsi for the first time, you should enter your identity information in the window appearing. The information required for successful 
sipping are:
\fB'Full Name'\fR, \fB'User part of SIP URL'\fR and \fB'Host part of SIP URL'\fR . Filling the \fB'Outbound Proxy'\fR Field is recommended, possibly necessary. 
Study you provider's documentation for the details. The field should contain your proxy's IP address or that of a Media Gateway if used. If your provider 
demands authentication, the \fBAuthentication user'\fR may be entered here, too. The password is asked for in a special popup-window Mark the auto register 
selection if you wish kpsi to automatically register your identity on application start. If you want to use multiple identities, start KPhone SI 
with the '-u' option and the username of your selection.
.TP 0
If you are behind a NAT (Network Address Translation) device, you should go to \fBPreferences->SIP-Preferences->Socket\fR and check the '\fBUse STUN Server\fR' and specify  the STUN server's IP address
.TP 0
There are different ways to start a call. The basic way is to type the SIP-URI and  then select the call type (audio, video, ...)
If you select the call type first, you may type the URI right after the Call Widged popped up.
.TP 0 
A started call is displayed in the \fB'Session Info'\fR window, right clicking an entry pops up  a window with the controls for that call.
This is the place where you release the call (hangup entry). Alternatively you may double click the entry. Then the call widget 
pops up,  to present the control functions.  
.TP 0
You should edit the \fBPhone Book\fR and enter all your favorite contacts there.  Of course this is not mandatory but it increases 
your efficiency with kpsi. When adding contacts to your Phone Book, simply click Add and enter the person's real name,  SIP identity and the 
optional description. If you double click a existing contact,  the URI appears in the dial window. 
.TP 0
The Phone Book is the place to request \fBpresence information\fR. The presence info is displayed in the \fBpresence window (buddy list)\fR. If you right click an entry, it pops up a window with the media types allowing you to start a call with that buddy.