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122 lines
3.8 KiB
Plaintext
122 lines
3.8 KiB
Plaintext
This is SFLphone, release 0.9.6
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Copyright (c) Savoir-faire Linux, Inc 2004-2009
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<sflphoneteam@savoirfairelinux.com>
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This program is free software; you can redistribute it and/or modify
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it under the terms of the GNU General Public License as published by
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the Free Software Foundation; either version 3 of the License, or
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(at your option) any later version.
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This program is distributed in the hope that it will be useful,
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but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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GNU General Public License for more details.
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You should have received a copy of the GNU General Public License
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along with this program; if not, write to the Free Software
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Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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Introduction
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------------
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SFLPhone is a Voice-over-IP software phone. We want it to be:
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- user friendly (fast, sleek, easy to learn interface)
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- corporate grade (transfers, holds, perfect audio quality)
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- fully compatible with Asterisk (SIP and IAX protocols)
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- customizable
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As the SIP/audio daemon and the user interface are separate processes,
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it is easy to provide different user interfaces. SFLPhone0.8 comes with
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a GTK graphical user interface, an interactive command line interface,
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and even scripts to control the daemon from the shell.
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SFLPhone is actually used by the support team of Savoir-Faire Linux Inc.
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More information is available on the project homepage:
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http://www.sflphone.org/
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Short description of content of source tree
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-------------------------------------------
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- ringtones/ contains the different ringtones.
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- stund/ is an implementation of the protocol STUN used when there is a NAT.
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- utilspp/ allows to implement a singleton.
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- src/ is the core of SFLphone. It contains the main.cpp, managerimpl.cpp
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files, audio and gui directories, and files about signalisation SIP. Later,
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it should be better, when IAX will be implemented, that a directory groups
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these protocols.
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The ManagerImpl class is the intermediaire between all the layer in the tree.
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- src/audio/ is the audio layer. It contains all about tones, dtmf,
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audiodriver, rtp layer, audio codec ulaw, alaw and gsm.
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- src/dbus, the dbus xml interfaces, and c++ bindings
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About Savoir-Faire Linux
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------------------------
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Savoir-Faire Linux is a consulting company based in Montreal, Quebec.
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For more information, please check out our website:
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http://www.savoirfairelinux.com/
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How to compile
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--------------
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# First compile the SIP library
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cd libs/pjproject-1.0.3
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./autogen.sh
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./configure --disable-ilbc-codec && make clean && make dep && make
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sudo make install
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If you already have installed a different version of
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pjsip on your system, you will most likely have some errors
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such as :
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"ln: creating symbolic link `./libpjsip-sfl.a': File exists"
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after executing make install.
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Simply remove those files and run "make install" again.
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# Then the daemon
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cd ../..
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./autogen.sh --prefix=/usr --with-debug
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make
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sudo make install
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# And the GNOME client
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cd sflphone-client-gnome
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./autogen.sh --prefix=/usr --with-debug
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make
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sudo make install
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Done !
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Free SIP/IAx2 accounts
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---------------------
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You may create a free SIP/IAX2 account through the account creation wizard in the both client (KDE and GNOME).
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By doing this, you will be able to call other account registered to the same server.
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Registered to this kind of account, you will be able to test your headset by dialing *100
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ECHO test
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--------
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Dial *100, on sip.sflphone.org
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Contributing to SFLPhone
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------------------------
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Of course we love patches. And contributions. And spring rolls.
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Development website: http://projects.savoirfairelinux.net/wiki/sflphone
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Do not hesitate to join us and post comments, suggestions, questions
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and general feedback on the forge.novell mailing-list.
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-- The SFLPhone Team
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