This package is based on the package 'asterisk' from project 'network:telephony'.

Asterisk is a complete PBX in software. It provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in three protocols and can inter operate with almost all standards-based telephony equipment using relatively inexpensive hardware.

Asterisk provides voice mail services with directory, call conferencing, interactive voice response, and call queuing. It has support for three-way calling, caller ID services, ADSI, SIP, and H.323 (as both client and gateway).

This package is based on the package 'asterisk-addons' from project 'network:telephony'.

Asterisk is a complete PBX in software. It runs on Linux and provides
all of the features you would expect from a PBX and more. Asterisk
does voice over IP in three protocols, and can inter operate with
almost all standards-based telephony equipment using relatively
inexpensive hardware.
This package contains additional addons for asterisk.

This package is based on the package 'asterisk-app_ldap' from project 'network:telephony'.

LDAPget is an asterisk application which can lookup data in an LDAP directory (e.g. OpenLDAP or ActiveDirectory).
LDAPget does not require you to use a specific LDAP schema.
If the requested data is found in the directory it is assigned to a given variable an the priority is increased by 101.

This package is based on the package 'asterisk-flite' from project 'network:telephony'.

This provides the "Flite" dialplan application for the Asterisk Open Source
PBX, which allows you to use the CMU Flite text-to-speech engine with Asterisk.

The application uses the exact same syntax as the standard "Festival" Asterisk
application.

This package is based on the package 'asterisk-perl' from project 'network:telephony'.

asterisk-perl is a collection of perl modules to be used with the Asterisk PBX.

This package is based on the package 'callweaver-RC' from project 'network:telephony'.

Callweaver is an Open Source PBX and telephony development platform that can both replace a conventional PBX and act as a platform for developing custom telephony applications for delivering dynamic content over a telephone similarly to how one can deliver dynamic content through a web browser using CGI and a web server.

Callweaver talks to a variety of telephony hardware including BRI, PRI, POTS, Bluetooth headsets and IP telephony clients using SIP and IAX protocols protocols

This package is based on the package 'ccrtp' from project 'network:telephony'.

This package is based on the package 'ccrtp' from project 'openSUSE:Factory'.

The ccrtp package offers a generic framework for sending and receiving real-time streaming data over UDP packets using sending and receiving packet queues.

This package is based on the package 'check' from project 'network:telephony'.

This package is based on the package 'check' from project 'openSUSE:Factory'.

Check is a unit test framework for C. It features a simple interface
for defining unit tests, putting little in the way of the developer.
Tests are run in a separate address space, so Check can catch both
assertion failures and code errors that cause segmentation faults or
other signals. The output from unit tests can be used within source
code editors and IDEs.

This package is based on the package 'commoncpp2' from project 'network:telephony'.

This package is based on the package 'commoncpp2' from project 'openSUSE:Factory'.

Common C++ is a GNU package which offers portable "abstraction" of
system services such as threads, networks, and sockets. Common C++
also offers individual frameworks generally useful to developing
portable C++ applications including a object persistance engine, math
libraries, threading, sockets, etc. Common C++ is small, and highly
portable. Common C++ will support most Unix operating systems as well
as Win32, in addition to GNU/Linux.

This package is based on the package 'flite' from project 'network:telephony'.

This package is based on the package 'freePBX' from project 'network:telephony'.

ATTENTION !!!
This rpm could modify your mysql databases.

FreePBX is a Standardised Implementation of Asterisk that gives you a GUI to manage your system. If you’ve looked into Asterisk, you’d know that it doesn’t come with any ‘built in’ programming. You can’t plug a phone into it and make it work without editing configuration files, writing dialplans, and various messing about. FreePBX simplifies this by giving you a pre-written set of dialplans that allow you to have a fully functional PBX pretty much straight away. Some of the features that FreePBX supports out of the box are:

* Unlimited number of Voicemail boxes
* ‘Follow Me’ functionality
* Ring Groups with calls confirmation (so if, eg, a cellphone is out of range and diverts to voicemail, all the other phones keep ringing)
* Unlimitied number of Conferences (limited by available CPU power - about 300 simultaneous users in conferences on a P4 3ghz - 600 with a dual core!)
* Paging and Intercom functionality for SIP and IAX phones that support it (Eg, Snom, Aastra, Grandstream)
* Music on Hold (via MP3s, or streamed off the internet)
* Call Queues
* And many other features

This package is based on the package 'freeradius-client' from project 'network:telephony'.

This package is based on the package 'freeradius-client-snapshot' from project 'network:telephony'.

This package is based on the package 'iaxmodem' from project 'network:telephony'.

This package is based on the package 'iksemel' from project 'network:telephony'.

iksemel is an XML (eXtensible Markup Language) parser library designed for Jabber applications. It is coded in ANSI C for POSIX compatible environments, thus highly portable. It's a free software released under the GNU Lesser General Public License.

This package is based on the package 'ilbc' from project 'network:telephony'.

This package is based on the package 'ilbc' from project 'openSUSE:Factory'.

iLBC (internet Low Bitrate Codec) is a FREE speech codec suitable for robust voice communication over IP. The codec is designed for narrow band speech and results in a payload bit rate of 13.33 kbit/s with an encoding frame length of 30 ms and 15.20 kbps with an encoding length of 20 ms. The iLBC codec enables graceful speech quality degradation in the case of lost frames, which occurs in connection with lost or delayed IP packets.

This package is based on the package 'libconfuse' from project 'network:telephony'.

libConfuse is a configuration file parser library, licensed under the terms of the LGPL, and written in C. It supports sections and (lists of) values (strings, integers, floats, booleans or other sections), as well as some other features (such as single/double-quoted strings, environment variable expansion, functions and nested include statements). It makes it very easy to add configuration file capability to a program using a simple API.

The goal of libConfuse is not to be the configuration file parser library with a gazillion of features. Instead, it aims to be easy to use and quick to integrate with your code. libConfuse was called libcfg before, but was changed to not confuse with other similar libraries.

This package is based on the package 'libedit' from project 'network:telephony'.

libedit is a command line editing and history library. It is designed to be used by interactive programs that allow the user to type commands at a terminal prompt.

This package is based on the package 'libeXosip2' from project 'network:telephony'.

Library that hides the complexity of using the SIP protocol for multimedia
session establishement. This protocol is mainly to be used by VoIP
telephony applications (endpoints or conference server) but might be
also useful for any application that wish to establish sessions like
multiplayer games.

This package is based on the package 'libgsm' from project 'network:telephony'.

This package is based on the package 'libgsm' from project 'openSUSE:Factory'.

Contains libraries and binaries for a GSM speech compressor. libgsm
contains a standard implementation of the European GSM 06.10
provisional standard for full-rate speech transcoding, prI-ETS 300 036,
which uses RPE/LTP (residual pulse excitation/long term prediction)
coding at 13 kbit/s.

GSM 06.10 compresses frames of 160 13-bit samples (8 kHz sampling rate,
which is a frame rate of 50 Hz) into 260 bits. For compatibility with
typical UNIX applications, our implementation turns frames of 160
16-bit linear samples into 33-byte frames (1650 Bytes/s). The quality
of the algorithm is good enough for reliable speaker recognition. Even
music often survives transcoding in recognizable form (given the
bandwidth limitations of 8 kHz sampling rate).

The interfaces offered are a front-end modeled after compress(1) and a
library API. Compression and decompression run faster than real-time
on most SPARC stations. The implementation has been verified against
the ETSI standard test patterns.

This package is based on the package 'libosip2' from project 'network:telephony'.

This package is based on the package 'libosip2' from project 'openSUSE:Factory'.

This is the GNU oSIP library. It has been designed to provide the
Internet community with a simple way to support the Session Initiation
Protocol. SIP is described in the RFC 3261, which is available at
http://www.ietf.org/rfc/rfc3261.txt.

This package is based on the package 'libpri' from project 'network:telephony'.

libpri is a C implementation of the Primary Rate ISDN specification which has been extended to support Basic Rate ISDN as well.

This package is based on the package 'libresample' from project 'network:telephony'.

libresample by Dominic Mazzoni is a real-time library for sampling rate conversion

This package is based on the package 'libzrtpcpp' from project 'network:telephony'.

This package is based on the package 'libzrtpcpp' from project 'openSUSE:Factory'.

This library is a GPL licensed extension to the GNU RTP Stack, ccrtp,
that offers compatibility with Phil Zimmermann's zrtp/Zfone voice
encryption, and which can be directly embedded into telephony
applications.