Free voip software for linux

PlayVoIP is a webbased VoIP service enabler featured with user management, statistical report and registration application system based on LAMP and Asterisk.

At the beginning this project was specially designed for simple and free Indonesian VoIP network, VoIP Rakyat.

Tapioca is a framework for Voice over IP (VoIP) and Instant Messaging (IM). Its main goal is to provide an easy way for developing and using VoIP and IM services in any kind of application.
Tapioca VoIP project was designed to be cross-platform, lightweight, thread-safe, having mobile devices and applications in mind.
Main features:
- Create a solution that integrates all components used by VoIP and IM applications in a single, reliable and easy to use framework, which is able to work on different platforms.
- Spare resources, providing central services for multiple applications. Eg.: The control of all incoming and outgoing SIP requests are managed by the SIP service, avoiding the creation of one SIP stack and allocation of a network port for each SIP-based application.
- Reduce the overhead of control layers and library dependencies.

AdminsParadise VoIP PBX is a full Web-based phone and fax solution that integrates the best-of-breed open source VoIP software.
It runs Asterisk 1.4.1, hylafax, avantfax, and PHP5 with a themed, easy-to-use, Web-based interface.
Main features:
- Asterisk 1.4.1
- HylaFax
- AvantFax
- Scheduled Conferencing
- Integrated authentication
Enhancements:
- This initial release provides a live CD and an installation CD.

Here you will find tutorials, how-to guides, and instructions for setting up an enterprise network using free open-source software.

The primary goal is to show you how to setup your entire network using open-source solutions and save you a TON of money doing so.

And when we said entire network, we meant EVERYTHING including domain controlers, phone systems, email servers, file servers, backup servers, self-monitoring security systems...EVERYTHING!

We will provide guides that have as many screenshots, illustrations, or step-by-step "watch the movie" files as necessary to make the process as easy and clear as possible.
You can use the guides to complement your existing network or to replace existing components with more robust and less expensive solutions.

You will find that well step you through setting up a network that is enterprise grade. Well even show you how to inexpensively cluster your servers so downtime is something you dont even worry about anymore. And best of all the solutions will cost you practically nothing.

MD5: 6e098846a2c1265ba8b0a7d67ee7b5dc livecd.iso

AdminsParadise VoIP PBX is a full Web-based phone and fax solution that integrates the best-of-breed open source VoIP software.
The project runs Asterisk 1.4.2, hylafax, avantfax, and PHP5 with a themed, easy-to-use, Web-based interface.
Enhancements:
- Upgrades to Freepbx 2.2 and updates modules to the 2.2 level

VoiceBuntu (Ubunterisk) is an Ubuntu based live Linux distribution that uses Asterisk and VoiceOne to provide VoIP service.
VoiceBuntus focus is to run asterisk with no installation needed. VoiceBuntu - ubuntu itself - has built in boot prompt cheap code one can use called persistent.
This feature allows the user to use VoiceBuntus persistent feature in order to keep settings on a flash disk or memory stick.
To start, just burn the ISO file on a CD-ROM then start the live cd distribution and wait until everything is ready.
Please note, that apache will start at the very last moment of the boot cycle. So you may a to wait 1 or 2 minutes to let everything load.
Once everything is prepared, start your browser and point to the IP address of your running VoiceBuntu (i.e. http://192.168.0.10).
If you have no DHCP Server available you can press F6 at boot screen and add additionally add the command parameter
static=xxx.xxx.xxx.xxx
netmask=xxx.xxx.xxx.xxx
bcast=xxx.xxx.xxx.xxx
gateway=xxx.xxx.xxx.xxx
dns1=xxx.xxx.xxx.xxx
dns2=xxx.xxx.xxx.xxx
where xxx.xxx.xxx.xxx is the IP of your choice.
By that VoiceBuntu boots with static IP address on eth0 network device.
Enhancements:
- This live CD release is based on Ubuntu 7.04 and VoiceOne 0.5.0_2 due to a minor bug in VoiceOne version 0.5.0.

Sofia-SIP is an open-source SIP User-Agent library, compliant with the IETF RFC3261 specification.
Sofia-SIP project can be used as a building block for SIP client software for uses such as VoIP, IM, and many other real-time and person-to-person communication services.
The primary target platform for Sofia-SIP is GNU/Linux. Sofia-SIP is based on a SIP stack developed at the Nokia Research Center. Sofia-SIP is licensed under the LGPL.
Main features:
SIP features
- Sofia-SIP implementation follows RFC3261 and related key RFCs. INFO, UPDATE and REFER methods are supported. Also supported is SIMPLE presence and instant messaging, with the MESSAGE, SUBSCRIBE/NOTIFY and PUBLISH methods. Features such as early sessions, provisional responses, early media, caller preferences and session timers are included. Full set of transports, including both TCP and UDP over either IPv4 or IPv6, are supported.
SIP Offer-Answer module
- Sofia-SIP provides an implementation of the SDP offer-answer negotiation as specified in RFC3264. This is an essential component in using SIP to establish media sessions such as VoIP and video conferencing.
NAT traversal support
- Support for STUN as specified in RFC3489. STUN functionality is available via a separate module, so it can also be used independently from the base SIP stack. SIP extensions such as symmetric response routing (RFC3581/rport) are supported as well.
SIP security support
- Signaling can be secured by use of SSL/TLS. Also HTTP basic and digest authentication methods are supported.

vpnd provides a Virtual Private Network Daemon - encrypted TCP/IP.
vpnd is a daemon which connects two networks on network level either via TCP/IP or a (virtual) leased line attached to a serial interface.
All data transfered between the two networks are encrypted using the unpatented free Blowfish encryption algorithm with a key length of up to 576 bits (may be downgraded to a minimum of 0 bits to suit any legal restrictions).
vpnd is not intended as a replacement of existing secured communications software like ssh or tunneling facilities of the operating system.
It is, however, intended as a means of securing transparent network interconnection across potentially insecure channels.
vpnd acquires a pseudo terminal (a pty/tty device pair) and attaches a SLIP line discipline to it. The effect of this is that vpnd now has its own network interface, a SLIP interface which is named slx where x is some number.
All IP packets sent to this interface are read as a datastream by vpnd and the datastream written by vpnd reappears as IP packets on this interface.
vpnd now encrypts the datastream read and sends it through a TCP connection or over a serial line to its peer vpnd. The datastream received by vpnd from its peer is decrypted and then written to the pseudo terminal.
As vpnd doesnt parse the datastream from the pseudo terminal all packets written by the kernel to the SLIP interface get transported.
Thus vpnd tunnels network traffic between two systems even as it is a user level daemon.
Enhancements:
- fixed minor bug in generic whitening code
- fixed ppp mru setup on Linux
- port to x86_64
- added packetize option for slip/ppp interoperability and rtp header compression (SIP VoIP)
- added smallrtp option for forced use of simple checksum for rtp (SIP VoIP) packets in packetize mode for reduced bandwidth requirements