Wammu is a mobile phone manager that uses Gammu as its backend. It works with any phone that Gammu supports, including many models from Nokia, Siemens, and Alcatel. It has complete support (read, edit, delete, copy) for contacts, todo, and calendar. It can read, save, and send SMS. It includes an SMS composer for multi-part SMS messages, and it can display SMS messages that include pictures. Currently, only text and predefined bitmaps or sounds can be edited in the SMS composer. It can export messages to an IMAP4 server (or other email storage).
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oRTP is a library implementing the Real-time Transport Protocol (RFC3550), written in C. It is easy to use and provides a packet scheduler for sending and receiving packets on time, adaptive jitter compensation, automatic sending of RTCP compound packets, and the RTCP parser API. It works with IPv6.
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PHP Voice (formerly known as PHP VXML) contain four classes that assist in developing voice application using PHP. It supports Speech Synthesis Markup Language 1.0, Speech Recognition Grammar Specification 1.0, Voice Browser Call Control: CCXML 1.0, and Voice Extensible Markup Language (VoiceXML) 2.0.
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Ip phone Scanning Made Easy (ISME) scans a VOIP environment, adapts to enterprise VOIP, and exploits the possibilities of being connected directly to an IP Phone VLAN. It seeks to get the phone's configuration file directly from a TFTP server, enable SIP/SIPS (TCP/UDP), communicate with an embedded Web server and Web server banner, identify the editor by MAC address, and identify potential default login/password combinations which should be changed.
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The Alamin GSM SMS Gateway is a group of daemons that allows you to send/receive SMS messages from any GSM device that supports AT+ commands (GSM modems or GSM mobile phones) or supported by Gnokii. A client program allows you to send messages from any IP client. An SMTP interface is provided to allow MTAs to send SMS directly to the GSM network. IMP (Incoming Message Processor) modules allows you to extend functionality to implement banking, network administration, bd querys, etc. from a GSM mobile phone.
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Linuxsms is a Perl script to send SMS to GSM phones. There is support for multiple servers/gateways (some servers require registration). The script has an adressbook, the ability to compress SMS, and the ability to check for new versions.
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VoiceOne is a Linux distribution that gives you the ability to install a PBX platform with an easy to use Web-based GUI. It also provides a framework for building a communication server adding various plugins. Its main features are Asterisk 1.8 with realtime configuration with MySQL, a Ubuntu 10.04 base, and support for both hard disk and Compact Flash card installation.
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SER or SIP Express Router is a very fast and flexible SIP (RFC3261) server. It can handle thousands of calls per second on low-budget hadware. A C shell-like scripting language provides full control over the server's behavior. Its modular architecture allows only required functionality to be loaded. The following modules are available: accounting, digest authentication, CPL scripts, ENUM support, instant messaging, MySQL support, PostgreSQL support, a presence agent, Radius authentication and accounting, Diameter authentication, record routing, an SMS gateway, a Jabber gateway, NAT traversal support transaction module, a registrar, and user location.
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GNU SIP Witch is a secure peer-to-peer VoIP server. Calls can be made even behind NAT firewalls, and without requiring service providers. SIP Witch can be used on the desktop to create bottom-up secure calling networks and as a free software alternative to Skype. It can also be used as a stand-alone SIP-based office telephone server, or to create secure VoIP networks for an existing IP-PBX such as Asterisk, FreeSWITCH, or Yate.
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libre is a generic library for real-time communications with asynchronous I/O support. It is written in portable POSIX source code that conforms to the ANSI C89 and ISO C99 standards. It is robust and fast, with a low memory footprint. It also features RFC compliance and support for IPv4 and IPv6. Protocol implementations include SIP, SDP, RTP/RTCP, BFCP, DNS, STUN/TURN/ICE, HTTP, and WebSockets.
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Real time communication software built to provide face-to-face advantages to remote gamers.
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Asterisell is a Web application for rating, showing to customers, and billing of Asterisk VoIP calls.
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letterize generates alphabetic mnemonics for a phone number, then
filters them for phonetic plausibility in English.
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1VideoConverence is Web2.0 audio-video conference call software for Asterisk with support for Web, phone, MSN, Skype, Yahoo, and Jabber clients. This VoIP and VVoIP conferencing app for business, government, education, and health care is based on C#, WinFX, XAML, and .NET 3.0.
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SEMS is a media and application server for SIP based VoIP services. It shows good performance doing basic services like announcements and conference for combination with external application servers. Thanks to its easy-to-use and flexible application development framework and back-to-back user agent support, application logic and media serving can be combined in the same process. Basic applications like announcement, pre-call announcement, RBT, conference, voicemail, mailbox, and lots of example applications are available. Scripting can be done in Python and a simple state machine description language. Support All commonly used free codecs (including g711, gsm, iLBC, speex, adpcm, and l16) are supported. Other features include wideband, ZRTP encryption, a SIP registrar client, an XMLRPC server/client, and a DIAMETER client.