Kamailio (formerly OpenSER) is a high-performance SIP (RFC3261) server with a flexible architecture and many extensions. The server implements proxy, registrar, redirect, and location SIP/VoIP services. It has support for UDP, TCP, TLS, and SCTP transport layers, DNSsec, ENUM, AAA via database, RADIUS, DIAMETER, gateways to SMS and XMPP, least cost routing, load balancing, NAT traversal, and call processing language. Kamailio implements SIMPLE presence and instant messaging extensions, and includes an embedded XCAP server and MSRP relay. It can be also used as a routing SIP sever for WebRTC via WebSocket.
WombatDialer is a platform to provide mass outbound calling. This can be used to implement many different services. By offering a set of ready-to-use components and a monitoring GUI, it lets you create complex solution in minutes. It can work on predefined call lists or can dynamically create them over an API (e.g., dial number X after 10:30AM). It shares the load on one or more PBX servers and has flexible rescheduling logic to handle missed calls. It is built to be used with your existing Asterisk PBX, and does not require separate servers or a separate set of lines. It can call over VoIP or through the public telephone network. It is built to integrate with your business processes, and can receive calls to be made over HTTP and/or notify an external system in realtime of calls made and results gathered. It works natively with the QueueMetrics Call-Center Monitoring Suite to produce state-of-the-art campaign analyses and insight.
Yate is a next-generation telephony engine. While currently focused on Voice over Internet Protocol (VoIP) and PSTN, its power lies in its ability to be easily extended. It supports SIP, H.323, IAX, MGCP, Jingle, Jabber, E1, T1, analogic, robbed bit, ISDN PRI, BRI, and SS7. YateClient is an Instant Messenger and Voice application for Jabber/Jingle IM, SIP, H.323, and IAX VoIP protocols.
WinTariff is a program that collects and processes information about telephone calls. It receives data from an office PBX about the date/time, duration, and dialled number of phone calls, then calculates the cost of the call and determines the direction. Various reports about all calls during the month, calls to a certain internal subscriber, etc. can be generated. A data collection module is available for Linux/FreeBSD.
The GNU Gatekeeper is a free H.323 gatekeeper based on the OpenH323 project. You can use it to manage a Voice-over-IP network and let endpoints (e.g., Netmeeting) communicate through symbolic names. It also has an external interface for billing and other applications. It runs on a number of Unix versions (including Linux and Solaris) and Windows.
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.
Oreka is an enterprise telephony recording and retrieval system with a Web-based user interface. It supports recording voice from VoIP SIP, Cisco Skinny (aka SCCP), raw RTP, and audio sound devices, and runs on multiple operating systems and database systems. It can record audio from most PBX and telephony systems, such as BroadWorks, Metaswitch, Asterisk, FreeSwitch, OpenSIPS, Avaya, Nortel, Mitel, Siemens, Cisco Call Manager, Cosmocom, NEC, etc. It is being used in call centers and contact centers for quality monitoring (QM) purposes.