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.
4PSA VoipNow is PBX software for service providers and enterprises. Easy to use and deploy, it can run on x86 or PowerPC based servers and delivers advanced IP telephony services. It can be deployed in various working scenarios to provide hosted VoIP services in high availability xSP environments, advanced call center services, or enterprise telephony in organizations of all sizes. A free Express version is available.
The REMWAVE Carrier Core platform is a stable platform to set up a voice over IP carrier (ITSP). It contains a complete billing, rating, and routing engine and is used by some of the largest ITSPs. The Carrier Core is free for deployment and contains a billing engine (based on Oracle express 10g edition, though larger editions are available), a SIP proxy and registrar, a customer care system, and templates to set up a customer self service portal.
DialFOX is an express dialplan report generator that is used with the Asterisk PBX system. It is able to make an inventory of any device (such as SIP phones, softphones, and ATA) that is active in a local network. It lists their extensions, IP address, username, caller queue, device info, and comments. It can easily access any SIP device that is found in the LAN. Furthermore, DialFOX provides additional information about phone devices like firmware release, key functions, and more.
Noojee Receptionist for Asterisk provides a software-based interface for receptionists. It provides the ability to dispatch calls using a database of all staff and external parties or by phone or extension number. When looking up internal extensions, the status of each extension is displayed, saving the receptionist time as they no longer have to dial the number to determine if the party is available. When an extension is busy, the call can be placed into a "hold and transfer" mode that will automatically transfer the caller when the extension becomes available. It can also periodically inform the caller that the party is busy and gives the option to continue waiting, leave a voice message, or be put back through to the receptionist.
The SIP Switch is a Web application which allows you to use multiple SIP accounts on the same phone. This lets you use the call plans of different VoIP companies (using SIP) with one IP phone device. It does the third party SIP registration so you can receive all your calls on the sipswitch account. On dial out, you can set prefixes to make your outgoing calls with such or such provider.