Yet Another Telephony Engine

Yet Another Telephony Engine (Yate) is an open source, portable, flexible and completely free next-generation telephony engine focused on Voice over Internet Protocol (VoIP). The software is implemented in C++ and can only be used via a command-line interface.

Voice, video, data and instant messaging can all be unified under Yet Another Telephony Engine's flexible routing engine, maximizing communications efficiency and minimizing infrastructure costs for businesses.

It’s a very flexible telephony engine

Yet Another Telephony Engine can be used as a VoIP server, a VoIP client, a SS7 (Signalling System No. 7) switch, a Jabber server, a Jabber client, a VoIP to PSTN (Public Switched Telephone Network) gateway, a H.323 gatekeeper, a SIP router, a Phone2PC and PC2Phone gateway, as well as an IP Telephony client and/or server.

Additionally, you can transform Yate into a conference server that supports up to 200 voice channels in a single conference, an ISDN active and passive recorder, an ISDN (Integrated Services Digital Network) or RSB analog passive recorder, a call center server, an IVR (Interactive Voice Response) engine, as well as postpaid and/or prepaid cards system.

The software can also be used as a H.323 multiple endpoint server, a SIP (Session Initiation Protocol) registration server, a MGCP (Media Gateway Control Protocol) server (also known as call agent), a SIPH.323 proxy, a Jingle server or client, a SIP session border controller, or a IAX (Inter-Asterisk eXchange) client and/or server.

It’s an easily extensible software

While the software is written entirely in the C++ programming language, it supports scripting in various other programming languages, such as Python, Perl or PHP. It’s an easily extensible software, which means that developers can use the integrated Perl, PHP and Python libraries to build external functionalities for Yet Another Telephony Engine (Yate).

What is new in this release:

• Better Javascript support, and adds libygsm for the YateBTS project.
• More stable SIP support.

What is new in version 5.0.0:

• Supports IPv6 for LTE
• Enhanced JavaScript functionality

What is new in version 4.3.0:

• This version adds XML support in Javascript, SCCP-GTT routing between different networks, and stability improvements.

What is new in version 4.2.0:

• This version adds SIP flood protection and improves Jingle / Google Voice support.
• The Javascript is usable and the SIGTRAN links instability was solved.

What is new in version 4.1.0:

• This version brings a GVoice module for sending DTMF at answer time, an iSAC codec, and support for new Wanpipe drivers.
• Minor fixes: T.38 timings and client issues on the Mac.

What is new in version 4.0.0:

• Added features:
• SS7 components for mobile operators - SCCP, TCAP, MAP and CAMEL
• Javascript new implementation for fast prototyping of telephony applications.
• LNP over ANSI SS7
• T.38 fax support over MGCP digital gateways
• TCP and TLS transports for SIP, multiple listening interfaces
• Alerts, monitoring and logging.
• YateClient got a brand new face
• Minor improvements:
• Better debugging informations
• Atomic operations used if architecture and compiler support them

What is new in version 3.3:

• Support for GMail chat conference was added.
• Minor fixes were done in SS7 M2PA and ANSI.
• Fixes were also made in H.323, SIP, and RTP.

What is new in version 3.2:

• This version adds support for CNAM/LNP lookup by SIP INVITE/3xx.
• It has bugfixes in SIGTRAN/MGCP/SS7.

What is new in version 3.1.0:

• Telephony calls using Google Voice as service
• Chat room conversations are archived
• Tones can be customized form config file, sample file includes 40 countries
• Minor features added to SS7 ISUP, STP and M2PA
• Ringback can be provided from remote MGCP gateways
• Some infrequent memory leaks were fixed