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Yate Changelog

What's new in Yate 6.0.0-1

Sep 15, 2017
  • Yate now includes a Radio device API and a bladeRF module with automatic frequency calibration
  • A lot of small fixes and improvements regarding behavior of SIP
  • Additions and many minor fixes in the Javascript implementation
  • Configuration files now support inclusion
  • Improvements in logging, default levels and interactive use
  • Performance improvements and performance measurements
  • Better support for creating applications based on Yate

New in Yate 5.5.0-1 (Jul 11, 2016)

  • SQLite database
  • Support for YateBTS 5.0
  • Stability improvements

New in Yate 5.0.0-1 (Oct 24, 2013)

  • Support IPv6 for LTE
  • Enhanced JavaScript functionality

New in Yate 4.2.0 (Oct 24, 2013)

  • Added:
  • Flood protection in the SIP channel
  • iLBC codes based on fixed point WebRTC library
  • Proper build on MIPS 64-bit targets
  • Fixes:
  • Various changes to improve Jingle and Google Voice support
  • Javascript is now usable in routing scripts
  • SIGTRAN links failures fixed by always setting the SCTP socket timers

New in Yate 4.1.0 (Oct 24, 2013)

  • Added:
  • iSAC codec
  • better Gvoice support
  • support for new Wanpipe drivers.
  • Fixes:
  • T.38
  • Mac client issues.

New in Yate 4.0.0 (Feb 29, 2012)

  • 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

New in Yate 3.3.2 (Jan 30, 2012)

  • Fixes Jingle calls to Google Voice dropping at 5 minutes
  • When RTP forward is used SIP reINVITES are sent as MDCX on MGCP circuits

New in Yate 3.3.0 (Apr 5, 2011)

  • Support for GMail private chat conference rooms
  • Internal microphones work on MacOS, now compiles on Mac OS X 10.6 too
  • Handle some ANSI specific ISUP procedures and parameters
  • Better detection of SS7 links misconfiguration
  • Major stability issue fixed in H.323
  • Minor stability fixes in SIP and RTP

New in Yate 3.2.0 (Mar 17, 2011)

  • Look up CNAM and LNP using SIP INVITE / 3xx redirect mechanism
  • Fixes in error handling of M2PA (SIGTRAN), ISUP and MGCP
  • Support for some ANSI specific features in SS7 MTP and ISUP
  • Speed up loading of large rosters in the Jabber client
  • Extended default timeouts when connecting to Jabber servers

New in Yate 3.1.0 (Feb 3, 2011)

  • 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

New in Yate 3.0.0 (Feb 3, 2011)

  • Client with full Jabber IM and voice capabilities
  • Built-in Jabber server
  • Built-in SNMP agent for querying the server state
  • CPU load evaluation and thresholds
  • SIP domains support
  • Wideband audio support
  • XML library using Yate classes
  • RTP stats, sends RTCP reports
  • SSL support for remote manager connections
  • Linux kernel SCTP support for SIGTRAN
  • SIGTRAN M2PA, M2UA and IUA implementation
  • SS7 MTP supports transfer function (STP mode)
  • SS7 ISUP corrections and improvments
  • SS7 passes ITU certification tests for MTP3 (SP or STP) and ISUP

New in Yate 2.2.0 (Apr 30, 2010)

  • This is mainly a maintenance release including stability and performance fixes
  • The SIP parser performance was improved 2.5x
  • Fix for 100% CPU load on client when Qt use the glib event loop
  • Support for T.38 terminal mode fax in SIP

New in Yate 2.1.0 (Apr 30, 2010)

  • An implementation of the Cisco SLT and RUDP protocols allows Yate to control both signaling and voice on Cisco 5350 and 54xx XM universal gateway series
  • Fixes in handling of PSTN circuits over MGCP

New in Yate 2.0.0 (Apr 30, 2010)

  • This release includes many small improvments and bug fixes
  • Several border conditions in SIP are handled better by default and more of the SIP handling is configurable, including support for subscribing to mailbox and line status notifications
  • The PBX is able to store and copy any message parameter
  • Matching in the regular expressions module can be performed on function result and messages can be renamed or duplicated easily
  • More initial startup time is allowed by the supervisor as well as properly reaping all zombie child processes that may be left by external scripts
  • In client the ENTER key now works as expected by starting a call and adding to the callto combobox
  • The event window can be used to display debugging information

New in Yate 1.3.0 (Apr 30, 2010)

  • This release includes many small improvments and bug fixes.
  • Several border conditions in SIP are handled better by default and more of the SIP handling is configurable, including support for subscribing to mailbox and line status notifications
  • The PBX is able to store and copy any message parameter
  • Matching in the regular expressions module can be performed on function result and messages can be renamed or duplicated easily
  • More initial startup time is allowed by the supervisor as well as properly reaping all zombie child processes that may be left by external scripts
  • In client the ENTER key now works as expected by starting a call and adding to the callto combobox
  • The event window can be used to display debugging information

New in Yate 1.2.0 (Apr 30, 2010)

  • A Jingle channel suitable for implementing gateways is now available together with some STUN support for the RTP data
  • The PBX was improved a lot and quite complex operations can be described from the config file
  • Lots of minor improvments and bug fixes were made in the VoIP protocol drivers
  • Considerable improvements of the call forking module allow it to be used for
  • special purposes like generating fake early media
  • A bug that could be used to crash a Yate process (but no way to take control) over the SIP protocol was fixed

New in Yate 1.1.0 (Apr 30, 2010)

  • Fax channel was almost entirely rewritten to work with a more recent version of spandsp. A tone detector module allows detecting fax tones and diverting the call
  • The database module can provide several fallback routes
  • A few TCP sockets are reusable so there is no problem quickly stopping
  • and restarting Yate
  • A minor bug causing temporary noise in conference was fixed, also bugs in timestamp generation and RTP data splitting
  • Support for external Python scripts was considerably improved
  • The client got an event log window