Main • F.A.Q. - Frequently Asked Questions

Q.1.: What does Yet Another Telephony Engine mean?

A.1.: Most of the telephony technologies (ISDN,H.323,SIP) are point to point. This means that there is no big difference between a server and a client. A "telephony engine" tries to give you support to let you use it as a client or as a server (or proxy) in the way that is suitable for your enviroment.

Q.2.: Why is Yate written in C++ ?

A.2.: C++ is a superset of C, so we get the flexibility and power of C together with the OO (Object Oriented) support of C++. We need C++ OO technology mainly because otherwise Yate can become a difficult to maintain project. A nice explanation about C++ can be found at http://www.cplusplus.com/info/description.html and reasonable one at http://www.star.bnl.gov/public/comp/train/WhyC++.txt

Q.3.: From where comes "We don't dereference NULL"?

A.3.: This sentence comes from before Yate, and we hope it defines the Null Team. Anyway it means that we prefer to write good, well designed code so then we can spend more time on developing new features than fixing bugs in already written code. And we take bugs very seriously, trying to Honour the bug

Q.4.: What does Yate have to do with Asterisk?

A.4.: Yate is not an Asterisk fork; in fact, the original idea for Yate comes mostly from Bayonne.

Q.5.: What kind of SIP support does Yate have?

A.5.: The Yate developers have explored several different open source SIP libraries but found them lacking, so Yate uses its own SIP stack, which may be reusable in a single or multi thread module for applications like clients, servers, or proxies. Handling of custom SIP messages can be implemented in internal or external (script) modules.

Q.6.: Why has the Yate project forked the IAX library? What makes the fork worth all the trouble usually associated with such a move?

A.6.: We like to contribute to other free software projects. For IAX2 the solution was to use either libiax2 (a client library) or to write everything from scratch.
We got libiax2 but the original copyright holder requires a disclaimer to add our patch, and we didn't give the disclaimer.
Our libiax2 have a few bugs fixed like in the way it is dealing with codecs, and it can be used for an iax2 server (like Yate).
In libyateiax we have support for trunking which is important to reduce bandwidth usage between servers.

Q.7.: Why is Yate sending a CANCEL and a BYE to tear down a not yet answered SIP call?

A.7.: To proactively prevent problems with the call being answered just before the other side receives the CANCEL. Either the CANCEL or (almost always) the BYE will generate a harmless 481 answer.

If you don't find your question answered here, please check our mailing archive to see whether your issue was addressed before.

You can always contact us, we are looking forward to it.

July 2014:
Yate 5.4 and YateBTS 4 launched. Added JSON and DNS support in Javascript, Handover support in YateBTS.

March 2014:
YateBTS 2.0 launched. Added authentication and WebGUI. Added USSD support in commercial version.

March 2014:
Yate 5.2 launched. Better JavaScript support and a fixed memory leak.

Jan 2014:
YateBTS 1.0 launched. The first GSM Basestation which works with an IMS/VoLTE core network.

Jan 2014:
Yate 5.1 launched. Better JavaScript support and added libygsm. Elisa chatbot added in RManager

Oct 2013:
OpenHSS is the Yate based HLR/HSS solution for MVNO and LTE carriers.

Oct 2013:
Yate 5 released. Added IPv6 support in SIP for LTE. Improved JavaScript support. Download NOW

Jan 2013:
Yate 4.3 released: Added XML support in Javascript. SCCP - GTT routing between different networks. Stability improvements.
Download NOW

Aug 2012:
Yate 4.2 released: SIP flood protection. Better Jabber/Google Voice support. Usable Javascript. Fixed SIGTRAN links fluctuations.
Download NOW

Apr 2012:
YateClient was accepted in the Mac Store.

Yate 4.1 released: better Gvoice support, iSAC codec, support for new Wanpipe drivers. Fixes T.38 and Mac client issues.

Mar 2012:
SS7Cloud is launched today, 1st March, 2012, by NullTeam, Yate creators. Having all you need to be a US CLEC, it brings SS7 services in a cloud.

Feb 2012:
Yate 4.0 released.
SCCP, TCAP, MAP and CAMEL, TCP and TLS in SIP, Javascript fast prototyping of telephony applications and brand new face for YateClient.

Nov 2011:
Here is a video that, quote "demonstrates the truly awesome power of the YATE engine, as it easily handles 3 simultaneous calls to an audio player application including dtmf (button press) handling "(from PaintedRockComm).

Nov 2011:
Yate will attend ORR - OPENRHEINRUHR (November 12 - 13).

04 May 2011:
sipgate chooses open source project Yate for core infrastructure.

12 Apr 2011:
Yate 3.3.2 released.
Fix for Jingle calls to Google Voice dropping after 5 minutes.
4 Apr 2011:
Yate 3.3 released.
Support for GMail chat conference, fixes for internal microphone in MacOS. Minor fixes in SS7 M2PA and ANSI. Fixes in H.323, SIP and RTP.

9 Mar 2011:
Yate 3.2 released.
Bug fixes in SIGTRAN/MGCP/SS7 and added support for CNAM/LNP lookup by SIP INVITE/3xx.

Feb 2011:
Yate will attend FOSDEM and XMPP summit.

31 Jan 2011:
Yate 3.1 released.
Yate client support for Google Voice. Support for any country tones in tonegen.

20 Dec 2010:
Yate 3.0 released.
SS7 ITU certified. SS7 STP added. Client supports Jabber IM (Google Talk + Facebook).

3 May 2010:
Yate 3.0.0 alpha 3 released. Featuring the new Jabber server and wideband audio.

8 March 2010:
Yate 2.2 released. Mostly bug fixes. Dahdi compatible. Latest 2 release before 3.0.

6-7 February 2010:
Yate booth at FOSDEM 2010. Free CD with Freesentral available.

2 Nov 2009:
Yate 2.1 launched. Can replace a Cisco PGW2200 to control a Cisco AS54xx.

6 Aug 2008:
Yate and OpenSIPS (former OpenSER) join to build IP based clusters.

4 Aug 2008:
Yate 2 launched.

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