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Resources -> Compatibility

Mobile Phone

ProductYate VersionProtocolCommentTester
Nokia N80cvs 08.02.2007SIPOKdiana@voip.null.ro
WM6 smartphone11.3SIPOKvmarco@hotmail.com
Nokia N812.0 releaseSIPOK2paulc@voip.null.ro
Nokia E712.0 releaseSIPOK2paulc@voip.null.ro
Nokia E722.1 releaseSIPOK3paulc@voip.null.ro

  1. I tried on HTC S730 but voip-client is the same for all WM6.0 smartphone
  2. Nokia SIP VoIP Sett. is required to make the setting for keeping a NAT hole open via CRLF packets. Auth Realm on the phone must be set to match the server's (default Yate)
  3. Needs Nokia SIP VoIP Sett. version 2.0 (Nokia Forum login required) for settings and to be able to make any call since the phone does not have a built-in VoIP dialer interface. You must not download the newest version as it doesn't include the dialer. The user interface is much worse than E71.

Softphone

A Softphone is software you can run on your computer.

ProductYate VersionProtocolCommentTester
HolaSP1.0SIPOKcristian.seva@voiplab.com
VoipPhone1.0H323OKcristian.seva@voiplab.com
OhPhone0.8H.323OKdiana@voip.null.ro
OpenPhone0.8H.323OKdiana@voip.null.ro
HolaPhone3.0??H323OKcristian.seva@voiplab.com
Bria2.2.3SIPOKsiudak@h-e-s.de
Sipek20.3.xSIPOKraffaele.p.guidi @ g m a i l . c o m
X-Lite4.1SIPOK1marian@null.ro
  1. In Preferences -> Audio Codecs. Make sure the 'Accept the first codec offered ...' check box is checked to make sure calls from yate to X-Lite will have audio from X-Lite. If not checked, X-Lite will send audio using the first enabled codec ignoring the order of negotiated codecs

IP Phone

An IP Phone is a device you plug into your network which principally does VoIP.

ProductYate VersionProtocolCommentTester
AddPac IP series (200, 300)1.0SIPOKramscanner(at)gmail.com
PA1688 based ip-phones (like a FreeSet)1.0SIPOKramscanner(at)gmail.com
Grandstream Budgetone 1000.9 cvsSIPOKmmenaz att mail dot com
AT-3200.9SIPOKdavid at iaxtalk.com
ZyXEL P-2000W0.9.0pre3SIPOK1paulc@voip.null.ro
ACTel P104 (v02.08.14)0.8 cvsSIPOKaaractingi@libertysurf.fr
Dlink PAP22.2v??SIPOKcristian.seva@voiplab.com
T-COM TC 3001.2.0SIPOK2paulc@voip.null.ro
SNOM 3001.3.0SIPOK2marc[at]electronics-design.nl
Nokia E511.3.0SIPOK1marc[at]electronics-design.nl
Perfectone IP-300, IP-5001.3.0SIPOKmarc[at]electronics-design.nl
SNOM 3601.3.0SIPOKpolsakiewicz[at]gmail.com
Thomson ST20301.3.0SIPOKpolsakiewicz[at]gmail.com
SNOM 3202.0.0SIPOKpolsakiewicz[at]gmail.com
SNOM 8202.0.0SIPOKpolsakiewicz[at]gmail.com
SNOM 3702.0.0SIPOKpolsakiewicz[at]gmail.com
SNOM M32.0.0SIPOKpolsakiewicz[at]gmail.com

  1. Tested over 802.11b
  2. Phone is NAT aware, RPORT used, STUN not needed

ATA (Analog Telephone Adaptor) or FXS devices

These devices allow you to connect your existing analog handset into a VoIP environment.

ProductYate VersionProtocolCommentTester
AG-1680.9SIP (1 FXS port)OKdavid at iaxtalk.com
AG-4680.9SIP (4 FXS ports)OKdavid at iaxtalk.com
D-link 14020.9SIP (2 FXS ports)OKanet@uzpak.uz
Telco Systems 232 (v4.45)0.8 cvsSIP (2 FXS ports)OKaaractingi@libertysurf.fr
Welltech 1501 (v1afxs.202)0.8 cvsSIP (1 FXS port)OKaaractingi@libertysurf.fr
Telsey CPVA500-SIP (2 FXS ports)OKvmarco@hotmail.com
Cisco 827-4V-H323 (4 FXS ports)OKvmarco@hotmail.com
Cisco 827-4V-SIP (4 FXS ports)OK1vmarco@hotmail.com
IP fone 17101.2.0H323 (1 FXS port)OK2paulc@voip.null.ro
Yoda VS100 1.3.0SIPOKmarc[at]electronics-design.nl
Yoda VS211 1.3.0SIPOK3marc[at]electronics-design.nl
Perfectone GW2111.3.0SIPOK3marc[at]electronics-design.nl
Perfectone GW2001.3.0SIPOKmarc[at]electronics-design.nl
Perfectone GW2201.3.0SIPOKmarc[at]electronics-design.nl
tiptel cyberBOX 22 SIP1.3.0SIPBroken Codecmarc[at]electronics-design.nl
Grandstream GXW-40243.0.0SIP (24FXS)OKvpol[at]vpol.org.ru
  1. With IOS Version 12.3(11)T5
  2. Early media must not be signalled to the ATA else voice will fail
  3. DailerID from PSTN not put through to connected phone.

Gateway

A gateway is a device which connects either to ISDN (PRI/BRA) or an analog (FXO) line. Typically a gateway has no handset attached (i.e. no mechanism for a user to place or receive calls directly).

ProductYate VersionProtocolCommentTester
AudioCodes Mediant2000 (v4.2)0.8 cvsH.323 (16 PRI)Not OK 1aaractingi@libertysurf.fr
Cisco 3640 (IOS 12.2(24))0.8 cvsH.323 (2 PRI)OKaaractingi@libertysurf.fr
Innovaphone IP400 (v5.01)0.8 cvsH.323 (2BRI)OKaaractingi@libertysurf.fr
Innovaphone IP3000 (v5.01)0.8 cvsH.323 (1PRI)OKaaractingi@libertysurf.fr
Cisco 7200 (IOS 12.2(24a))1.1 cvsH.323OKgustavo.espeche@holaphone.com
Cisco vg200 (IOS 12.2(24a))1.1 cvsH.323OKgustavo.espeche@holaphone.com
2n stargate1.1 cvsH.323OKgustavo.espeche@holaphone.com
Topex multiACCESS1.1 cvsH.323/SIPOKgustavo.espeche@holaphone.com
  1. Need FastStart - didn't test yet

Uncategorised

ProductYate VersionTypeProtocolCommentTester
AddPac AP series (100, 190, 200, 1000, 1002, 1005, 1100, 2460)1.0Gateway FXS-FXOSIPOKramscanner(at)gmail.com
AddPac VP 3001.0VideophoneSIPOK, but w/o videoramscanner(at)gmail.com
Cisco AS54001.0GatewayH323OK with IOS Version 12.3(13b)ramscanner(at)gmail.com
Samsung IPOffice 40001.0IP and traditional SOHO PBXSIPOKramscanner(at)gmail.com
Planet VIP-4500.9 cvsSIP GatewaySIPOKfede@datahost.com.ar
AlterPSS0.9PSSsipOKanet@uzpak.
Avaya PBX0.9PBXH.323OKIftikharQureshi@yahoo.com
Vizufon CIP4500 (v1.2)0.8 cvsVideophoneH.323 and SIPOK (G723 only)aaractingi@libertysurf.fr
Asterisk (v1.0)0.8 cvsVoicemailH.323 and SIPOKaaractingi@libertysurf.fr
GNUGK0.8GatekeeperH323OKkutluturk@yahoo.com
Mediatrix 1102/11040.8SIP GatewaySIPOKkutluturk@yahoo.com
Alcatel OMNIPCX R3 and R4-PBX (gatekeeper)H323OKvmarco@hotmail.com
Alcatel OMNIPCX R3 and R4-PBX (gateway)H323OK (registered on yate without password)vmarco@hotmail.com
Cisco 7905SIPVoIP phoneSIPOKdiana@voip.null.ro
Cisco VG200-GatewayH323OKcristian.seva@voiplab.com
GNUGK2.2.4GatekeeperH323OKcristian.seva@voiplab.com
IVR HolaPhone2.0IVRH323/SIPOKcristian.seva@voiplab.com
Netsynt VRC-Gateway (FXS,PRA,BRA )H323OKvmarco@hotmail.com
Netsynt Slimcase-Gateway(FXS,PRA,BRA)H323OKvmarco@hotmail.com
Aastra Ascotel2.0PBXSIPOKpeter[at]visionutv.se
3COM NBX V30002.0PBXSIPOKpeter[at]visionutv.se
Avaya IP Office2.0PBXH323OKpeter[at]visionutv.se
Avaya Communications Manager2.0PBXSIPOKpeter[at]visionutv.se
Alcatel OmniPCX (OXO/OXE)2.0PBXSIPOKpeter[at]visionutv.se
Nortel BCM2.0PBXH323OKpeter[at]visionutv.se

Incompatibilities

There are cases when some equipments or services do not work with Yate or need special configurations. These are due to bugs or limitations in either that equipment or Yate. While we try to add more features and fix bugs in Yate some problems are simply outside our control.

You may look up the bug tracker and the mailing list for known compatibility issues with specific equipments or services as well as possible workarounds.

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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