Main • Transfers

On my opinion, assisted transfer should work this way: The goals are: be fast to use (less key to type as possible), with a predictable behavior (you should be able to guess what will happen), and reduce/recover from errors (don't panic if you do a mistake!), Should be easy add the capability of: a) 3 way conversation b) blind transfer (that is the less used in a office)

Of course this is MY dream and my own vision.

C = Caller from outside (like Customer) S = Secretary (the “Switch” that routes the call with the transfer) T = the one that is the Target of the call of C. T extension is 999

Let's put that the “*” key enables the assisted transfer (of course, should be configurable and “multi key” enabled, so I could set it to the “flash” key, or have “**” instead of the single “*”, or “#9”, or whatever)

Different scenarios:

1. C calls and S answers:

a) for blind transfer: S presses *999 and hangs up. If T is busy or timeouts, the call goes back to C, if C is still available, or becomes “parked” somewhere (a special FIFO “park zone”) while S is busy (S hears a background sound telling her that someone has been parked).

b) for a successful assisted transfer: S press *999, T answers, S announces C, then S hangs up. If instead S presses “3”, the 3 way calling is enabled (C, S and T can talk together).

c) T does not want to talk with C: S press *999, T answers, S announces C, T does not want to talk with him, so T hangs up.

d) T is busy or don't answer: S can press *XXX to go to a new extension XXX (so S can try to transfer to someone else, or if has typed a non-existent extension, can immediately retry), or S can press ** (the defined key twice, so in the unfortunate situation where it has been defined “**” will be “****”), or hang up and immediately pick up (good for “panic situations”, even better if a “hangup key” is defined and it's picked up automatically). (note that there should be no timeout here, so S can try the transfer as many times as needed. But if really a timeouts is needed S should go back with conversation with C)

2. C calls T

a) in a database "C calling T" is specified with a priority: - High: T receives the call from C (automatic routing) - Medium: S receives an event notifycation, and can pick up the call in the first 2-3 rings or park it (even if he/she is busy talking to someone else), othewise T receives the call from C - Low: the call is "parked" or S answers

b) C is not in the database at all or C calls someone else difrent from T, so S answeres and we go back to 1. type scenarios

Note: the database can be automaticly / manualy updated, it is recomended to configure the databse with known C persons, before first (official) usage to make sure that their routed the right way, and dayly checkings should be made

N.B.: transfer and 3 way calling can be used also if you want to call to another external number. I.e. you call Tom, then you decide that you have also to hear Peter's opinion. You press *0123456, where “123456” is Peter's number, and “0” is the prefix for outgoing calls in your dialplan, and then you talk briefly with Peter and press “3” to have you, Tom and Peter talk together.

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.

EditHistoryBacklinksRecent ChangesSearch