Non che non ci fosse già arrivato, con soluzioni software come Skip2PBX o hardware come VoSky (basate su Linux e di fatto su varie sessioni simultanee del client Skype per Linux, poco scalabili secondo Stefan Öberg VP & GM Telecom di Skype).

L’annuncio di Skype per Asterisk è una notizia davvero significativa. Skype doveva arrivare sui PBX aziendali e in Digium trova il partner ideale. Il prodotto non sarà opensource ma commerciale. Gli utenti di un centralino Asterisk potranno chiamare da e verso estensioni Skype, trasferire chiamate etc, tutto quello che potete attendervi da un’integrazione simile.

Skype su Asterisk significa anche che attraverso una macchina con installato Asterisk sarà possibile interconnettere SIP e Skype, e forse far comunicare gli utenti di un network SIP, come quello di Gizmo, con gli utenti Skype… sempre che questo non sia espressamente vietato dalla licenza con cui Skype fornirà il software necessario. Siamo all’alba dell’interconnessione SIP/Skype, quella che il CEO di Gizmo chiedeva a gran voce solo la settimana scorsa?

Qui si può richiedere accesso alla beta (privata).

Una possibilità interessante per una piccola azienda potrebbe essere dotare tutti i suoi dipendenti di Skypephones di 3, con cui potranno essere raggiunti dal centralino aziendale, ovunque si trovino, senza alcun costo.

Di seguito altri dettagli dalla presentazione ufficiale, raccolti da Tom Keating. A partire dalla spiegazione su come sarà possibile gestire i trasferimenti di chiamata dal PBX all’utente Skype.

We’ve got a couple of ways to do it. The first and most simple way would be to create a local numeric alias for the Skype name. In that case you simply transfer the call to the numeric alias which then sends the call out the Skype channel. The extensions.conf logic looks like this:

exten => 6101,1,Dial(Skype/ssokol.digium)

In the above example the extension number is 6101 and the Skype name to which the call is forwarded is ssokol.digium.

Another mode of transfer would involve a graphical user interface like the Switchvox Switchboard. In that case the user would simply drag and drop the call on an appearance that maps to the Skype name. Under the covers it would use the Manager API to execute the transfer.

I’m sure that there are a number of other modes or techniques that could be used. Our developer community is very good at inventing clever solutions.

When asked how Skype IP-PBX gateway appliances are affected by this announcement, Stefan Öberg VP & GM Telecom for Skype said, “The appliances that are out there now have built their solutions on standard Linux client. They’ve used the public API on that and basically are running many instances of Skype Linux client. Obviously, that’s not the way the Linux client was meant to be implemented. So those solutions are not scalable or reliable to the extend that businesses would want them to be. The difference with this solution is that we’ve built it together to scale and to be reliable.”

When asked, “What about video integration?” Danny Wyndam responded, “The beta product that is available today does not support video. It is our plan to be able to support everything you can do in Skype through Asterisk. It’s just an evolution of the connector to this platform that we can add the video support.”

Da FierceVoIP la notizia dell’annuncio.

Digium and Skype have announced a “collaboration” between the two companies to bring together the Asterisk telephony platform and Skype’s service.

Delivering this morning’s keynote address at the conference, Skype VP Stefan Oberg said business users utilizing Skype want a PBX to route their calls. The Skype on Asterisk solution will allow PBX users to make calls using Skype names or via normal phone numbers.

The partnership is being touted as bringing together the 338 million worldwide Skype users to the tens of millions of Asterisk users. Since about 30 percent of Skype users are business users, there’s a lot of potential to introduce more businesses to Digium and Asterisk.

Currently, the solution is being tested in a private beta; people can apply to participate at www.astricon.net/skype.

Connectivity will be delivered through a channel driver. Licensing will be done on a per channel basis and the software will be a commercial, not open source, license. Pricing will be announced at the time of official release of the product.

On the Skype side, Oberg said businesses will be able to purchase a package to provide some extra support, calling credits and some “additional value-added things.” The exact details on how the packaging will work are not yet fully defined. “The details we’ll work out together.”

The Skype-Asterisk channel driver will support Skype’s wideband codec, and Asterisk will have the capability to transcode between Skype’s native codecs and other codecs. It will allow users to register as multiple Skype users, so a particular extension can ring to a Skype name. Multiple calls per user are also supported.

Via AstriCon 2008: Digium & Skype announce interoperability collaboration – FierceVoIP