Kudos!!! 'Nuf said!
> -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > John Todd > Sent: Thursday, January 26, 2006 2:44 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Skype-to-Asterisk(SIP): progress > > > I'm sitting in the Emerging Telephony Conference, so this > seems a particularly apt place to pre-announce this... > > I've wanted to be able to gateway calls between Skype and > Asterisk for a while, which of course would require some type > of protocol converter (IAX or SIP to Skype, probably.) This > of course is directly not in Skype's interest, since they > would like to keep the network closed (boo!) so that users > are forced to use their PSTN gateway and other > revenue-generating systems. On the other hand, I'm trying to > crack this open so that any VoIP channel can talk to any > other VoIP channel. Asterisk provides the ideal platform for > this type of conversion, if only Skype were accessible... > > Please hold flames about how Skype is the enemy of open > telephony standards. I don't disagree. However, for a small > sub-set of users that I work with, Skype is a channel that is > preferred for audio in some circumstances, and I feel that > it's worthwhile to have some ability to connect with users > who have expressed that preference. > > There exists a commercial program called "PSGW" > (http://www.rsdevs.com/) which runs on (booo!) Windows and > does SIP to Skype conversion. It's about $29 USD. It uses > the Skype API to create calls in both directions, and then > uses somewhat of a kludge using software audio "cables" > between a SIP/RTP driver system and the Skype API. It works > reasonably well, but to date has been somewhat limited > because it will only terminate calls to a specific Skype user > on the far end which is mapped in the program itself. This > has been somewhat limiting, since that means I can't > arbitrarily specify a user in the SIP invite to whom I want > to communicate. > > I have contacted the company (programmer) that sells this > software, and I've negotiated a payment to him to patch the > code such that PSGW will allow arbitrary specification of > Skype-side user choice, as I've asked that this be released > as part of the general distribution of this commercial > software. He says that this should be ready within the next > week or two for testing by me, and then I've asked that the > code is released into the next versions of PSGW. So > basically, I'm putting out a press release about someone > else's commercial software, but I think it's worth noting > because of the usefulness of this when used in conjunction > with Asterisk. > > I'll keep the list updated with the progress of the code and > tests with Asterisk. > > JT > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
