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
