[Asterisk-Users] Skype-to-Asterisk(SIP): progress

[Alexander Lopez]

Kudos!!!

'Nuf said!


> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com 
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of 
> John Todd
> Sent: Thursday, January 26, 2006 2:44 PM
> To: asterisk-users at 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
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