Skype-to-Asterisk(SIP): Progress - Part 2
So, we have liftoff.
I have received the updated code from the programmer, and it does work as requested. I can now make calls (via SIP) to a Windows system running Skype and the revised PSGW software, and that system then turns the calls around and uses the SIP destination as the Skype username, and the call is terminated on the Skype network to the appropriate user. I've tested to various users and the echo123 test, and it sounds great! So I have the first SIP-to-Skype end-user specifiable gateway running. To avoid being Slashdotted, I won't post my test SIP destinations publicly, and of course the system can only handle one Skype conversation at a time.
Plans for the system: it will become (among other things) a find-me, follow-me type extension in our Asterisk diaplan - not only will the system ring your desk phone and cell phone, but it'll also try your Skype account.
Downsides: there is no caller ID, at least not as part of the signalling. All the calls to the Skype destinations get whatever account information is associated with the Skype account that is running on the Windows system. Of course, other methods could be used here, such as text-to-speech pre-call completions, DTMF (?? not sure how that works with Skype), or semi-out-of-band instant messaging exchanges. None of these methods are implemented yet, but it appears that they are being developed.
I've asked the programmer if he plans to release the revised code as his next update to PSGW.
JT
http://www.loligo.com/asterisk/
Click Here to Read Skype to Asterisk Part 1
blinklist : 
BoingBoing : 
del.icio.us : 
digg : 
furl :
shadows : 
simpy : 
Slashdot : 
spurl : 
yahoo
|
Posted to Asterisk Development