[asterisk-users] Integration with skype

[Markus]

Am 23.05.2013 16:04, schrieb Richard Kenner:
>> For voice, you can use SipToSis. Works flawlessly with Asterisk and the
>> best part, it's free. :)
>>
>> www.mhspot.com/sts/
>> (site is down right now)
>
> And that's related to the problem with it: it hasn't been maintained for
> quite a while.

True, but it's still working if you follow the instructions carefully! 
:)  If you know of another FREE alternative let me know. I think there 
is also something available for FreeSWITCH. The only problem with 
SipToSis that I found is that if you run it in a VM (KVM in my case), 
and have more than 1 concurrent call going on, the audio of the second 
call will drift away and become asynchronous after a few seconds into 
the call. Couldn't find a fix so I set up several VMs for multiple 
concurrent calls. :D  If you use a dedicated server you probably won't 
have that issue...