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...




--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
              http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to