My SIP Switch is handling only SIP messages, no audio packets passes through it. So there is not even a probability that delays or jitters affects your call quality because of My SIP Switch.According to the site the calls don't actually pass through their server anyway.
Yes.Looks like you basically have to write a script manually on their site to get it to work?
Hey thanks man. Some simple how to register an account for incoming and outgoing calls, would go a long way! (I simply could not get my FWD account to dial through to my ATA)We'll work on some proper documentation (we are seriously lacking of it)
err ... That one is done though. Once you are logged in there is a help page.Some simple how to register an account for incoming and outgoing calls, would go a long way