1st post here so I thought a Hello would be a good start. I've been a definite "voip end user" for that last 5 years and have set up UK DIDs for my family who live abroad and saved them and us hundreds of pounds in calls over that time. I've used a SPA3102 and a Gigaset N300a - both of which are great bits of kit.
In the next 6 months we will be giving up our BT landline and moving to super duper FTTH broadband. I'm here so I can learn and set up a home voip setup that rocks.
I registered here so I could ask (and hopefully help) questions about freepbx on the pi, but I realy need to understand some underlying SIP concepts first. I'm a IT sysadmin (once upon a time I was a unix admin) so you can probably get as tecnical as you need to without issue.
So my 1st question is this: Once a incoming call is connected through an asterisk server can you then "break out" to another extension? I know the receiveing user on an asterisk extenstion can transfer (blind or attendeded) but can you configure it so the asterisk server will listen for DTMF on an incoming trunk and act upon it?
Alternatively can you get a SPA3102 or OBi110 to do the same trick? i.e. can you allow an incoming (authenticated) user to transfer themselves away from the middle of a VMail recording to another extension - possibly an asterisk server?