Insane amounts of yucky satanic code to make transfer and that kind of thing work.
Transfers work better when both legs of the call live in thier own channel eg bridged calls
A -> B where you want a to make B -> C

when you route a call to an IVR or playback app you are not really bridging you have
A all alone executing the script so it's hard to transfer that.

I do have it aparently working but it's goofy and you are better off
putting your IVR on it's own switch so they are all inbound calls
then you have A -> B -> IVR
now A can happily transfer B who can stay on line with IVR without stopping
the execution.  You can also accomplish this by calling in a loop back to the same box
if you dont want to have 2 boxes.


Also the beginning effort at bridging calls with no media is here
set this magic variable in your dialplan to convince mod_sofia
to pass A's sdp as it's own to B and return B's sdp back to A on 200 or 183

<action application="set" data="no_media=true"/>
<action application="bridge" data="sofia/id@host.com"/>

You will need a new sofia tarball for this version


There is a bunch of other odds and ends added like a function or 2 etc
Oh,

And don't be suprised if it introduces all kinds of bugs!



git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2992 d0543943-73ff-0310-b7d9-9358b9ac24b2