[Asterisk-Users] Hypothetical IAX2 situation

[Sean Kennedy]

Two * servers:  *a and *b.

Outside call comes in *b, and is automatically routed to *a.  Someone on 
a sip phone connected to *a then decides to transfer the call to someone 
on a sip phone connected to *b.  The transfer works.

At this point, is *a still in the converstation?  Or is * smart enough 
to see where the data stream is going/coming from?


Thanks for any help in advanced, and sorry if this has been asked 
before.  I didn't know what I was looking for, so my searching was limited.

Sean