[Asterisk-Users] SNOM, Asterisk and Attended transfer (bug?)

[Steve Davies]

Hi,

I am using a number of snom190 phones, and an asterisk "gateway"
server, and recently started experimenting with call transfers. The
snom phones provide support for attended and un-attended call
transfer, so I would rather use that than call-parking.

I have found that un-attended transfer works fine, and that attended
transfer works fine if the originating phone call is NON-SIP (ie.
ISDN)

I hope that some of this makes sense...

When I look at the SIP trace for the sequence of A calls B and is
transferred to C, I see:
A makes call to B:
  A calls B
  B picks up
  A and B are bridged (re-INVITEd) and talk directly.
B then puts A on hold:
  (A and B are both INVITE to talk via Asterisk)
B makes a call to C, I see:
  B calls C
  C picks up
  B and C are bridged (re-INVITEd) and talk directly.
B presses transfer:
  (Same as putting B and C on hold, B and C are re-INVITEd to talk via Asterisk)
B selects which line to transfer to C
  B REFERs A to C by asking Asterisk. Asterisk accepts this.
  B is notified that A is disconnected
  B gets "BYE" for call to A
  B gets "BYE" for call to C
  C gets INVITE to talk to B via Asterisk <<<<<<<< Why????? Why not to 'A'
  B requests that call to A is closed down.

The upshot of all this is that B is correctly out of the loop, and
that Both A and C think they have opened communications with a new
phone, both via Asterisk. Unfortunately there is no Audio. If one of
the parties hangs up, the connection is correctly closed.

I am curious why Asterisk would put a "From:" of "B" in the final
INVITE to bridge the calls. Perhaps this is just how SIP spoofs the
communication so that C does not need to know about the 2 callers?

Is there some way I can track down where my audio is going? As
mentioned, this problem only seems to occur if A,B,C are all SIP
phones, but not if A is an ISDN call.

Thanks,
Steve