asterisk users - Jun 2005 - SNOM, Asterisk and Attended transfer (bug?)

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

> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-
> bounces@lists.digium.com] On Behalf Of Steve Davies
> Sent: Monday, June 13, 2005 6:17 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] SNOM, Asterisk and Attended transfer (bug?)
> 
> 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
> _______________________________________________
Upgrade your snom firmware to the latest and make sure break key = off
in the setup.

Use the transfer feature in asterisk for attended transfers.

On 6/13/05, Steve Davies <davies147@gmail.com>
wrote:
> 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)
> 
[snip]

The attended transfer problem is solved :) It turns out that an
attended transfer results in a different path for the RTP packets, and
it was our firewall rules not-expecting this behaviour that defeated
everything. *blush*

Thanks for all of the help and suggestions in the meantime.
Regards,
Steve