[Asterisk-Users] SIP and consultative transfer

[Peter Gulezian]

hello all-

Long time listener, first time caller. This is a great list and has  
given me tons of help as I've set up * for the first time.

I've got an asterisk system up and running at a new company, and it  
does about 99% of what we need it to do. TelephonyWare has been our  
equipment supplier, and has been great with support, but I've got an  
issue that has us both stumped. It's to the point where this is a  
must-have feature for our staff and if I can't provide it, we're  
going to have to abandon ship and get something else in here.

Before I start, here's our setup:
     - gentoo 2.6 system, dual xeon, 1GB RAM.. beefy machine.
     - TE110P interface to a 24-channel SBC voice T1
     - TDM400 providing four ports for faxes and etc.
     - Six Sipura 841 phones, and we're evaluating the Polycom IP600  
and the Cicco 7960

Here's the situation. Our office manager wants to be able to easily  
create conferences on her deskphone and then hang up on them while  
the other participants keep on talking. Sample sequence as she  
described it happening at a previous employer:

     1. Customer calls in over our T1 into the main line.  
Receptionist picks up.
     2. Customer wants to talk to boss. Receptionist hits  
'conference' and dials boss's cellphone.
     3. Boss picks up. Receptionist briefs him and hits 'conference'  
again to join the three parties.
     4. All three parties talk for a minute.
     5. Receptionist hangs up; boss and customer continue to talk,  
joined by the PBX.
     (note the total of 2 non-dialing button presses)

Now, we tried this with the IP 600 that we just received (which is a  
spectacular phone, by the way, especially in terms of voice quality -  
it was described by someone on the other end as "almost too clear" !)  
and what happened was that the phone took responsibility for the  
conference and did the audio mixing of all three parties. This is  
great, but then there's no way (it seems) to separate the boss and  
customer out so they can continue talking with the receptionist out  
of the picture, because if she hangs up, the whole thing is dropped.  
I can understand why this happens, technically, and am worried that  
what she is describing might not be possible in a receptionist- 
friendly way. It worked on her old system (I assume) because it was a  
digital PBX and the phone never took responsibility for the conference.

By 'easy,' I mean not involving dialing other extensions to transfer  
each line individually, or dealing with creating MeetMe rooms (which  
do work great for other things). A couple button presses, max. End  
result being that the conferenced parties finish their conversation  
in some hidden place managed by the PBX. Nobody else can join, but  
that's fine.

The end result, I assume, would look the same as if a standard  
transfer is done (this was done with the transfer key on the polycom)  
- I called in on my cell, and then was transferred to another cell  
phone:
pbx*CLI> show channels
         Channel  (Context    Extension    Pri )   State  
Appl.         Data
         Zap/2-1  (default    s            1   )      Up Bridged  
Call  Zap/1-1
         Zap/1-1  (international 914085551212 2   )      Up  
Dial          Zap/g1/14085551212


I've tried setting values in the Polycom's sip.cfg - specifically the  
<conference voIpProt.SIP.conference.address=""/> value. The only  
thing that makes a difference is setting address="5500 at pbx" (where  
5500 is a meetme dynamic conference maker) - but that just seems to  
throw the second call into limbo and drop the polycom into the newly  
made conference. Could this value work with meetme somehow to do what  
we want?

I saw http://bugs.digium.com/view.php?id=4297 - would that help?

It seems that this issue has been gone over a few times before:

http://lists.digium.com/pipermail/asterisk-users/2004-June/ 
050872.html is the post of an administrator with different equipment,  
but the same goal.

And: On Jun 20, 2005, at 1:53 PM, Pavel Jezek wrote:
> it can be done with sccp/skinny protocol with callmanager, it's  
> possible to make ad-hoc multiparty conference with audio mixing in  
> callmanager (not in phone)
> but probably, it's not possible with sip :-(
> even with sip phones, this is not true multiparty konference, but  
> only three-way calling :-(
> imho, currently only way to make true multiparty conference with  
> sip/asterisk is meetme, but it's not very usable, because calls  
> must be initiated from each user...
>
.. so that makes me doubtful.

So! The questions are :
     1. Is this something that I can configure in asterisk with the  
polycom? (optimal solution)
     2. Does the cisco 7960 have this ability? The manual describes  
handing off a conference, but I haven't been able to test it yet.


Any ideas?

-p