[Asterisk-Users] Concept for line appearances and bridging: anyone?

[John Todd]

OK, here's a configuration challenge: I want to have certain line 
appearances able to be "interrupted" by various other line apperances 
elsewhere in the office.  This is harder to describe than it is to 
demonstrate, so I'll do that:

Let's assume I have Cisco 7960's on all desks.

  1) Call comes from inbound line X destined on extension 1234

  2) Phones A, B, C all ring on line appearance 1234 (there is a 
specific line labelled "1234" on each phone)

  3) User A picks up the ringing call on 1234.   Line X and User A are bridged.

  4) User B saw the caller ID on the call before it was picked up by 
user A, but she wants to talk to the caller as well since she has 
some relevant information.  User B picks up the phone and pushes the 
"1234" extension button.  A warning tone is played into the 
conversation between X and User A, and then User B is bridged into 
the conversation.  User B then talks with X and User A, and then 
hangs up.

This is _extremely_ relevant to office PBX systems.  In fact, it's 
one of the most used features - the ability to share a call with 
other people in the office just by hitting the right "line 
appearance" button.  Has anyone come up with a reasonable solution to 
delivering this feature?  For small offices, this is really a 
mandatory feature though as the number of calls increases this 
becomes more useless in an inbound setting (though as a workgroup 
feature it gains usefulness with size of the organization.  I'll skip 
the business cases for why this is a good idea and leave it as an 
exercise for the reader.)

I have come up with ideas on doing this with some really horrible, 
nasty, awful ideas that involve MeetMe rooms, but.... <shudder>... 
they're really not the right way to do it.  There must be some clever 
way of doing this with a new channel specification that would allow 
bridging into an existing channel identifier.  I.E.: 
Dial(Bridge/SIP/2203-bed5)


Other related topics:

  - The auto-dial I can handle with PLAR ("hotline" calling - pick up 
the phone, and automatically a number is dialed) and DISA on the 
Asterisk side.  In other words, when someone picks up line #1 on 
their Cisco 7960 (or whatever phone) I can have the system auto-dial 
into my * server.  Using the caller ID, I can determine what line 
they're calling from.  If there is nobody on that "line appearance", 
then I can give them a DISA to allow them to dial a regular call, as 
if the auto-ringdown didn't happen.

  - This feature becomes useful now that we have some phones that 
support "SUBSCRIBE" methods to allow other phones to show who is on 
what lines.  We can _see_ who is on the line, but there is no ability 
to add other lines to the call without transferring to a MeetMe 
(which then causes call control to be lost, and is a hassle, etc. 
etc. etc.)

JT