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

[Todd Lieberman]

John, i think MGCP has this feature.

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of John Todd
Sent: Friday, May 07, 2004 5:55 PM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Concept for line appearances and bridging:
anyone?




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
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