asterisk users - May 2004 - 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

John, i think MGCP has this feature.

-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com]On Behalf Of John Todd
Sent: Friday, May 07, 2004 5:55 PM
To: asterisk-users@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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In order to try and keep some standardisation in Bridged Lines implementation on
Asterisk using SIP, you should look at the following IETF draft.

http://www.ietf.org/internet-drafts/draft-anil-sipping-bla-01.txt

There is work in progress within the SIP community on Bridged Lines, and there
are already a number of end point implementations using this draft.

Steve Towlson
Citel Technologies.

-----Original Message-----
From: John Todd [mailto:jtodd@loligo.com]
Sent: 08 May 2004 17:23
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Concept for line appearances and bridging:
anyone?


At 9:27 AM -0400 on 5/8/04, Todd Lieberman wrote:
>John, i think MGCP has this feature.
>
>-----Original Message-----
>From: asterisk-users-admin@lists.digium.com
>[mailto:asterisk-users-admin@lists.digium.com]On Behalf Of John Todd
>Sent: Friday, May 07, 2004 5:55 PM
>To: asterisk-users@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:
>
[snip]

MGCP may have this feature, but Asterisk should be able to provide 
this functionality on any channel type, not just MGCP.  The fact that 
we can manipulate the audio on the server with * implies that we can 
mix any two channels in an arbitrary way.  This implies (of course) 
that we keep the audio channel going through our * server, but for 
most PBX environments this isn't a concern, and in fact is a 
desirable goal.

JT

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Steve Towlson wrote:
> In order to try and keep some standardisation in Bridged Lines
implementation on Asterisk using SIP,
> you should look at the following IETF draft.
> 
> http://www.ietf.org/internet-drafts/draft-anil-sipping-bla-01.txt
> 
> There is work in progress within the SIP community on Bridged Lines, and
there are already a number
 > of end point implementations using this draft.
This draft requires
* Support for multiple UAs registrating the same AOR, which Asterisk does not
support today
* Support for the server to subscribe to extension states on the UA, which
Asterisk does not support today

/O

On Friday 07 May 2004 09:55 pm, John Todd wrote:
> 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 doubt this suggestion is of much use to you, yet it may help someone 
scanning the archive: the Polycom SoundPoint IP 600 (and probably the 500) 
has this ability.  I'm not sure this proves Asterisk can create this
feature;
yet, it strongly suggests that at least this function is possible in 
Asterisk.

Anon