[asterisk-dev] Caller/connected line ID updates * I NEED YOUR INPUT!!!

Matt Florell astmattf at gmail.com
Sun Jul 6 05:47:55 CDT 2008


Hello,

I agree that most of my suggestions are outside of your scope and
would take quite a bit of work, thanks for including them in your
appendix though.

I will be at Astricon in September and would like to meet on the
subject if you get a gathering together.

Thanks,

MATT---

On 7/6/08, Johansson Olle E <oej at edvina.net> wrote:
> While updating my document, I went through your comments again. While
>  being very valid,
>  I wonder how much we should change the caller ID handling in this
>  work, or if we could
>  take it step by step. I have other changes as well, that I deem
>  outside of the scope here,
>  but it's good to have in mind in the process, like UTF8 caller ID name
>  handling,
>  and SIP uri handling.
>
>  If I can make it to Astricon, which I don't know yet, I would like to
>  sit down with
>  a group of people and go through this whole area. Before that, I need to
>  complete the caller ID updates we have now, since it's a customer
>  project
>  too.
>
>  So, a new version of the document to review is at:
>
> >> http://svn.digium.com/view/asterisk/team/oej/calleridupdate/README-calleridupdate.txt?view=co
>
>
>
> /O
>
>
>  4 jul 2008 kl. 12.01 skrev Matt Florell:
>
>
> > Hello,
>  >
>  > I've dealt a lot with CallerID-related issues with all sorts of
>  > carriers, clients and applications all over the world on many
>  > different types of systems. Here is a quick example of what I think
>  > Asterisk should do with callerID information in trunk.
>  >
>  > Each channel would have the following callerID containers:
>  > - inbound_caller_id_num
>  > - inbound_caller_id_name
>  > - inbound_ani
>  > - outbound_caller_id_num
>  > - outbound_caller_id_name
>  > - outbound_ani
>  > - xfer_caller_id_num
>  > - xfer_caller_id_name
>  > - xfer_ani
>  > - caller_id_active
>  > - call_label
>  >
>  > The callerIDs are separated by name, number and ANI as well as the
>  > end-point that the specific IDs came from.
>  >
>  > caller_id_active is the definition of the callerID name and number
>  > that would be used when an action like a Redirect takes place
>  > (inbound, outbound, xfer).
>  >
>  > call_label would be a definable label within Asterisk for a call that
>  > would follow a channel just like callerID currently does. This is
>  > present because channel variables do not always traverse all channel
>  > types and channel bridging and this field would allow more complete
>  > and much easier tracking of a channel by 3rd party applications.
>  >
>  > With all of these containers of data, just about any kind of CallerID
>  > could be served no matter the technology or application.
>  >
>  >
>  > I hope this helps,
>  >
>  > MATT---
>  >
>  >
>  > On 7/4/08, Johansson Olle E <oej at edvina.net> wrote:
>  >> Friends,
>  >>
>  >> We need to implement a way to update caller and callee identies
>  >> during
>  >> a bridged call
>  >> or a one-way call in Asterisk. Several people in the Asterisk
>  >> community has been working
>  >> on this for a long time and we need to make a few design decisions
>  >> and
>  >> move forward.
>  >>
>  >> To update a caller ID, we need a new AST_CONTROL message with or
>  >> without payload.
>  >> The channel driver needs to be able to send and or receive this
>  >> message and do what
>  >> they can do with it.
>  >>
>  >> We also need to have a way to configure whether this should be done,
>  >> per device
>  >> and per call. Do we want to send the phone number and name of someone
>  >> in a
>  >> call center that answers a call? If a secretary answer's the boss
>  >> phone by call pickup,
>  >> do we send the secretary's caller ID?
>  >>
>  >> I've tried to compile my thoughts and input from various documents as
>  >> well as
>  >> a review of "gareth"'s work in the bug tracker.
>  >> http://svn.digium.com/view/asterisk/team/oej/calleridupdate/README-calleridupdate.txt?view=co
>  >>
>  >> I know that Connected line IDs are supported in EuroISDN. Any
>  >> information about these
>  >> functions that I can add in the document is appreciated.
>  >>
>  >> Please take time to review this document, provide additional input
>  >> and
>  >> we can
>  >> summarize around next week (after the fourth of July weekend in the
>  >> States).
>  >>
>  >> This will be integrated into svn trunk, since it's a massive change
>  >> to
>  >> many
>  >> modules as well as the core of Asterisk. It won't be part of 1.6.0,
>  >> but I'm sure we can provide a
>  >> backport for that branch too. Let's not focus on that in this
>  >> discussion.
>  >>
>  >> For those of you in the States: Have a nice Fourth of July weekend!
>  >> For the rest of you: Have a nice summer weekend!
>  >>
>  >> /O
>  >>
>  >> _______________________________________________
>  >> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>  >>
>  >> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
>  >> Register Now: http://www.astricon.net
>  >>
>  >> asterisk-dev mailing list
>  >> To UNSUBSCRIBE or update options visit:
>  >>   http://lists.digium.com/mailman/listinfo/asterisk-dev
>  >>
>  >
>  > _______________________________________________
>  > --Bandwidth and Colocation Provided by http://www.api-digital.com--
>  >
>  > AstriCon 2008 - September 22 - 25 Phoenix, Arizona
>  > Register Now: http://www.astricon.net
>  >
>  > asterisk-dev mailing list
>  > To UNSUBSCRIBE or update options visit:
>  >   http://lists.digium.com/mailman/listinfo/asterisk-dev
>
>
> ---
>
> * Olle E Johansson - oej at edvina.net
>  * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden
>
>
>
>
>
>  _______________________________________________
>  --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
>  AstriCon 2008 - September 22 - 25 Phoenix, Arizona
>  Register Now: http://www.astricon.net
>
>  asterisk-dev mailing list
>  To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-dev
>



More information about the asterisk-dev mailing list