[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