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

Sun Jul 6 08:15:51 CDT 2008

6 jul 2008 kl. 12.47 skrev Matt Florell:

> 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 guess that you understand that one has to limit the scope in order
to get anywhere. However, I need all kinds of feedback on this issue
so that we can define the scope. I'm very grateful for your information
and anyone else that gives feedback. Please don't hesitate, let me
process all information and sort it up :-)

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

I hope that I can be there too.

/O

> 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
>>>>
>>>> _______________________________________________
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>>>>
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>>>>
>>>
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>>
>>
>> ---
>>
>> * 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--
>>
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>> Register Now: http://www.astricon.net
>>
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>>   http://lists.digium.com/mailman/listinfo/asterisk-dev
>>
>
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>
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