[asterisk-dev] [Code Review] Private representation of caller, connected and redirecting party ids
Kaloyan Kovachev
kkovachev at varna.net
Wed Jul 11 10:07:08 CDT 2012
There is a long waiting review at https://reviewboard.asterisk.org/r/1676/
which also adds connected and redirecting party ids to sig_ss7 and either
this patch should cover sig_ss7 too or the one at 1676 should be updated
(once again) to include the private representation too.
https://reviewboard.asterisk.org/r/2025 (and probably others) also touches
functionality which is added/extended in 1676 and it is quite difficult to
follow all the changes for such a big change (patch).
Please review and commit 1676 or it may end up outdated even before it is
reviewed, as it is a pain to keep it fresh and to follow all the changes at
different parts of the functionality which just have not existed before.
On Wed, 11 Jul 2012 14:26:06 -0000, "Guenther Kelleter"
<reviewboard at asterisk.org> wrote:
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/2030/
> -----------------------------------------------------------
>
> (Updated July 11, 2012, 9:26 a.m.)
>
>
> Review request for Asterisk Developers, Matt Jordan and Thomas Arimont.
>
>
> Summary
> -------
>
> (Info for Matt Jordan: I wanted to post this on Thomas Arimont's account
> because he implemented this, but I cannot login with the login data you
> sent him, so I'm posting it on my account.)
>
> This patch adds the feature "Private representation of caller, connected
> and redirecting party ids", as previously discussed with us (DATUS) and
> Digium.
>
> 1. Feature motivation
> Until now it is quite difficult to modify a party number or name which
can
> only be seen by exactly one particular instantiated technology channel
> resp. subscriber.
> One example where a modified party number or name on one channel is
spread
> over several channels are supplementary services like call transfer or
> pickup. To implement these features asterisk internally copies caller
and
> connected ids from one channel to another.
> Another example are extension subscriptions. The monitoring entities
> (watchers) are notified of statechanges and - if desired - of party
> numbers or names which represent the involving call parties.
> One major feature where a private representation of party names is
> essentially needed, i.e. where a party name shall exclusively signaled
to
> only one particular user, is a private user-specific name resolution for
> party numbers. A lookup in a private destination-dependent telephone
book
> shall provide party names which cannot be seen by any other user at any
> time.
>
> 2. Feature Description
> This feature comes along with the implementation of additional private
> number
> and name elements for caller id, connected id and redirecting ids inside
> asterisk channels.
>
> The private number and private name elements can be read or set by the
> user using asterisk dialplan functions.
>
> When a technology channel is initiating a call, receives an internal
> connectedline update event or receives an internal redirecting update
> event, it first checks if there is a valid connected id or redirecting
id
> private name or number element present. If this is the case it uses this
> private representation for protocol signaling .
>
> -->The channel technologies which initially do support the private id
> representation are SIP (chan_sip), mISDN (chan_misdn) and PRI
> (chan_dahdi).<--
>
> Once a private name or number on a channel is set and (implicitly) made
> valid, it is generally used for any further protocol signaling until it
is
> rewritten or invalidated.
> To simplyfy the invalidation of private ids all internally generated
> connected/redirecting update events and also all connected/redirecting
> update events which are generated by technology channels – receiving
> regarding protocol information - automatically trigger the invalidation
of
> private ids.
>
> --> If not using the private number and name representation feature at
> all, i.e. if using only the 'regular' callerid, connected and
redirecting
> related functions, the current characteristic of asterisk is not
affected
> by the new extended functionality. <--
>
> 3. User interface Description
> To grant access to the private name and number representation from the
> asterisk dialplan, the read and write functions of the variables
CALLERID,
> CONNECTEDLINE and REDIRECTING are extended by the following datatypes.
The
> formats of these datatypes are equal to the corresponding regular
> 'non-private' already existing datatypes:
>
> CALLERID:
> privnum privnum-valid privnum-pres privnum-plan
> privname privname-valid privname-pres privname-charset
>
> CONNECTEDLINE:
> privnum privnum-valid privnum-pres privnum-plan
> privname privname-valid privname-pres privname-charset
>
> REDIRECTING:
> orig-privnum orig-privnum-valid orig-privnum-pres orig-privnum-plan
> orig-privname orig-privname-valid orig-privname-pres
orig-privname-charset
> from-privnum from-privnum-valid from-privnum-pres from-privnum-plan
> from-privname from-privname-valid from-privname-pres
from-privname-charset
> to-privnum to-privnum-valid to-privnum-pres to-privnum-plan
> to-privname to-privname-valid to-privname-pres to-privname-charset
>
>
> Diffs
> -----
>
> /trunk/channels/chan_misdn.c 369918
> /trunk/channels/chan_sip.c 369918
> /trunk/channels/sig_pri.c 369918
> /trunk/funcs/func_callerid.c 369918
> /trunk/include/asterisk/channel.h 369918
> /trunk/main/channel.c 369918
> /trunk/main/features.c 369918
>
> Diff: https://reviewboard.asterisk.org/r/2030/diff
>
>
> Testing
> -------
>
>
> Thanks,
>
> Guenther
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