[svn-commits] oej: branch oej/calleridupdate r128415 - /team/oej/calleridupdate/

[SVN commits to the Digium repositories]

Author: oej
Date: Sun Jul  6 04:39:31 2008
New Revision: 128415

URL: http://svn.digium.com/view/asterisk?view=rev&rev=128415
Log:
Updating the README discussion document, based on feedback.

Modified:
    team/oej/calleridupdate/README-calleridupdate.txt

Modified: team/oej/calleridupdate/README-calleridupdate.txt
URL: http://svn.digium.com/view/asterisk/team/oej/calleridupdate/README-calleridupdate.txt?view=diff&rev=128415&r1=128414&r2=128415
==============================================================================
--- team/oej/calleridupdate/README-calleridupdate.txt (original)
+++ team/oej/calleridupdate/README-calleridupdate.txt Sun Jul  6 04:39:31 2008
@@ -1,5 +1,5 @@
-Olle E. Johansson					2008-07-04
-Edvina.net, Sollentuna, Sweden
+Olle E. Johansson					2008-07-06
+Edvina.net, Sollentuna, Sweden				v 2.0
 
 
 
@@ -29,7 +29,7 @@
 connected line ID updates in both directions. In some 
 protocols there are ways to update the phone devices.
 
-Updating caller IDs during a bridged call in Asterisk
+Updating caller IDs during a call in Asterisk
 -----------------------------------------------------
 There are multiple situations where a caller ID for connected line or 
 caller can happen in Asterisk calls.
@@ -52,9 +52,17 @@
 That means that channel drivers doesn't have any indication
 that it can forward to the phone to update the displays.
 
+In some of these cases, we have a bridged call. In many,
+like the call pickup examples, the call is in early
+state and not bridged.
+
 Connected line ID updates in ISDN
 ---------------------------------
-[Contributions to this section welcome!]
+Connected line ID in EuroISDN is covered in the following ETSI standard:
+	ETSI EN 300 094 V2.1.1 (2000-06)
+	European Standard (Telecommunications series)
+	Integrated Services Digital Network (ISDN); Connected Line Identification Presentation (COLP)
+	supplementary service;
 
 Connected line ID updates in SIP
 ---------------------------------
@@ -114,7 +122,7 @@
 
 Dialplan application needed
 ---------------------------
-A dialplan application is also needed so taht one-way
+A dialplan application is also needed so that one-way
 calls can change connected line ID.
 (Like setting a topic/name before connecting to a meetme).
 
@@ -136,12 +144,6 @@
 It implements a new structure in channel.h that replicates the
 current caller ID structure. I think we could stay with one 
 format instead of two very similar.
-This structure is added to ast_channel. For me this is
-questionnable. I would expect that the other call -the bridged
-call - is updated with this information and that we can get
-the information from there instead of storing it in the
-channel as a duplicate. If this works, we could also avoid
-sending data as a payload to the AST_CONTROL message.
 
 +/* \brief Structure for all kinds of connected line ID indentifications.
 + * \note All string fields here are malloc'ed, so they need to be
@@ -164,8 +166,12 @@
 needed. A big thank you to "gareth" for this work, and for being
 very responsive to input during the process.
 
+Since I'm focusing on SIP myself, I see a few changes needed to
+chan_sip in order to support more ways of updating the caller
+ID on the phone display. We also need to look into the 
+privacy issues.
 
------
-This branch is just a copy of trunk until we've made a few decisions 
-on the way forward.
+----
+Thanks to Gareth, Ramon Peek, Raj Jain and others who has given
+feedback on this document