[asterisk-commits] oej: branch oej/calleridutf8 r89568 - /team/oej/calleridutf8/README.thisbranch

[SVN commits to the Asterisk project]

Author: oej
Date: Sun Nov 25 15:18:48 2007
New Revision: 89568

URL: http://svn.digium.com/view/asterisk?view=rev&rev=89568
Log:
Add README and todolist

Added:
    team/oej/calleridutf8/README.thisbranch   (with props)

Added: team/oej/calleridutf8/README.thisbranch
URL: http://svn.digium.com/view/asterisk/team/oej/calleridutf8/README.thisbranch?view=auto&rev=89568
==============================================================================
--- team/oej/calleridutf8/README.thisbranch (added)
+++ team/oej/calleridutf8/README.thisbranch Sun Nov 25 15:18:48 2007
@@ -1,0 +1,49 @@
+Caller ID utf8
+--------------
+Goal:
+- Implement Caller ID names in utf8 for SIP, IAX2 and jingle
+- Implement Caller ID domains for SIP and Jingle
+
+The domains will be used for SIP2SIP calls primarily, as domain is  
+part of the address and should not be replaced in the outbound call.
+
+I've expanded the caller-ID structure in channel.h for these new  
+fields and implement them in various functions that handle caller ID's.
+
+When murf has integrated libiconv for the dialplan handling, I'll  
+implement conversion between these. For now, IAX2, SIP and Jingle will use the UTF8 version  
+of the caller ID name if it exists, otherwise the Ascii name.
+
+In jingle, iax.conf and SIP.conf, you will be able to set an extra utf8 caller  
+ID name to be used in channels that can handle it. This can be implemented in all channels,  
+so that calls TO an utf8 enabled channel will get the full Caller ID Name.
+
+Manager stays Ascii for now, maybe we need to change that. XML manager
+should be converted to utf8 and have both names.
+
+This way, calls between IAX2 and SIP phones will be able to preserve  
+the full display name.
+
+I don't know if the IAX2 protocol needs to be extended to carry the  
+ASCII transliterated caller ID name in addition to the UTF8 which now is standard Caller ID  
+name in IAX2.
+This would be handy for trunking.
+
+Feedback, ideas and suggestions is always welcome.
+
+/Olle
+
+----------------
+Done:
+- Fix internal APIs 
+	- ast_channel_alloc
+	- ast_set_caller_id
+- Fix func_callerid
+- Add to pvts for jingle, gtalk, sip, iax2
+- Add to peers/users in SIP
+
+Todo:
+- Configuration option in channel.conf files
+- Implement utf8 caller ID handling in channels
+- Implement domain handling
+- Add to peers/users in IAX2

Propchange: team/oej/calleridutf8/README.thisbranch
------------------------------------------------------------------------------
    svn:eol-style = native

Propchange: team/oej/calleridutf8/README.thisbranch
------------------------------------------------------------------------------
    svn:keywords = Author Date Id Revision

Propchange: team/oej/calleridutf8/README.thisbranch
------------------------------------------------------------------------------
    svn:mime-type = text/plain