[asterisk-commits] oej: branch oej/sip-callpickup-1.2 r73055 - /team/oej/sip-callpickup-1.2/

[SVN commits to the Asterisk project]

Author: oej
Date: Tue Jul  3 07:46:43 2007
New Revision: 73055

URL: http://svn.digium.com/view/asterisk?view=rev&rev=73055
Log:
Adding readme for this branch

Added:
    team/oej/sip-callpickup-1.2/README.callpickup   (with props)

Added: team/oej/sip-callpickup-1.2/README.callpickup
URL: http://svn.digium.com/view/asterisk/team/oej/sip-callpickup-1.2/README.callpickup?view=auto&rev=73055
==============================================================================
--- team/oej/sip-callpickup-1.2/README.callpickup (added)
+++ team/oej/sip-callpickup-1.2/README.callpickup Tue Jul  3 07:46:43 2007
@@ -1,0 +1,22 @@
+This branch is to test and play with the idea of SNOM call pickup.
+
+There's an existing patch, originally made by Frank Sautter, floating around. This patch
+works, but has some architecture issues we're trying to work around in this branch.
+I have some old ruins around, that will be removed when this work is completed.
+
+The idea is something like this:
+
+- When we get a notification for RINGING we add a fake call ID to the SIP message,
+  based on the Asterisk Call ID
+- When we get an INVITE with replaces, we walk the channels to find the proper channel
+  and start doing our magic to capture this call (call pickup)
+
+This way we can handle pickup on any channel we can subscribe to, not only SIP calls.
+We might use the pickupgroups for security.
+
+-----------------------------------------------
+This branch is based on 1.2. The code will be a bit less intrusive on 1.4, since there's
+already support for INVITE/replaces in that code base.
+-----------------------------------------------
+
+** This work is done together with Frank and Enercon in Germany.

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