Merged revisions 58931 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r58931 | russell | 2007-03-15 17:25:12 -0500 (Thu, 15 Mar 2007) | 13 lines

Merge changes from svn/asterisk/team/russell/LaTeX_docs.

* Convert most of the doc directory into a single LaTeX formatted document
  so that we can generate a PDF, HTML, or other formats from this
  information.
* Add a CLI command to dump the application documentation into LaTeX format
  which will only be include if the configure script is run with 
  --enable-dev-mode.
* The PDF turned out to be close to 1 MB, so it is not included.  However, you
  can simply run "make asterisk.pdf" to generate it yourself.  We may include
  it in release tarballs or have automatically generated ones on the web site,
  but that has yet to be decided.

........


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@58932 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 0 insertions, 15 deletions

diff --git a/doc/model.txt b/doc/model.txt
deleted file mode 100644
index 10d2d0e05..000000000
--- a/doc/model.txt
+++ /dev/null
@@ -1,15 +0,0 @@
-Description of call model:
-
-Incoming Call:
-
-	Channel backend waits for a RING or equivalent on some sort of
-interface. Typically this is done in its own thread.  When a RING is
-detected, the backend should create a channel structure and then call
-ast_pbx_start() on that channel, which will create a thread to monitor
-that interface.  At this point, the PBX and/or applications it launches
-will manage the interface, and it need not be monitored by the
-aforementioned thread.  When the applications are finished, the requisite
-hangup function will be called, at which the channel can be considered to
-be no longer valid, and the thread that controls it will imminently be
-terminated. 
-