[svn-commits] rmudgett: branch group/CCSS r240221 - /team/group/CCSS/doc/tex/ccss.tex

[SVN commits to the Digium repositories]

Author: rmudgett
Date: Thu Jan 14 12:55:00 2010
New Revision: 240221

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=240221
Log:
Update CCSS documentation for 1.8 and fix some typos.

Modified:
    team/group/CCSS/doc/tex/ccss.tex

Modified: team/group/CCSS/doc/tex/ccss.tex
URL: http://svnview.digium.com/svn/asterisk/team/group/CCSS/doc/tex/ccss.tex?view=diff&rev=240221&r1=240220&r2=240221
==============================================================================
--- team/group/CCSS/doc/tex/ccss.tex (original)
+++ team/group/CCSS/doc/tex/ccss.tex Thu Jan 14 12:55:00 2010
@@ -1,6 +1,6 @@
 \section{Introduction}
 
-	A new feature for Asterisk 1.6.4 is Call Completion Supplementary
+	A new feature for Asterisk 1.8 is Call Completion Supplementary
 Services. This document aims to explain the system and how to use it.
 In addition, this document examines some potential troublesome points
 which administrators may come across during their deployment of the
@@ -32,7 +32,7 @@
 recipient's phone is busy, the caller will have the opportunity to
 request CCBS. When the recipient's phone is no longer busy, the caller
 will be alerted. The means by which the caller is alerted is dependent
-upon the type of agent used  by the caller.
+upon the type of agent used by the caller.
 
 \item CCNR: Call Completion on No Response. When a call fails because the
 recipient does not answer the phone, the caller will have the opportun-
@@ -278,14 +278,14 @@
 is that Asterisk will happily let you violate the advice given and
 allow you to set up a trunk with a generic monitor or agent. While this
 will not cause anything catastrophic to occur, the behavior will most
-definitely be buggy.
+definitely not be what you want.
 
 \item At the time of this writing (2009 Oct), Asterisk is the only
 known SIP stack to write an implementation of
 draft-ietf-bliss-call-completion-04. As a result, it is recommended
 that for your SIP phones, use a generic agent and monitor. For SIP
 trunks, you will only be able to use CC if the other end is
-terminated by another Asterisk server running version 1.6.4 or later.
+terminated by another Asterisk server running version 1.8 or later.
 
 \item If the Dial application is called multiple times by a single
 extension, CC will only be offered to the caller for the parties called
@@ -328,7 +328,7 @@
 and dialplan to show basic usage of generic call completion.
 It is likely that if you have a more complex setup, you will
 need to make use of items like the CALLCOMPLETION dialplan
-funtion or the CC\_INTERFACES channel variable.
+function or the CC\_INTERFACES channel variable.
 
 First, let's establish a very simple sip.conf to use for this