Merged revisions 58902 via svnmerge from

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

........
r58902 | russell | 2007-03-14 12:04:38 -0500 (Wed, 14 Mar 2007) | 2 lines

Add a more basic example setup to the examples section

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58904 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/doc/sla.pdf b/doc/sla.pdf
index e61bdee..c9f927e 100644 (file)
Binary files a/doc/sla.pdf and b/doc/sla.pdf differ

diff --git a/doc/sla.tex b/doc/sla.tex
index 735285e..c1159ce 100644 (file)
--- a/doc/sla.tex
+++ b/doc/sla.tex
@@ -47,34 +47,7 @@ system control each individual light on every phone to ensure that it shows the
 correct state of the line.  The phones must subscribe to the state of each of their
 line appearances.
 
-\begin{verbatim}
-       
-[line1]
-exten => s,1,SLATrunk(line1)
-
-[line2]
-exten => s,2,SLATrunk(line2)
-
-[sla_stations]
-exten => station1,1,SLAStation(station1)
-exten => station1_line1,hint,SLA:station1_line1
-exten => station1_line1,1,SLAStation(station1_line1)
-exten => station1_line2,hint,SLA:station1_line2
-exten => station1_line2,1,SLAStation(station1_line2)
-
-exten => station2,1,SLAStation(station2)
-exten => station2_line1,hint,SLA:station2_line1
-exten => station2_line1,1,SLAStation(station2_line1)
-exten => station2_line2,hint,SLA:station2_line2
-exten => station2_line2,1,SLAStation(station2_line2)
-
-exten => station3,1,SLAStation(station3)
-exten => station3_line1,hint,SLA:station3_line1
-exten => station3_line1,1,SLAStation(station3_line1)
-exten => station3_line2,hint,SLA:station3_line2
-exten => station3_line2,1,SLAStation(station3_line2)
-
-\end{verbatim}
+Please refer to the examples section for full dialplan samples for SLA.
 
 \subsection{Trunks}
 \label{trunks}
@@ -186,6 +159,74 @@ Here are some hints on configuring a SIP phone for use with SLA:
 
 
 \section{Configuration Examples}
+\subsection{Basic SLA}
+
+This is an example of the most basic SLA setup.  It uses the automatic
+dialplan generation so the configuration is minimal.
+
+sla.conf:
+\begin{verbatim}
+[line1]
+type=trunk
+device=Zap/1
+autocontext=line1
+
+[line2]
+type=trunk
+device=Zap/2
+autocontext=line2
+
+[station](!)
+type=station
+trunk=line1
+trunk=line2
+autocontext=sla_stations
+
+[station1](station)
+device=SIP/station1
+
+[station2](station)
+device=SIP/station2
+
+[station3](station)
+device=SIP/station3
+       
+\end{verbatim}
+
+With this configuration, the dialplan is generated automatically.  The first
+zap channel should have its context set to "line1" and the second should be
+set to "line2" in zapata.conf.  In sip.conf, station1, station2, and station3
+should all have their context set to "sla\_stations".
+
+For reference, here is the automatically generated dialplan for this situation:
+\begin{verbatim}
+[line1]
+exten => s,1,SLATrunk(line1)
+
+[line2]
+exten => s,2,SLATrunk(line2)
+
+[sla_stations]
+exten => station1,1,SLAStation(station1)
+exten => station1_line1,hint,SLA:station1_line1
+exten => station1_line1,1,SLAStation(station1_line1)
+exten => station1_line2,hint,SLA:station1_line2
+exten => station1_line2,1,SLAStation(station1_line2)
+
+exten => station2,1,SLAStation(station2)
+exten => station2_line1,hint,SLA:station2_line1
+exten => station2_line1,1,SLAStation(station2_line1)
+exten => station2_line2,hint,SLA:station2_line2
+exten => station2_line2,1,SLAStation(station2_line2)
+
+exten => station3,1,SLAStation(station3)
+exten => station3_line1,hint,SLA:station3_line1
+exten => station3_line1,1,SLAStation(station3_line1)
+exten => station3_line2,hint,SLA:station3_line2
+exten => station3_line2,1,SLAStation(station3_line2)   
+\end{verbatim}
+
+
 \subsection{SLA and Voicemail}
 \label{voicemail}
 
@@ -284,8 +325,6 @@ exten => station3_line2,1,SLAStation(station3_line2)
 This section is intended to describe how Asterisk handles calls inside of the
 SLA system so that it is clear what behavior is expected.
 
-Note that this section is not yet complete.
-
 \subsection{Station goes off hook (not ringing)}
 
 When a station goes off hook, it should initiate a call to Asterisk with the