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Add a more basic example setup to the examples section
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58902 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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doc/sla.pdf
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doc/sla.pdf
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doc/sla.tex
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doc/sla.tex
@@ -47,34 +47,7 @@ system control each individual light on every phone to ensure that it shows the
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correct state of the line. The phones must subscribe to the state of each of their
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line appearances.
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\begin{verbatim}
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[line1]
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exten => s,1,SLATrunk(line1)
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[line2]
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exten => s,2,SLATrunk(line2)
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[sla_stations]
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exten => station1,1,SLAStation(station1)
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exten => station1_line1,hint,SLA:station1_line1
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exten => station1_line1,1,SLAStation(station1_line1)
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exten => station1_line2,hint,SLA:station1_line2
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exten => station1_line2,1,SLAStation(station1_line2)
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exten => station2,1,SLAStation(station2)
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exten => station2_line1,hint,SLA:station2_line1
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exten => station2_line1,1,SLAStation(station2_line1)
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exten => station2_line2,hint,SLA:station2_line2
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exten => station2_line2,1,SLAStation(station2_line2)
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exten => station3,1,SLAStation(station3)
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exten => station3_line1,hint,SLA:station3_line1
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exten => station3_line1,1,SLAStation(station3_line1)
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exten => station3_line2,hint,SLA:station3_line2
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exten => station3_line2,1,SLAStation(station3_line2)
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\end{verbatim}
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Please refer to the examples section for full dialplan samples for SLA.
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\subsection{Trunks}
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\label{trunks}
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@@ -186,6 +159,74 @@ Here are some hints on configuring a SIP phone for use with SLA:
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\section{Configuration Examples}
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\subsection{Basic SLA}
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This is an example of the most basic SLA setup. It uses the automatic
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dialplan generation so the configuration is minimal.
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sla.conf:
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\begin{verbatim}
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[line1]
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type=trunk
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device=Zap/1
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autocontext=line1
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[line2]
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type=trunk
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device=Zap/2
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autocontext=line2
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[station](!)
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type=station
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trunk=line1
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trunk=line2
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autocontext=sla_stations
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[station1](station)
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device=SIP/station1
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[station2](station)
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device=SIP/station2
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[station3](station)
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device=SIP/station3
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\end{verbatim}
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With this configuration, the dialplan is generated automatically. The first
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zap channel should have its context set to "line1" and the second should be
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set to "line2" in zapata.conf. In sip.conf, station1, station2, and station3
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should all have their context set to "sla\_stations".
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For reference, here is the automatically generated dialplan for this situation:
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\begin{verbatim}
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[line1]
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exten => s,1,SLATrunk(line1)
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[line2]
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exten => s,2,SLATrunk(line2)
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[sla_stations]
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exten => station1,1,SLAStation(station1)
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exten => station1_line1,hint,SLA:station1_line1
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exten => station1_line1,1,SLAStation(station1_line1)
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exten => station1_line2,hint,SLA:station1_line2
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exten => station1_line2,1,SLAStation(station1_line2)
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exten => station2,1,SLAStation(station2)
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exten => station2_line1,hint,SLA:station2_line1
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exten => station2_line1,1,SLAStation(station2_line1)
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exten => station2_line2,hint,SLA:station2_line2
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exten => station2_line2,1,SLAStation(station2_line2)
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exten => station3,1,SLAStation(station3)
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exten => station3_line1,hint,SLA:station3_line1
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exten => station3_line1,1,SLAStation(station3_line1)
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exten => station3_line2,hint,SLA:station3_line2
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exten => station3_line2,1,SLAStation(station3_line2)
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\end{verbatim}
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\subsection{SLA and Voicemail}
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\label{voicemail}
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@@ -284,8 +325,6 @@ exten => station3_line2,1,SLAStation(station3_line2)
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This section is intended to describe how Asterisk handles calls inside of the
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SLA system so that it is clear what behavior is expected.
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Note that this section is not yet complete.
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\subsection{Station goes off hook (not ringing)}
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When a station goes off hook, it should initiate a call to Asterisk with the
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