Update with info about SIP channels and queues
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@ -11,6 +11,25 @@ Asterisk Call Queues
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* Using dynamic queue members
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* SIP channel configuration
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---------------------------
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Queues depend on the channel driver reporting the proper state
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for each member of the queue. To get proper signalling on
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queue members that use the SIP channel driver, you need to
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enable a call limit (could be set to a high value so it
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is not put into action) and also make sure that both inbound
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and outbound calls are accounted for.
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Example:
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[general]
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limitonpeer = yes
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[peername]
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type=friend
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call-limit=10
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* Other references
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-------------------
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