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Update with info about SIP channels and queues

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Olle Johansson
2007-02-02 20:05:52 +00:00
parent 0205c27767
commit a29a9d9564
1 changed files with 19 additions and 0 deletions
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19
doc/queue.txt
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@@ -11,6 +11,25 @@ Asterisk Call Queues
* Using dynamic queue members
-----------------------------
* SIP channel configuration
---------------------------
Queues depend on the channel driver reporting the proper state
for each member of the queue. To get proper signalling on
queue members that use the SIP channel driver, you need to
enable a call limit (could be set to a high value so it
is not put into action) and also make sure that both inbound
and outbound calls are accounted for.
Example:
[general]
limitonpeer = yes
[peername]
type=friend
call-limit=10
* Other references
-------------------