From a29a9d9564c4c20e41465fe7c2aa2aad9013948f Mon Sep 17 00:00:00 2001 From: Olle Johansson Date: Fri, 2 Feb 2007 20:05:52 +0000 Subject: [PATCH] Update with info about SIP channels and queues git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53127 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- doc/queue.txt | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) diff --git a/doc/queue.txt b/doc/queue.txt index b554b41518..11047f83fc 100644 --- a/doc/queue.txt +++ b/doc/queue.txt @@ -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 -------------------