Update with info about SIP channels and queues

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53127 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/doc/queue.txt b/doc/queue.txt
index b554b41518ab435fc37b42020c99d1a2ae5cad35..11047f83fcf8709eb57d5f703c3704a679fec3f5 100644 (file)
--- 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
 -------------------