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authoroej <oej@f38db490-d61c-443f-a65b-d21fe96a405b>2007-02-02 20:05:52 +0000
committeroej <oej@f38db490-d61c-443f-a65b-d21fe96a405b>2007-02-02 20:05:52 +0000
commitf5e599aa5aeb788214cf736dd5938bca28d99fdf (patch)
tree43c4bd4db2c931138971fd478613520e57886ab3 /doc
parentb55ad96cc5234d9c342ee099daf37e0748c2a9f1 (diff)
Update with info about SIP channels and queues
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@53127 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'doc')
-rw-r--r--doc/queue.txt19
1 files changed, 19 insertions, 0 deletions
diff --git a/doc/queue.txt b/doc/queue.txt
index b554b4151..11047f83f 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
-------------------