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| author | Olle Johansson <oej@edvina.net> | 2007-02-02 20:05:52 +0000 |
|---|---|---|
| committer | Olle Johansson <oej@edvina.net> | 2007-02-02 20:05:52 +0000 |
| commit | a29a9d9564c4c20e41465fe7c2aa2aad9013948f (patch) | |
| tree | 43c4bd4db2c931138971fd478613520e57886ab3 /doc/queue.txt | |
| parent | 0205c2776745803eb9dcbf5d25e6fb3c11b35a20 (diff) | |
Update with info about SIP channels and queues
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'doc/queue.txt')
| -rw-r--r-- | doc/queue.txt | 19 |
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 ------------------- |