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author | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-02-02 20:05:52 +0000 |
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committer | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-02-02 20:05:52 +0000 |
commit | f5e599aa5aeb788214cf736dd5938bca28d99fdf (patch) | |
tree | 43c4bd4db2c931138971fd478613520e57886ab3 /doc/queue.txt | |
parent | b55ad96cc5234d9c342ee099daf37e0748c2a9f1 (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/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 ------------------- |