call-limit = number in sip.conf for peers 2008/3/17, Rajkumar S <[EMAIL PROTECTED]>: > > Hi, > > I am using asterisk-1.4.15, My sip configs is like > > [2501] > type=friend > username=2501 > secret=2501 > canreinvite=no > host=dynamic > dtmfmode=rfc2833 > context = sip > disallow=all > allow=ulaw > incominglimit=1 > nat=1 > > queue.conf is like > > [gen-enq] > joinempty = yes > musiconhold = default > strategy = rrmemory > servicelevel = 60 > timeout = 60 > retry = 5 > wrapuptime=5 > announce-frequency = 90 > announce-holdtime = yes > monitor-format = wav > ringinuse = no > > I am using AddQueueMember to add SIP interface to the queue. Each sip > interface is member of multiple queues. Occasionally I get messages > like > > [Mar 17 11:33:01] ERROR[9253]: chan_sip.c:3232 update_call_counter: > Call to peer '2505' rejected due to usage limit of 1 > [Mar 17 11:33:01] ERROR[9254]: chan_sip.c:3232 update_call_counter: > Call to peer '2509' rejected due to usage limit of 1 > [Mar 17 11:33:01] ERROR[9255]: chan_sip.c:3232 update_call_counter: > Call to peer '2502' rejected due to usage limit of 1 > [Mar 17 11:33:01] ERROR[9256]: chan_sip.c:3232 update_call_counter: > Call to peer '2506' rejected due to usage limit of 1 > > in my asterisk console. At this point the mentioned sip phones are > busy. My understanding is that if ringinuse is set to no, queue should > not try and ring phones that are busy, but some how it is trying. How > can I disable this behavior? > > With regards, > raj > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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