Rajkumar S
2008-Mar-17 06:06 UTC
[asterisk-users] update_call_counter: Call to peer '2509' rejected due to usage limit of 1?
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
Grygoriy Dobrovolskyy
2008-Mar-17 12:58 UTC
[asterisk-users] update_call_counter: Call to peer '2509' rejected due to usage limit of 1?
call-limit = number in sip.conf for peers 2008/3/17, Rajkumar S <rajkumars at gmail.com>:> > 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080317/00384cdb/attachment.htm