Forgot to add:<br>Multiple queues fo sip phone, it is normal that sometimes it is ringed, as reported busy for 1 queue and free for another. you limitited incoming call to max 1 '
incominglimit=1' so ;)<br><br><div><span class="gmail_quote">2008/3/17, Grygoriy Dobrovolskyy <<a href="mailto:megahohol@gmail.com">megahohol@gmail.com</a>>:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
call-limit = number in sip.conf for peers<br><br><div><span class="gmail_quote">2008/3/17, Rajkumar S <<a href="mailto:rajkumars@gmail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">rajkumars@gmail.com</a>>:</span><div>
<span class="e" id="q_118bcd188e1a9162_1"><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Hi,<br> <br> I am using asterisk-1.4.15, My sip configs is like<br> <br> [2501]<br> type=friend<br> username=2501<br> secret=2501<br> canreinvite=no<br> host=dynamic<br> dtmfmode=rfc2833<br> context = sip<br> disallow=all<br>
allow=ulaw<br> incominglimit=1<br> nat=1<br> <br> queue.conf is like<br> <br> [gen-enq]<br> joinempty = yes<br> musiconhold = default<br> strategy = rrmemory<br> servicelevel = 60<br> timeout = 60<br> retry = 5<br> wrapuptime=5<br>
announce-frequency = 90<br> announce-holdtime = yes<br> monitor-format = wav<br> ringinuse = no<br> <br> I am using AddQueueMember to add SIP interface to the queue. Each sip<br> interface is member of multiple queues. Occasionally I get messages<br>
like<br> <br> [Mar 17 11:33:01] ERROR[9253]: chan_sip.c:3232 update_call_counter:<br> Call to peer '2505' rejected due to usage limit of 1<br> [Mar 17 11:33:01] ERROR[9254]: chan_sip.c:3232 update_call_counter:<br>
Call to peer '2509' rejected due to usage limit of 1<br> [Mar 17 11:33:01] ERROR[9255]: chan_sip.c:3232 update_call_counter:<br> Call to peer '2502' rejected due to usage limit of 1<br> [Mar 17 11:33:01] ERROR[9256]: chan_sip.c:3232 update_call_counter:<br>
Call to peer '2506' rejected due to usage limit of 1<br> <br> in my asterisk console. At this point the mentioned sip phones are<br> busy. My understanding is that if ringinuse is set to no, queue should<br> not try and ring phones that are busy, but some how it is trying. How<br>
can I disable this behavior?<br> <br> With regards,<br> raj<br> <br> _______________________________________________<br> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">http://www.api-digital.com</a> --<br>
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