Andrew Martin
2015-May-06 20:44 UTC
[asterisk-users] Phones don't stop ringing when queue is answered
Hello, I am running Asterisk 11 on CentOS 6.4 with about 150 local SIP clients on a LAN. The SIP clients are a mixture of Yealink phones (e.g SIP-T32G, SIP-T42G, etc). I have configured the system as follows: sip.conf: [169] secret=111111 dtmfmode=rfc2833 directmedia=no directrtpsetup=yes canreinvite=no context=main host=dynamic type=friend port=5060 call-limit=5 nat=force_rport,comedia callcounter=yes queues.conf: [queue_level_1] musiconhold=default music=default strategy=ringall joinempty=yes timeout=18 member => SIP/178 member => SIP/146 member => SIP/169 extensions.conf: [test-queue] exten => s,1,WaitExten(2) same => n,Queue(queue_level_1,rtnC,18) same => n,Playback(transfer_exten) same => n,WaitExten(2) same => n,Queue(queue_level_1,rtnC,18) same => n,Playback(user_unavail) same => n,Voicemail(169 at myvm,s) same => n,WaitExten(2) same => n,Hangup() This rings a group of phones for 18 seconds, and if no one answers it repeats ringing that same group. If no one answers the second time, it goes to voicemail. I have noticed an intermittent problem where if the caller hangs up or a particular phone, say 146, answers the call, the other phones in the queue will continuing ringing for several seconds before realizing that the caller is no longer in the queue. This is very problematic since users are then answering the queue to find no one there. The log does not show anything amiss (calling from 265 into the queue): -- Executing [s at test-queue:2] Queue("SIP/265-00002931", "queue_level_1,rtnC,18") in new stack == Using SIP RTP CoS mark 5 == Using SIP RTP CoS mark 5 == Using SIP RTP CoS mark 5 -- SIP/146-00002934 is ringing -- SIP/146-00002934 is ringing -- SIP/178-00002933 is ringing -- SIP/178-00002933 is ringing -- Nobody picked up in 18000 ms -- Nobody picked up in 18000 ms -- Nobody picked up in 18000 ms -- Exiting on time-out cycle -- Executing [s at test-queue:3] Playback("SIP/265-00002931", "transfer_exten") in new stack -- <SIP/265-00002931> Playing 'transfer_exten.slin' (language 'en') -- Executing [s at test-queue:4] WaitExten("SIP/265-00002931", "2") in new stack -- Timeout on SIP/265-00002931, continuing... -- Executing [s at test-queue:5] Queue("SIP/265-00002931", "queue_level_1,rtnC,18") in new stack == Using SIP RTP CoS mark 5 == Using SIP RTP CoS mark 5 == Using SIP RTP CoS mark 5 -- SIP/146-00002937 is ringing -- SIP/146-00002937 is ringing -- SIP/178-00002936 is ringing == Spawn extension (test-queue, s, 5) exited non-zero on 'SIP/265-00002931' This only happens occassionally; most of the time the phones will all immediately stop ringing once one of them picks up. Do you have any ideas about what could be wrong here or what else I can do to debug? Thanks, Andrew Martin
James Thomas
2015-May-07 15:20 UTC
[asterisk-users] Phones don't stop ringing when queue is answered
What purpose do the WaitExten()s serve here? Are you really allowing the caller to connect to different extensions in the test-queue context? Have you tried without the WaitExten()s? -------------- next part -------------- An HTML attachment was scrubbed... URL: <lists.digium.com/pipermail/asterisk-users/attachments/20150507/d97ed1a4/attachment.html>
Andrew Martin
2015-May-07 22:03 UTC
[asterisk-users] Phones don't stop ringing when queue is answered
James, The WaitExten()s just provide a pause between the two Queue() calls to let the first group of phones finish ringing. In this example I am ringing the same group (queue_level_1) twice, however in a real-world scenario I would ring queue_level_1 and then ring queue_level_2 which each have a different list of phones. Thanks, Andrew ----- Original Message -----> From: "James Thomas" <jthomasdpu at gmail.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> > Sent: Thursday, May 7, 2015 10:20:10 AM > Subject: Re: [asterisk-users] Phones don't stop ringing when queue is answered > > What purpose do the WaitExten()s serve here? Are you really allowing the > caller to connect to different extensions in the test-queue context? Have > you tried without the WaitExten()s? > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > lists.digium.com/mailman/listinfo/asterisk-users