asterisk users - May 2015 - 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

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?
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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?
> 
> --
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