[asterisk-users] Queue and Interface time out

[James Fromm]

Does anyone have ringinuse=no and autopause=yes working together in 
queues.conf?

We assign members to our customer service queue from an application 
based on actions the agents take on their PCs.  No static agents are 
defined in agents.conf and no members are specified in queues.conf.  All 
member interfaces are SIP with only the basics configured in sip.conf.

Even with 'ringinuse=no' configured, the Queue application continues to 
send callers to busy members causing them to get paused when their SIP 
device returns that it's busy.

Does the Queue application need hints for member interfaces to determine 
their status?

Thanks,
James

James Fromm wrote:
> No, call-limit is not being used.  Do you have ringinuse=no working? Has 
> anyone seen it work?
> 
> Each SIP device has a very minimal config in sip.conf.  Here's a show 
> sip peer:
> 
>   * Name       : 3207
>   Secret       : <Set>
>   MD5Secret    : <Not set>
>   Context      : outbound
>   Subscr.Cont. : <Not set>
>   Language     :
>   AMA flags    : Unknown
>   Transfer mode: open
>   CallingPres  : Presentation Allowed, Not Screened
>   Callgroup    :
>   Pickupgroup  :
>   Mailbox      : 3207 at omnis
>   VM Extension : asterisk
>   LastMsgsSent : 0/0
>   Call limit   : 0
>   Dynamic      : Yes
>   Callerid     : "Sam" <3207>
>   MaxCallBR    : 384 kbps
>   Expire       : 40
>   Insecure     : no
>   Nat          : RFC3581
>   ACL          : No
>   T38 pt UDPTL : No
>   CanReinvite  : No
>   PromiscRedir : No
>   User=Phone   : No
>   Video Support: No
>   Trust RPID   : No
>   Send RPID    : No
>   Subscriptions: Yes
>   Overlap dial : Yes
>   DTMFmode     : rfc2833
>   LastMsg      : 0
>   ToHost       :
>   Addr->IP     : 216.239.128.189 Port 5060
>   Defaddr->IP  : 0.0.0.0 Port 5060
>   Def. Username: 3207
>   SIP Options  : (none)
>   Codecs       : 0x8000e (gsm|ulaw|alaw|h263)
>   Codec Order  : (ulaw:20)
>   Auto-Framing:  No
>   Status       : OK (14 ms)
>   Useragent    : PolycomSoundPointIP-SPIP_650-UA/2.0.3.0131
>   Reg. Contact : sip:3207 at 216.239.128.189
> 
> 
> Watkins, Bradley wrote:
>>  
>>
>>> -----Original Message-----
>>> From: asterisk-users-bounces at lists.digium.com 
>>> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of James 
>>> Fromm
>>> Sent: Thursday, January 18, 2007 10:29 AM
>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>> Subject: Re: [asterisk-users] Queue and Interface time out
>>>
>>> I guess I'm missing something else.  'ringinuse = no' doesn't change 
>>> anything.  While on a call, the queue still sends another call and 
>>> proceeds to set the member paused after receiving 'Busy Here' back 
>>> from the SIP device.
>>>
>>> My queues.conf is:
>>>
>>> [general]
>>>
>>>     persistentmembers = no
>>>
>>> [customerservice]
>>>
>>>     persistentmembers = no
>>>     musiconhold = default
>>>     reportholdtime = no
>>>     strategy = leastrecent
>>>     timeout = 20
>>>     retry = 5
>>>     wrapuptime = 30 ;allow agents 30 seconds to wrap up work
>>>     maxlen = 0 ;unlimited callers on hold
>>>     servicelevel = 60 ;calls must be answered within 60 seconds
>>>     announce-holdtime = no
>>>     autopause = yes
>>>     ringinuse = no
>>>     joinempty = yes
>>>     leavewhenempty = no
>>>
>>> Am I missing something obvious?
>>>
>>
>>
>> What do your SIP peers look like?  Are you using the call-limit feature?
>>
>> - Brad
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