[asterisk-users] Queue and Interface time out

[James Fromm]

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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