[Asterisk-bugs] [Asterisk 0010166]: The device state of this queue member, SIP/XXXXX, is still 'Not in Use' when it probably should not be!

[noreply at bugs.digium.com]

The following issue has been RESOLVED. 
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http://bugs.digium.com/view.php?id=10166 
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Reported By:                bcnit
Assigned To:                file
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Project:                    Asterisk
Issue ID:                   10166
Category:                   Applications/app_queue
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     resolved
Asterisk Version:           1.4.6 
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!):  
Disclaimer on File?:        Yes 
Request Review:              
Resolution:                 not fixable
Fixed in Version:           
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Date Submitted:             07-10-2007 02:05 CDT
Last Modified:              07-10-2007 07:33 CDT
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Summary:                    The device state of this queue member, SIP/XXXXX, is
still 'Not in Use' when it probably should not be!
Description: 

This appears to be related to bug http://bugs.digium.com/view.php?id=0008580.

sip, queues and queue members are being driven from realtime (MySQL).

I get several warnings when queueing calls:
    -- Executing [s at cust-in:44] Queue("SIP/103-095182f0", "0844304|n") in
new stack
[Jul 10 07:49:00] WARNING[6677]: translate.c:163 framein: no samples for
gsmtolin
    -- Started music on hold, class 'default', on SIP/103-095182f0
    -- SIP/101-09522e00 is ringing
    -- SIP/101-09522e00 is ringing
    -- SIP/101-09522e00 is ringing
[Jul 10 07:49:02] WARNING[4029]: chan_sip.c:3140 update_call_counter:
Inringing for peer '101' < 0?
    -- SIP/101-09522e00 answered SIP/103-095182f0
    -- Stopped music on hold on SIP/103-095182f0
[Jul 10 07:49:02] WARNING[6677]: app_queue.c:2646 try_calling: The device
state of this queue member, SIP/101, is still 'Not in Use' when it probably
should not be! Please check UPGRADE.txt for correct configuration
settings.
  == Spawn extension (cust-in, s, 44) exited non-zero on
'SIP/103-095182f0'


I have followed the instructions in UPGRADE.txt (see below).

The result is that calls are offered to handsets which are already on a
call and this has broken a 1.2 based application when ported to 1.4.

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---------------------------------------------------------------------- 
 file - 07-10-07 07:33  
---------------------------------------------------------------------- 
This is a known issue of realtime, just like using qualify and mailbox
without caching. Call limits were not designed to work under those
circumstances and without major changes can't. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
07-10-07 07:33  file           Status                   new => resolved     
07-10-07 07:33  file           Resolution               open => not fixable 
07-10-07 07:33  file           Assigned To               => file            
07-10-07 07:33  file           Note Added: 0067042                          
======================================================================