[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]

A NOTE has been added to this issue. 
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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:                     feedback
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:              
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Date Submitted:             07-10-2007 02:05 CDT
Last Modified:              07-10-2007 08:53 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 08:53  
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That's not something we do... and there, chan_sip portion done. That's the
best I can do! 

Issue History 
Date Modified   Username       Field                    Change               
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07-10-07 08:53  file           Note Added: 0067060                          
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