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<a href="http://bamboo.asterisk.org/browse/AST14-FREEBSD81-I386/log" style="font-family: Arial, sans-serif; font-size: 15px; font-weight:bold; color:#000">AST14-FREEBSD81-I386-123</a>
<span style="font-family: Arial, sans-serif; font-size: 14px;"> has been queued, but there's no agent capable of building it.</span>
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<a href="http://bamboo.asterisk.org/browse/AST14-FREEBSD81-I386/log" style="text-decoration: none; font-family: Arial, sans-serif;text-align:left;font-size:16px;font-weight:bold;color:#036" >Code Changes</a>
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<a href="http://bamboo.asterisk.org/browse/author/twilson" style="font-family: Arial, sans-serif; font-size: 13px; font-weight:bold; color:#000">
twilson</a><br>
Re-fix queue round-robin<br/>
<br/>
This part of the change for r315596 was incorrect. No bridge occurs<br/>
when doing a roundrobin dial and no one answers, so this code shouldn't<br/>
have been removed.<br/>
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(317574)
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<a href="http://bamboo.asterisk.org/browse/author/rmudgett" style="font-family: Arial, sans-serif; font-size: 13px; font-weight:bold; color:#000">
rmudgett</a><br>
Regression after r297603 (Improve handling of REGISTER requests with multiple contact headers.) <br/>
<br/>
Uninitialized variable.<br/>
<br/>
(issue #18640)<br/>
<br/>
(closes issue #18785)<br/>
Reported by: pnlarsson<br/>
Patches:<br/>
issue18785_enegaard.patch uploaded by enegaard (license 1197)<br/>
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(317719)
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<a href="http://bamboo.asterisk.org/browse/author/rmudgett" style="font-family: Arial, sans-serif; font-size: 13px; font-weight:bold; color:#000">
rmudgett</a><br>
Merged revisions 318671 via svnmerge from<br/>
<a href="https://origsvn.digium.com/svn/asterisk/branches/1.8">https://origsvn.digium.com/svn/asterisk/branches/1.8</a><br/>
<br/>
* The applicable fixes for v1.4 are the SIP deadlock and the in progress<br/>
masquerade check for multiple parties trying to pickup the same call.<br/>
issue18654_v1.4.patch uploaded by rmudgett (license 664)<br/>
<br/>
* Backported to v1.6.2.<br/>
issue18654_v1.6.2.patch uploaded by rmudgett (license 664)<br/>
<br/>
........<br/>
r318671 | alecdavis | 2011-05-13 10:52:08 +1200 (Fri, 13 May 2011) | 30 lines<br/>
<br/>
Fix directed group pickup feature code *8 with pickupsounds enabled<br/>
<br/>
Since 1.6.2, the new pickupsound and pickupfailsound in features.conf cause many issues.<br/>
<br/>
1). chan_sip:handle_request_invite() shouldn't be playing out the fail/success audio, as it has 'netlock' locked.<br/>
2). dialplan applications for directed_pickups shouldn't beep.<br/>
3). feature code for directed pickup should beep on success/failure if configured.<br/>
<br/>
Created a sip_pickup() thread to handle the pickup and playout the audio, spawned from handle_request_invite.<br/>
<br/>
Moved app_directed:pickup_do() to features:ast_do_pickup().<br/>
<br/>
Functions below, all now use the new ast_do_pickup()<br/>
app_directed_pickup.c:<br/>
pickup_by_channel()<br/>
pickup_by_exten()<br/>
pickup_by_mark()<br/>
pickup_by_part()<br/>
features.c:<br/>
ast_pickup_call()<br/>
<br/>
(closes issue #18654)<br/>
Reported by: Docent<br/>
Patches:<br/>
ast_do_pickup_1.8_trunk.diff.txt uploaded by alecdavis (license 585)<br/>
Tested by: lmadsen, francesco_r, amilcar, isis242, alecdavis, irroot, rymkus, loloski, rmudgett<br/>
<br/>
Review: <a href="https://reviewboard.asterisk.org/r/1185/">https://reviewboard.asterisk.org/r/1185/</a><br/>
........<br/>
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(318734)
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