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                    <a href="http://bamboo.asterisk.org/browse/AST162-FREEBSD81-I386/log" style="font-family: Arial, sans-serif; font-size: 15px; font-weight:bold; color:#000">AST162-FREEBSD81-I386-97</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/author/twilson" style="font-family: Arial, sans-serif; font-size: 13px; font-weight:bold; color:#000">
                       twilson</a><br>
                    Merged revisions 293493 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/>
........<br/>
&nbsp;&nbsp;r293493 | twilson | 2010-11-01 09:58:00 -0500 (Mon, 01 Nov 2010) | 14 lines<br/>
&nbsp;&nbsp;<br/>
&nbsp;&nbsp;Only offer codecs both sides support for directmedia<br/>
&nbsp;&nbsp;<br/>
&nbsp;&nbsp;When using directmedia, Asterisk needs to limit the codecs offered to just<br/>
&nbsp;&nbsp;the ones that both sides recognize, otherwise they may end up sending audio<br/>
&nbsp;&nbsp;that the other side doesn't understand.<br/>
&nbsp;&nbsp;<br/>
&nbsp;&nbsp;(closes issue 0017403)<br/>
&nbsp;&nbsp;Reported by: one47<br/>
&nbsp;&nbsp;Patches: <br/>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;sip_codecs_simplified4 uploaded by one47 (license 23)<br/>
&nbsp;&nbsp;Tested by: one47, falves11<br/>
&nbsp;&nbsp;<br/>
&nbsp;&nbsp;Review: <a href="https://reviewboard.asterisk.org/r/967/">https://reviewboard.asterisk.org/r/967/</a> [^]<br/>
........<br/>
<br/>
Backporting a bugfix that should have been included.<br/>

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                     (302049)
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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 302172 via svnmerge from <br/>
<a href="https://origsvn.digium.com/svn/asterisk/branches/1.4">https://origsvn.digium.com/svn/asterisk/branches/1.4</a><br/>
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........<br/>
&nbsp;&nbsp;r302172 | rmudgett | 2011-01-18 12:04:36 -0600 (Tue, 18 Jan 2011) | 88 lines<br/>
&nbsp;&nbsp;<br/>
&nbsp;&nbsp;Issues with DTMF triggered attended transfers.<br/>
&nbsp;&nbsp;<br/>
&nbsp;&nbsp;Issue #17999<br/>
&nbsp;&nbsp;1) A calls B. B answers.<br/>
&nbsp;&nbsp;2) B using DTMF dial *2 (code in features.conf for attended transfer).<br/>
&nbsp;&nbsp;3) A hears MOH. B dial number C<br/>
&nbsp;&nbsp;4) C ringing. A hears MOH.<br/>
&nbsp;&nbsp;5) B hangup. A still hears MOH. C ringing.<br/>
&nbsp;&nbsp;6) A hangup. C still ringing until &quot;atxfernoanswertimeout&quot; expires.<br/>
&nbsp;&nbsp;For v1.4 C will ring forever until C answers the dead line. (Issue #17096)<br/>
&nbsp;&nbsp;<br/>
&nbsp;&nbsp;Problem: When A and B hangup, C is still ringing.<br/>
&nbsp;&nbsp;<br/>
&nbsp;&nbsp;Issue #18395<br/>
&nbsp;&nbsp;SIP call limit of B is 1<br/>
&nbsp;&nbsp;1. A call B, B answered<br/>
&nbsp;&nbsp;2. B *2(atxfer) call C<br/>
&nbsp;&nbsp;3. B hangup, C ringing<br/>
&nbsp;&nbsp;4. Timeout waiting for C to answer<br/>
&nbsp;&nbsp;5. Recall to B fails because B has reached its call limit.<br/>
&nbsp;&nbsp;<br/>
&nbsp;&nbsp;Because B reached its call limit, it cannot do anything until the transfer<br/>
&nbsp;&nbsp;it started completes.<br/>
&nbsp;&nbsp;<br/>
&nbsp;&nbsp;Issue #17273<br/>
&nbsp;&nbsp;Same scenario as issue 18395 but party B is an FXS port.&nbsp; Party B cannot<br/>
&nbsp;&nbsp;do anything until the transfer it started completes.&nbsp; If B goes back off<br/>
&nbsp;&nbsp;hook before C answers, B hears ringback instead of the expected dialtone.<br/>
&nbsp;&nbsp;<br/>
&nbsp;&nbsp;**********<br/>
&nbsp;&nbsp;Note for the issue #17273 and #18395 fix:<br/>
&nbsp;&nbsp;<br/>
&nbsp;&nbsp;DTMF attended transfer works within the channel bridge.&nbsp; Unfortunately,<br/>
&nbsp;&nbsp;when either party A or B in the channel bridge hangs up, that channel is<br/>
&nbsp;&nbsp;not completely hung up until the transfer completes.&nbsp; This is a real<br/>
&nbsp;&nbsp;problem depending upon the channel technology involved.<br/>
&nbsp;&nbsp;<br/>
&nbsp;&nbsp;For chan_dahdi, the channel is crippled until the hangup is complete.<br/>
&nbsp;&nbsp;Either the channel is not useable (analog) or the protocol disconnect<br/>
&nbsp;&nbsp;messages are held up (PRI/BRI/SS7) and the media is not released.<br/>
&nbsp;&nbsp;<br/>
&nbsp;&nbsp;For chan_sip, a call limit of one is going to block that endpoint from any<br/>
&nbsp;&nbsp;further calls until the hangup is complete.<br/>
&nbsp;&nbsp;<br/>
&nbsp;&nbsp;For party A this is a minor problem.&nbsp; The party A channel will only be in<br/>
&nbsp;&nbsp;this condition while party B is dialing and when party B and C are<br/>
&nbsp;&nbsp;conferring.&nbsp; The conversation between party B and C is expected to be a<br/>
&nbsp;&nbsp;short one.&nbsp; Party B is either asking a question of party C or announcing<br/>
&nbsp;&nbsp;party A.&nbsp; Also party A does not have much incentive to hangup at this<br/>
&nbsp;&nbsp;point.<br/>
&nbsp;&nbsp;<br/>
&nbsp;&nbsp;For party B this can be a major problem during a blonde transfer.&nbsp; (A<br/>
&nbsp;&nbsp;blonde transfer is our term for an attended transfer that is converted<br/>
&nbsp;&nbsp;into a blind transfer.&nbsp; :)) Party B could be the operator.&nbsp; When party B<br/>
&nbsp;&nbsp;hangs up, he assumes that he is out of the original call entirely.&nbsp; The<br/>
&nbsp;&nbsp;party B channel will be in this condition while party C is ringing, while<br/>
&nbsp;&nbsp;attempting to recall party B, and while waiting between call attempts.<br/>
&nbsp;&nbsp;<br/>
&nbsp;&nbsp;WARNING:<br/>
&nbsp;&nbsp;The ATXFER_NULL_TECH conditional is a hack to fix the problem.&nbsp; It will<br/>
&nbsp;&nbsp;replace the party B channel technology with a NULL channel driver to<br/>
&nbsp;&nbsp;complete hanging up the party B channel technology.&nbsp; The consequences of<br/>
&nbsp;&nbsp;this code is that the 'h' extension will not be able to access any channel<br/>
&nbsp;&nbsp;technology specific information like SIP statistics for the call.<br/>
&nbsp;&nbsp;<br/>
&nbsp;&nbsp;ATXFER_NULL_TECH is not defined by default.<br/>
&nbsp;&nbsp;**********<br/>
&nbsp;&nbsp;<br/>
&nbsp;&nbsp;(closes issue #17999)<br/>
&nbsp;&nbsp;Reported by: iskatel<br/>
&nbsp;&nbsp;Tested by: rmudgett<br/>
&nbsp;&nbsp;JIRA SWP-2246<br/>
&nbsp;&nbsp;<br/>
&nbsp;&nbsp;(closes issue #17096)<br/>
&nbsp;&nbsp;Reported by: gelo<br/>
&nbsp;&nbsp;Tested by: rmudgett<br/>
&nbsp;&nbsp;JIRA SWP-1192<br/>
&nbsp;&nbsp;<br/>
&nbsp;&nbsp;(closes issue #18395)<br/>
&nbsp;&nbsp;Reported by: shihchuan<br/>
&nbsp;&nbsp;Tested by: rmudgett<br/>
&nbsp;&nbsp;<br/>
&nbsp;&nbsp;(closes issue #17273)<br/>
&nbsp;&nbsp;Reported by: grecco<br/>
&nbsp;&nbsp;Tested by: rmudgett<br/>
&nbsp;&nbsp;<br/>
&nbsp;&nbsp;Review: <a href="https://reviewboard.asterisk.org/r/1047/">https://reviewboard.asterisk.org/r/1047/</a><br/>
........<br/>

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                     (302173)
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