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<a href="http://bamboo.asterisk.org/browse/AST14/" style="font-family: Arial, sans-serif; font-size: 15px; font-weight:bold; color:#d62829">Asterisk - 1.4</a> >
<a href="http://bamboo.asterisk.org/browse/AST14-CENTOS55/" style="font-family: Arial, sans-serif; font-size: 15px; font-weight:bold; color:#d62829">CentOS 5.5</a> >
<a href="http://bamboo.asterisk.org/browse/AST14-CENTOS55-77/" style="font-family: Arial, sans-serif; font-size: 15px; font-weight:bold; color:#d62829">#77</a>
<span class="failed" style="font-family: Arial, sans-serif; font-size: 14px;"> has failed.</span>
<span class="failed" style="font-family: Arial, sans-serif; font-size: 13px;">
<br/> This build occurred because it is a dependant of <a href="http://bamboo.asterisk.org/browse/AST14-LUCID-107">AST14-LUCID-107</a>.
<br/>No failed tests found, a possible compilation error.
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<a href="http://bamboo.asterisk.org/browse/AST14-CENTOS55-77/" style="text-decoration: none; font-family: Arial, sans-serif;text-align:left;font-size:16px;font-weight:bold;color:#036" >Failing Jobs</a>
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<a href="http://bamboo.asterisk.org/browse/AST14-CENTOS55-I386-77/" style="color:red;">i386</a>
<span style="color:#444444; font-size:11px; font-style:italic;">(Default Stage)</span>
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<b>Duration:</b> 4 minutes
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<b>Tests:</b> 4 passed
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<a href="http://bamboo.asterisk.org/browse/AST14-CENTOS55-I386-77/log">Logs</a> | <a href="http://bamboo.asterisk.org/browse/AST14-CENTOS55-I386-77/artifact">Artifacts</a>
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<a href="http://bamboo.asterisk.org/browse/AST14-CENTOS55-77/commit/" 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/AST14-CENTOS55-77/commit/" style="font-family: Arial, sans-serif; font-size: 13px; color:#036">View all 5 code changes</a><img src="http://bamboo.asterisk.org/images/icons/more.gif" width="15" height="15" style="vertical-align:middle;padding:2px">
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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>
Issues with DTMF triggered attended transfers.<br/>
<br/>
Issue #17999<br/>
1) A calls B. B answers.<br/>
2) B using DTMF dial *2 (code in features.conf for attended transfer).<br/>
3) A hears MOH. B dial number C<br/>
4) C ringing. A hears MOH.<br/>
5) B hangup. A still hears MOH. C ringing.<br/>
6) A hangup. C still ringing until "atxfernoanswertimeout" expires.<br/>
For v1.4 C will ring forever until C answers the dead line. (Issue #17096)<br/>
<br/>
Problem: When A and B hangup, C is still ringing.<br/>
<br/>
Issue #18395<br/>
SIP call limit of B is 1<br/>
1. A call B, B answered<br/>
2. B *2(atxfer) call C<br/>
3. B hangup, C ringing<br/>
4. Timeout waiting for C to answer<br/>
5. Recall to B fails because B has reached its call limit.<br/>
<br/>
Because B reached its call limit, it cannot do anything until the transfer<br/>
it started completes.<br/>
<br/>
Issue #17273<br/>
Same scenario as issue 18395 but party B is an FXS port. Party B cannot<br/>
do anything until the transfer it started completes. If B goes back off<br/>
hook before C answers, B hears ringback instead of the expected dialtone.<br/>
<br/>
**********<br/>
Note for the issue #17273 and #18395 fix:<br/>
<br/>
DTMF attended transfer works within the channel bridge. Unfortunately,<br/>
when either party A or B in the channel bridge hangs up, that channel is<br/>
not completely hung up until the transfer completes. This is a real<br/>
problem depending upon the channel technology involved.<br/>
<br/>
For chan_dahdi, the channel is crippled until the hangup is complete.<br/>
Either the channel is not useable (analog) or the protocol disconnect<br/>
messages are held up (PRI/BRI/SS7) and the media is not released.<br/>
<br/>
For chan_sip, a call limit of one is going to block that endpoint from any<br/>
further calls until the hangup is complete.<br/>
<br/>
For party A this is a minor problem. The party A channel will only be in<br/>
this condition while party B is dialing and when party B and C are<br/>
conferring. The conversation between party B and C is expected to be a<br/>
short one. Party B is either asking a question of party C or announcing<br/>
party A. Also party A does not have much incentive to hangup at this<br/>
point.<br/>
<br/>
For party B this can be a major problem during a blonde transfer. (A<br/>
blonde transfer is our term for an attended transfer that is converted<br/>
into a blind transfer. :)) Party B could be the operator. When party B<br/>
hangs up, he assumes that he is out of the original call entirely. The<br/>
party B channel will be in this condition while party C is ringing, while<br/>
attempting to recall party B, and while waiting between call attempts.<br/>
<br/>
WARNING:<br/>
The ATXFER_NULL_TECH conditional is a hack to fix the problem. It will<br/>
replace the party B channel technology with a NULL channel driver to<br/>
complete hanging up the party B channel technology. The consequences of<br/>
this code is that the 'h' extension will not be able to access any channel<br/>
technology specific information like SIP statistics for the call.<br/>
<br/>
ATXFER_NULL_TECH is not defined by default.<br/>
**********<br/>
<br/>
(closes issue #17999)<br/>
Reported by: iskatel<br/>
Tested by: rmudgett<br/>
JIRA SWP-2246<br/>
<br/>
(closes issue #17096)<br/>
Reported by: gelo<br/>
Tested by: rmudgett<br/>
JIRA SWP-1192<br/>
<br/>
(closes issue #18395)<br/>
Reported by: shihchuan<br/>
Tested by: rmudgett<br/>
<br/>
(closes issue #17273)<br/>
Reported by: grecco<br/>
Tested by: rmudgett<br/>
<br/>
Review: <a href="https://reviewboard.asterisk.org/r/1047/">https://reviewboard.asterisk.org/r/1047/</a><br/>
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(302172)
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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>
DTMF transfer plays the wrong sounds for wrong number or other call failure.<br/>
<br/>
* Set the default for features.conf.sample xferfailsound option to "beeperr"<br/>
as documented instead of "pbx-invalid" and corrected the use of it in DTMF<br/>
blind transfer (#1).<br/>
<br/>
* Improved DTMF blind transfer handling of wrong numbers.<br/>
<br/>
Most of the concerns in this issue were taken care of by the patch for<br/>
issue 17999: Issues with DTMF triggered attended transfers.<br/>
<br/>
(closes issue #18379)<br/>
Reported by: gincantalupo<br/>
Tested by: rmudgett<br/>
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(302671)
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<a href="http://bamboo.asterisk.org/browse/author/seanbright" style="font-family: Arial, sans-serif; font-size: 13px; font-weight:bold; color:#000">
seanbright</a><br>
Add configure.lineno to svn:ignore
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(864)
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<br/><a href="http://bamboo.asterisk.org/browse/AST14-CENTOS55-77/commit" style="font-family: Arial, sans-serif; font-size: 13px; color:#036">2 more changes...</a>
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