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<a href="http://bamboo.asterisk.org/browse/AST162-CENTOS55-I386/log" style="font-family: Arial, sans-serif; font-size: 15px; font-weight:bold; color:#000">AST162-CENTOS55-I386-87</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/AST162-CENTOS55-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>
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/>
r293493 | twilson | 2010-11-01 09:58:00 -0500 (Mon, 01 Nov 2010) | 14 lines<br/>
<br/>
Only offer codecs both sides support for directmedia<br/>
<br/>
When using directmedia, Asterisk needs to limit the codecs offered to just<br/>
the ones that both sides recognize, otherwise they may end up sending audio<br/>
that the other side doesn't understand.<br/>
<br/>
(closes issue 0017403)<br/>
Reported by: one47<br/>
Patches: <br/>
sip_codecs_simplified4 uploaded by one47 (license 23)<br/>
Tested by: one47, falves11<br/>
<br/>
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/>
r302172 | rmudgett | 2011-01-18 12:04:36 -0600 (Tue, 18 Jan 2011) | 88 lines<br/>
<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/>
........<br/>
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(302173)
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