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author | Richard Mudgett <rmudgett@digium.com> | 2011-09-19 19:03:38 +0000 |
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committer | Richard Mudgett <rmudgett@digium.com> | 2011-09-19 19:03:38 +0000 |
commit | 5c71a502a7c318d7efd33d8cb2f348fb710940e3 (patch) | |
tree | 2485c14ed0c9e00f2ba49bfc6aa05dc52671e4e6 /tests/Makefile | |
parent | f2fe72628e196ade288bc2c3b91d3bd7c56bdfeb (diff) |
Merged revisions 336659 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
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r336659 | rmudgett | 2011-09-19 13:51:19 -0500 (Mon, 19 Sep 2011) | 38 lines
Merged revisions 336658 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r336658 | rmudgett | 2011-09-19 13:46:40 -0500 (Mon, 19 Sep 2011) | 31 lines
Made Dial d and H options no longer immediately auto-answer the calling leg.
The Dial d and H options break DTMF attended transfer atxferdropcall
option.
1) Party A calls party B.
2) Party B does a DTMF attended transfer to Party C.
If the dialplan uses the Dial d or H options to call Party C then the Dial
application answers the call immediately before initiating the call leg to
Party C. The premature answer causes the transfer code to not invoke the
atxferdropcall=no behavior for a blonde transfer since Party C has
"answered". The transfer code thinks that Party B has "consulted" with
Party C when Party B hangs up and completes the transfer to Party A.
Party A now hears ringback until Party C actually answers.
ASTERISK-13294 Dial d option.
ASTERISK-11067 Dial H option to disconnect before answer.
The referenced issues made Dial answer with the d and H options because
many SIP and ISDN phones cannot send DTMF before the call is connected.
* Made require the dialplan to control when or if the call needs to be
answered to use the Dial application d and H options. (The call is no
longer surprise answered when using the Dial d or H options.)
Review: https://reviewboard.asterisk.org/r/1381/
JIRA AST-623
JIRA AST-666
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336662 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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