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authorRichard Mudgett <rmudgett@digium.com>2011-09-19 19:03:38 +0000
committerRichard Mudgett <rmudgett@digium.com>2011-09-19 19:03:38 +0000
commit5c71a502a7c318d7efd33d8cb2f348fb710940e3 (patch)
tree2485c14ed0c9e00f2ba49bfc6aa05dc52671e4e6 /UPGRADE-1.8.txt
parentf2fe72628e196ade288bc2c3b91d3bd7c56bdfeb (diff)
Merged revisions 336659 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ 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 ........ 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 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336662 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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diff --git a/UPGRADE-1.8.txt b/UPGRADE-1.8.txt
index ebafedd9e..230a5ef40 100644
--- a/UPGRADE-1.8.txt
+++ b/UPGRADE-1.8.txt
@@ -143,6 +143,12 @@ From 1.6.2 to 1.8:
events/responses output the connected line ID as caller ID. These party ID's
are now separate.
+* The Dial application d and H options do not automatically answer the call
+ anymore. It broke DTMF attended transfers. Since many SIP and ISDN phones
+ cannot send DTMF before a call is connected, you need to answer the call
+ leg to those phones before using Dial with these options for them to have
+ any effect before the dialed party answers.
+
* The outgoing directory (where .call files are read) now uses inotify to
detect file changes instead of polling the directory on a regular basis.
If your outgoing folder is on a NFS mount or another network file system,