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Restore SIP DTMF overlap dialing method.

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The recent fix for ASTERISK-17288 to get RFC3578 SIP overlap support
working correctly removed a long standing ability to do overlap dialing
using DTMF in the early media phase of a call.

See ASTERISK-18702 it has a very good description of the issue.

I started with Pavel Troller's chan_sip.diff patch on issue
ASTERISK-18702.

* Added 'dtmf' enum value to sip.conf allowoverlap config option.  The new
option value causes the Incomplte application to not send anything with
chan_sip so the caller can supply more digits via DTMF.

* Renames SIP_GET_DEST_PICKUP_EXTEN_FOUND to SIP_GET_DEST_EXTEN_MATCHMORE
since that is what it really means.

* Fixed get_destination() inconsistency with the pickup extension
matching.

* Fixed initialization of PAGE3 of global_flags in reload_config().

(closes issue ASTERISK-18702)
Reported by: Pavel Troller

Review: https://reviewboard.asterisk.org/r/1517/

Review: https://reviewboard.asterisk.org/r/1582/
........

Merged revisions 345273 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 345275 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345276 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Richard Mudgett
2011-11-14 22:05:39 +00:00
parent 1cef6cf8cd
commit 113612b9d6
4 changed files with 141 additions and 69 deletions
Download Patch File Download Diff File Expand all files Collapse all files

23
UPGRADE-1.8.txt
View File

@@ -20,6 +20,10 @@
From 1.6.2 to 1.8:
* chan_sip no longer sets HASH(SIP_CAUSE,<chan name>) on channels by default.
This must now be enabled by setting 'sipstorecause' to 'yes' in sip.conf.
This carries a performance penalty.
* Asterisk now requires libpri 1.4.11+ for PRI support.
* A couple of CLI commands in res_ais were changed back to their original form:
@@ -92,8 +96,8 @@ From 1.6.2 to 1.8:
* ExternalIVR will now send Z events for invalid or missing files, T events
now include the interrupted file and bugs in argument parsing have been
fixed so there may be arguments specified in incorrect ways that were
working that will no longer work.
Please see doc/externalivr.txt for details.
working that will no longer work. Please see
https://wiki.asterisk.org/wiki/display/AST/External+IVR+Interface for details.
* OSP lookup application changes following variable names:
OSPPEERIP to OSPINPEERIP
@@ -155,6 +159,13 @@ From 1.6.2 to 1.8:
changes to the files will not be detected. You can revert to polling the
directory by specifying --without-inotify to configure before compiling.
* The 'sipusers' realtime table has been removed completely. Use the 'sippeers'
table with type 'user' for user type objects.
* The sip.conf allowoverlap option now accepts 'dtmf' as a value. If you
are using the early media DTMF overlap dialing method you now need to set
allowoverlap=dtmf.
From 1.6.1 to 1.6.2:
* SIP no longer sends the 183 progress message for early media by
@@ -250,6 +261,11 @@ From 1.6.1 to 1.6.2:
* The cdr.conf file must exist and be configured correctly in order for CDR
records to be written.
* cdr_pgsql now assumes the encoding of strings it is handed are in LATIN9,
which should cover most uses of the extended ASCII set. If your strings
use a different encoding in Asterisk, the "encoding" parameter may be set
to specify the correct character set.
From 1.6.0.1 to 1.6.1:
* The ast_agi_register_multiple() and ast_agi_unregister_multiple()
@@ -311,6 +327,3 @@ From 1.6.0.x to 1.6.1:
which should be a char(8) or larger. This field specifies whether or not a
message has been designated to be "Urgent", "PRIORITY", or not.
* The 'sipusers' realtime table has been removed completely. Use the 'sippeers'
table with type 'user' for user type objects.