The Asterisk.org development team has released version
1.6.0-beta4.
Here are some highlights from the changes, with the
associated issue numbers
from bugs.digium.com if an issue was associated with the
change.
This release contains the following improvements:
- 12020, a CLI formatting improvement
- 11964, added the ability to get the original called
number on SS7 calls
- 11873, Added core API changes to handle T.38 origination
and termination
(The version of app_fax in Asterisk-addons now supports
this.)
- 11553, Added a status variable to the ChannelRedirect()
application
The changes in this release include fixes for the following
issues (trivial and
minor issues not included):
- 11960, a crash in chan_sip
- 12021, a crash related to invalid formats being specified
for voicemail
- 11779, fix enabling echo cancellation for incoming SS7
calls
- 11740, DTMF handling fixes
- 11864, Fixed device state reporting on incoming calls on
FXO
- 12012, a crash in chan_local
- Fix a regression in codec handling that was introduced in
1.6.0-beta3
A full list of changes can be found in the ChangeLog. This
release is available
for immediate download from http://downloads.digium
.com/.
Thank you for your support!
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.c
om--
asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-dev
|
