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!
I would love to hear your opinion about the current asterisk stability, both in 1.4 and 1.6 as compared to 1.2.
We are using 1.2 in our production environment and have no dared to do the switch.
Would you recomend to switch directly to 1.6?
Thanks