The release candidate of Asterisk 1.8.0. has been released by the Asterisk Development Team this release candidate 3, RC3 is available for immediate download from Asterisk.Org
Since any beta or pre release issues are tracked by greater participation of the Asterisk Community and all are invited and encouraged to participate in the 1.8 testing process. Please report any issues found to the Asterisk issue tracker. It is also very useful to see successful test reports those could be posted to the asterisk-dev mailing list.
Asterisk 1.8 is the next major release series of Asterisk and is slated to be a Long Term Support (LTS) release, just like the very successful, Asterisk 1.4. You can and should learn about Asterisk versions at Asterisk versions page.
The binary add-on modules for Asterisk produced by Digium have been updated to be compatible with Asterisk 1.8. The availability of these modules will assist with the testing of Asterisk 1.8.0 in a wider variety of situations and usage scenarios.
The release candidate RC3 contains fixes since the release candidate as reported by the community. A sampling of the changes in this release candidate include:
* Still build chan_sip even if res_crypto cannot be built (use, but not depend) (Reported by a user on the mailing list. Patched by tilghman) * Get notifications for call files only when a file is closed, not when created (Closes issue #17924. Reported by mkeuter. Patched by abeldeck)
* Fixes to chan_gtalk to allow outbound DTMF support to work correctly. Gtalk expects the DTMF to arrive on the RTP stream and not via jingle DTMF signalling.(Patched by dvossel. Tested by malcolmd)
* Fixes to allow chan_gtalk to communicate with the Gmail web client. (Patched by phsultan and dvossel)
* Fix to GET DATA to allow audio to be streamed via an AGI. (Closes issue #18001. Reported by jamicque. Patched by tilghman)
* Resolve dnsmgr memory corruption in chan_iax2.(Closes issue #17902. Reported by afried. Patched by russell, dvossel)
Some of the notable features includes:
* Secure RTP
* IPv6 Support in the SIP channel driver
* Connected Party Identification Support
* Calendaring Integration
* A new call logging system, Channel Event Logging (CEL)
* Distributed Device State using Jabber/XMPP PubSub
* Call Completion Supplementary Services support
* Advice of Charge support
* Much, much more!
A full list of new features can be found in the CHANGES file.
For a full list of changes in the current release candidate, please see the ChangeLog: