Il Team di Sviluppo di Asterisk ha annunciato il rilascio della terza beta della 1.8.0.
Dal post originale:
This release candidate 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
(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)
A short list of available 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