Denis at mobile.
Begin forwarded message: > From: Asterisk Development Team <asteriskt...@digium.com> > Date: 20 de setembro de 2012 16:40:00 BRT > To: asterisk-...@lists.digium.com > Subject: [asterisk-dev] Asterisk 11.0.0-beta2 Now Available! > Reply-To: Asterisk Developers Mailing List <asterisk-...@lists.digium.com> > > The Asterisk Development Team is pleased to announce the second beta release > of > Asterisk 11.0.0. This release is available for immediate download at > http://downloads.asterisk.org/pub/telephony/asterisk/releases > > All interested users of Asterisk are encouraged to participate in the > Asterisk 11 testing process. Please report any issues found to the issue > tracker, https://issues.asterisk.org/jira. It is also very useful to see > successful test reports. Please post those to the asterisk-dev mailing list. > All Asterisk users are invited to participate in the #asterisk-testing channel > on IRC to work together in testing the many parts of Asterisk. > > Asterisk 11 is the next major release series of Asterisk. It will be a Long > Term Support (LTS) release, similar to Asterisk 1.8. For more information > about > support time lines for Asterisk releases, see the Asterisk versions page: > https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions > > For important information regarding upgrading to Asterisk 11, please see the > Asterisk wiki: > > https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11 > > A short list of new features includes: > > * A new channel driver named chan_motif has been added which provides support > for Google Talk and Jingle in a single channel driver. This new channel > driver includes support for both audio and video, RFC2833 DTMF, all codecs > supported by Asterisk, hold, unhold, and ringing notification. It is also > compliant with the current Jingle specification, current Google Jingle > specification, and the original Google Talk protocol. > > * Support for the WebSocket transport for chan_sip. > > * SIP peers can now be configured to support negotiation of ICE candidates. > > * The app_page application now no longer depends on DAHDI or app_meetme. It > has been re-architected to use app_confbridge internally. > > * Hangup handlers can be attached to channels using the CHANNEL() function. > Hangup handlers will run when the channel is hung up similar to the h > extension; however, unlike an h extension, a hangup handler is associated > with > the actual channel and will execute anytime that channel is hung up, > regardless of where it is in the dialplan. > > * Added pre-dial handlers for the Dial and Follow-Me applications. Pre-dial > allows you to execute a dialplan subroutine on a channel before a call is > placed but after the application performing a dial action is invoked. This > means that the handlers are executed after the creation of the callee > channels, but before any actions have been taken to actually dial the callee > channels. > > * Log messages can now be easily associated with a certain call by looking at > a new unique identifier, "Call Id". Call ids are attached to log messages > for > just about any case where it can be determined that the message is related > to a particular call. > > * Introduced Named ACLs as a new way to define Access Control Lists (ACLs) in > Asterisk. Unlike traditional ACLs defined in specific module configuration > files, Named ACLs can be shared across multiple modules. > > * The Hangup Cause family of functions and dialplan applications allow for > inspection of the hangup cause codes for each channel involved in a call. > This allows a dialplan writer to determine, for each channel, who hung up and > for what reason(s). > > * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE() > lets you set some of the configuration options from the general section > of features.conf on a per-channel basis. FEATUREMAP() lets you customize > the key sequence used to activate built-in features, such as blindxfer, > and automon. > > * Support for DTLS-SRTP in chan_sip. > > * Support for named pickupgroups/callgroups, allowing any number of > pickupgroups > and callgroups to be defined for several channel drivers. > > * IPv6 Support for AMI, AGI, ExternalIVR, and the SIP Security Event > Framework. > > More information about the new features can be found on the Asterisk wiki: > > https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation > > A full list of all new features can also be found in the CHANGES file. > > http://svnview.digium.com/svn/asterisk/branches/11/CHANGES > > For a full list of changes in the current release, please see the ChangeLog. > > http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0-beta2 > > Thank you for your continued support of Asterisk! > > > > > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev
_______________________________________________ KHOMP Inovação: External Board Series Módulos de 1/2 rack e 1U para todas as interfaces e soluções Asterisk e FreeSWITCH. Tenha a External Series Experience na sua aplicação. Visite www.khomp.com _______________________________________________ DIGIVOICE Fabricante de Placas de Voz e Channel Bank 20 anos de experiência com E1(R2/ISDN), FXS, FXO e GSM Centro Treinamento - Curso de PABX IP - Asterisk - Site www.digivoice.com.br ________ YEALINK: Telefones IP e VídeoPhones IP com o melhor custo/benefício do mercado. email: yeal...@commlogik.com.br | www.commlogik.com.br | (11) 5503-1011 ______________________________________________ Para remover seu email desta lista, basta enviar um email em branco para asteriskbrasil-unsubscr...@listas.asteriskbrasil.org