Begin forwarded message:
> From: Asterisk Development Team <asteriskt...@digium.com> > Date: 8 de dezembro de 2010 18:23:53 BRST > To: Asterisk Development Team <asteriskt...@digium.com> > Subject: [asterisk-dev] Asterisk 1.8.1 Now Available > Reply-To: Asterisk Developers Mailing List <asterisk-...@lists.digium.com> > > The Asterisk Development Team has announced the release of Asterisk 1.8.1. > This > release is available for immediate download at > http://downloads.asterisk.org/pub/telephony/asterisk/ > > The release of Asterisk 1.8.1 resolves several issues reported by the > community and would have not been possible without your participation. > Thank you! > > The following is a sample of the issues resolved in this release: > > * Fix issue when using directmedia. Asterisk needs to limit the codecs offered > to just the ones that both sides recognize, otherwise they may end up > sending > audio that the other side doesn't understand. > (Closes issue #17403. Reported, patched by one47. Tested by one47, falves11) > > * Resolve issue where Party A in an analog 3-way call would continue to hear > ringback after party C answers. > (Patched by rmudgett) > > * Fix playback failure when using IAX with the timerfd module. > (Closes issue #18110. Reported, tested by tpanton. Patched by jpeeler) > > * Fix problem with qualify option packets for realtime peers never stopping. > The option packets not only never stopped, but if a realtime peer was not in > the peer list multiple options dialogs could accumulate over time. > (Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by > jpeeler) > > * Fix issue where it is possible to crash Asterisk by feeding the curl engine > invalid data. > (Closes issue #18161. Reported by wdoekes. Patched by tilghman) > > For a full list of changes in this release, please see the ChangeLog: > > http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.1 > > 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
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