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> From: Asterisk Development Team <asteriskt...@digium.com> > Date: 1 de junho de 2010 17:36:20 BRT > To: asteriskt...@digium.com > Subject: [asterisk-dev] Asterisk 1.6.2.8 Now Available > Reply-To: Asterisk Developers Mailing List <asterisk-...@lists.digium.com> > > > The Asterisk Development Team has announced the release of Asterisk 1.6.2.8. > This release is available for immediate download at > http://downloads.asterisk.org/pub/telephony/asterisk/ > > The release of Asterisk 1.6.2.8 resolves several issues reported by the > community, and would have not been possible without your participation. > Thank you! > > The following are a few of the issues resolved by community developers: > > * Enable auto complete for CLI command 'logger set level'. > (Closes issue #17152. Reported, patched by pabelanger) > > * Make the mixmonitor thread process audio frames faster. > (Closes issue #17078. Reported, tested by geoff2010. Patched by dhubbard) > > * Add missing 'useragent' field to sip-friends.sql file. > (Closes issue #17171. Reported, patched by thehar) > > * Add example dialplan for dialing ISN numbers (http://www.freenum.org) > (Closes issue #17058. Reported, patched by pprindeville) > > * Fix issue with double "sip:" in header field. > (Closes issue #15847. Reported, patched by ebroad) > > * Add ability to generate ASCII documentation from the TeX files by running > 'make asterisk.txt'. > (Closes issue #17220. Reported by lmadsen. Tested, patched by pabelanger) > > * When StopMonitor() is called, ensure that it will not be restarted by a > channel event. > (Closes issue #16590. Reported, patched by kkm) > > * Small error in the T.140 RTP port verbose log. > (Closes issue #16998. Reported, patched by frawd. Tested by russell) > > For a full list of changes in the current release, please see the > ChangeLog: > > http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.8 > > 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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