The change log link doesn't seem to be right, This seems to be the correct one
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-12.1.0-rc1 On Feb 6, 2014 11:15 PM, "Asterisk Development Team" < [email protected]> wrote: > The Asterisk Development Team has announced the first release candidate of > Asterisk 12.1.0. This release candidate is available for immediate > download at http://downloads.asterisk.org/pub/telephony/asterisk > > The release of Asterisk 12.1.0-rc1 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 candidate: > > * --- pjsip: fix support for allow=all > (Closes issue ASTERISK-23018. Reported by xrobau) > > * --- res_pjsip_session: Be less strict with core requested outgoing > capabilities. > (Closes issue ASTERISK-23082. Reported by xrobau) > > * --- res_stasis: Enable transfers and provide events when they occur. > (Closes issue ASTERISK-22984. Reported by David M. Lee) > > * --- ARI: Support channel variables in originate > (Closes issue ASTERISK-23051. Reported by Matt Jordan) > > * --- PJSIP: Fix address for ACK in NAT situations > (Closes issue ASTERISK-23106. Reported by Matt Jordan) > > For a full list of changes in this release candidate, please see the > ChangeLog: > > http://downloads.asterisk.org/pub/telephony/Asterisk/ChangeLog-12.1.0-rc1 > > 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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