On Feb 6, 2014 11:26 PM, "Daniel Jenkins" <[email protected]> wrote: > > 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
Sorry! Stupid phone top posted ( workman blames his tools!)
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