The Asterisk Development Team is pleased to announce the release of Asterisk 1.8.9.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.9.0 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: * AST-2012-001: prevent crash when an SDP offer is received with an encrypted video stream when support for video is disabled and res_srtp is loaded. (closes issue ASTERISK-19202) Reported by: Catalin Sanda * Handle AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loop. Failing to handle AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop causes the loop to exit prematurely. This causes a variety of negative side effects, depending on when the loop exits. This patch handles the frame by essentially swallowing the frame in the local loop, as the current channel drivers expect the RTP bridge to handle the frame, and, in the case of the local bridge loop, no additional action is necessary. (closes issue ASTERISK-19095) Reported by: Stefan Schmidt Tested by: Matt Jordan * Fix timing source dependency issues with MOH. Prior to this patch, res_musiconhold existed at the same module priority level as the timing sources that it depends on. This would cause a problem when music on hold was reloaded, as the timing source could be changed after res_musiconhold was processed. This patch adds a new module priority level, AST_MODPRI_TIMING, that the various timing modules are now loaded at. This now occurs before loading other resource modules, such that the timing source is guaranteed to be set prior to resolving the timing source dependencies. (closes issue ASTERISK-17474) Reporter: Luke H Tested by: Luke H, Vladimir Mikhelson, zzsurf, Wes Van Tlghem, elguero, Thomas Arimont Patched by elguero * Fix RTP reference leak. If a blind transfer were initiated using a REFER without a prior reINVITE to place the call on hold, AND if Asterisk were sending RTCP reports, then there was a reference leak for the RTP instance of the transferrer. (closes issue ASTERISK-19192) Reported by: Tyuta Vitali * Fix blind transfers from failing if an 'h' extension is present. This prevents the 'h' extension from being run on the transferee channel when it is transferred via a native transfer mechanism such as SIP REFER. (closes issue ASTERISK-19173) Reported by: Ross Beer Tested by: Kristjan Vrban Patches: ASTERISK-19173 by Mark Michelson (license 5049) * Restore call progress code for analog ports. Extracting sig_analog from chan_dahdi lost call progress detection functionality. Fix analog ports from considering a call answered immediately after dialing has completed if the callprogress option is enabled. (closes issue ASTERISK-18841) Reported by: Richard Miller Patched by Richard Miller * Fix regression that 'rtp/rtcp set debup ip' only works when a port was also specified. (closes issue ASTERISK-18693) Reported by: Davide Dal Reviewed by: Walter Doekes For a full list of changes in this release candidate, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.9.0 Thank you for your continued support of Asterisk!
|